From 56ec4ffcbae8aeac6c5245fc7b825d02e2e6cefd Mon Sep 17 00:00:00 2001 From: Jean-Michel Trivi Date: Fri, 23 Jan 2015 16:45:18 -0800 Subject: Refactor AudioPolicyManager AudioPolicyManager implementation is now split into the following files: files managerdefault/Gains.* class AudioGain class VolumeCurvePoint class StreamDescriptor files managerdefault/Devices.* class DeviceDescriptor class DeviceVector files managerdefault/Ports.* class AudioPort class AudioPortConfig class AudioPatch files managerdefault/IOProfile.* class IOProfile files managerdefault/HwModule.* class HwModule files managerdefault/AudioInputDescriptor.* class AudioInputDescriptor files managerdefault/AudioOutputDescriptor.* class AudioOutputDescriptor All files for libaudiopolicyservice are moved under service/ All files for libaudiopolicymanager are moved under manager/ Change-Id: I43758be1894e37d34db194b51a19ae24461e066e --- media/mediaserver/main_mediaserver.cpp | 2 +- services/audiopolicy/Android.mk | 24 +- services/audiopolicy/AudioPolicyClientImpl.cpp | 221 - .../audiopolicy/AudioPolicyClientImplLegacy.cpp | 316 - services/audiopolicy/AudioPolicyEffects.cpp | 673 -- services/audiopolicy/AudioPolicyEffects.h | 196 - services/audiopolicy/AudioPolicyFactory.cpp | 32 - services/audiopolicy/AudioPolicyInterfaceImpl.cpp | 664 -- .../audiopolicy/AudioPolicyInterfaceImplLegacy.cpp | 607 -- services/audiopolicy/AudioPolicyManager.cpp | 8108 -------------------- services/audiopolicy/AudioPolicyManager.h | 951 --- services/audiopolicy/AudioPolicyService.cpp | 1068 --- services/audiopolicy/AudioPolicyService.h | 524 -- services/audiopolicy/audio_policy_conf.h | 77 - .../audiopolicy/manager/AudioPolicyFactory.cpp | 32 + .../managerdefault/ApmImplDefinitions.h | 32 + .../managerdefault/AudioInputDescriptor.cpp | 100 + .../managerdefault/AudioInputDescriptor.h | 48 + .../managerdefault/AudioOutputDescriptor.cpp | 221 + .../managerdefault/AudioOutputDescriptor.h | 69 + .../managerdefault/AudioPolicyManager.cpp | 5766 ++++++++++++++ .../managerdefault/AudioPolicyManager.h | 560 ++ .../managerdefault/ConfigParsingUtils.cpp | 121 + .../managerdefault/ConfigParsingUtils.h | 159 + services/audiopolicy/managerdefault/Devices.cpp | 282 + services/audiopolicy/managerdefault/Devices.h | 75 + services/audiopolicy/managerdefault/Gains.cpp | 446 ++ services/audiopolicy/managerdefault/Gains.h | 112 + services/audiopolicy/managerdefault/HwModule.cpp | 279 + services/audiopolicy/managerdefault/HwModule.h | 46 + services/audiopolicy/managerdefault/IOProfile.cpp | 139 + services/audiopolicy/managerdefault/IOProfile.h | 51 + services/audiopolicy/managerdefault/Ports.cpp | 844 ++ services/audiopolicy/managerdefault/Ports.h | 122 + .../audiopolicy/managerdefault/audio_policy_conf.h | 77 + .../audiopolicy/service/AudioPolicyClientImpl.cpp | 221 + .../service/AudioPolicyClientImplLegacy.cpp | 316 + .../audiopolicy/service/AudioPolicyEffects.cpp | 673 ++ services/audiopolicy/service/AudioPolicyEffects.h | 196 + .../service/AudioPolicyInterfaceImpl.cpp | 664 ++ .../service/AudioPolicyInterfaceImplLegacy.cpp | 607 ++ .../audiopolicy/service/AudioPolicyService.cpp | 1068 +++ services/audiopolicy/service/AudioPolicyService.h | 524 ++ 43 files changed, 13867 insertions(+), 13446 deletions(-) delete mode 100644 services/audiopolicy/AudioPolicyClientImpl.cpp delete mode 100644 services/audiopolicy/AudioPolicyClientImplLegacy.cpp delete mode 100644 services/audiopolicy/AudioPolicyEffects.cpp delete mode 100644 services/audiopolicy/AudioPolicyEffects.h delete mode 100644 services/audiopolicy/AudioPolicyFactory.cpp delete mode 100644 services/audiopolicy/AudioPolicyInterfaceImpl.cpp delete mode 100644 services/audiopolicy/AudioPolicyInterfaceImplLegacy.cpp delete mode 100644 services/audiopolicy/AudioPolicyManager.cpp delete mode 100644 services/audiopolicy/AudioPolicyManager.h delete mode 100644 services/audiopolicy/AudioPolicyService.cpp delete mode 100644 services/audiopolicy/AudioPolicyService.h delete mode 100644 services/audiopolicy/audio_policy_conf.h create mode 100644 services/audiopolicy/manager/AudioPolicyFactory.cpp create mode 100644 services/audiopolicy/managerdefault/ApmImplDefinitions.h create mode 100644 services/audiopolicy/managerdefault/AudioInputDescriptor.cpp create mode 100644 services/audiopolicy/managerdefault/AudioInputDescriptor.h create mode 100644 services/audiopolicy/managerdefault/AudioOutputDescriptor.cpp create mode 100644 services/audiopolicy/managerdefault/AudioOutputDescriptor.h create mode 100644 services/audiopolicy/managerdefault/AudioPolicyManager.cpp create mode 100644 services/audiopolicy/managerdefault/AudioPolicyManager.h create mode 100644 services/audiopolicy/managerdefault/ConfigParsingUtils.cpp create mode 100644 services/audiopolicy/managerdefault/ConfigParsingUtils.h create mode 100644 services/audiopolicy/managerdefault/Devices.cpp create mode 100644 services/audiopolicy/managerdefault/Devices.h create mode 100644 services/audiopolicy/managerdefault/Gains.cpp create mode 100644 services/audiopolicy/managerdefault/Gains.h create mode 100644 services/audiopolicy/managerdefault/HwModule.cpp create mode 100644 services/audiopolicy/managerdefault/HwModule.h create mode 100644 services/audiopolicy/managerdefault/IOProfile.cpp create mode 100644 services/audiopolicy/managerdefault/IOProfile.h create mode 100644 services/audiopolicy/managerdefault/Ports.cpp create mode 100644 services/audiopolicy/managerdefault/Ports.h create mode 100644 services/audiopolicy/managerdefault/audio_policy_conf.h create mode 100644 services/audiopolicy/service/AudioPolicyClientImpl.cpp create mode 100644 services/audiopolicy/service/AudioPolicyClientImplLegacy.cpp create mode 100644 services/audiopolicy/service/AudioPolicyEffects.cpp create mode 100644 services/audiopolicy/service/AudioPolicyEffects.h create mode 100644 services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp create mode 100644 services/audiopolicy/service/AudioPolicyInterfaceImplLegacy.cpp create mode 100644 services/audiopolicy/service/AudioPolicyService.cpp create mode 100644 services/audiopolicy/service/AudioPolicyService.h diff --git a/media/mediaserver/main_mediaserver.cpp b/media/mediaserver/main_mediaserver.cpp index af1c9e6..263dd32 100644 --- a/media/mediaserver/main_mediaserver.cpp +++ b/media/mediaserver/main_mediaserver.cpp @@ -33,7 +33,7 @@ #include "CameraService.h" #include "MediaLogService.h" #include "MediaPlayerService.h" -#include "AudioPolicyService.h" +#include "service/AudioPolicyService.h" #include "SoundTriggerHwService.h" using namespace android; diff --git a/services/audiopolicy/Android.mk b/services/audiopolicy/Android.mk index 188fc89..351ed79 100644 --- a/services/audiopolicy/Android.mk +++ b/services/audiopolicy/Android.mk @@ -3,19 +3,19 @@ LOCAL_PATH:= $(call my-dir) include $(CLEAR_VARS) LOCAL_SRC_FILES:= \ - AudioPolicyService.cpp \ - AudioPolicyEffects.cpp + service/AudioPolicyService.cpp \ + service/AudioPolicyEffects.cpp ifeq ($(USE_LEGACY_AUDIO_POLICY), 1) LOCAL_SRC_FILES += \ - AudioPolicyInterfaceImplLegacy.cpp \ - AudioPolicyClientImplLegacy.cpp + service/AudioPolicyInterfaceImplLegacy.cpp \ + service/AudioPolicyClientImplLegacy.cpp LOCAL_CFLAGS += -DUSE_LEGACY_AUDIO_POLICY else LOCAL_SRC_FILES += \ - AudioPolicyInterfaceImpl.cpp \ - AudioPolicyClientImpl.cpp + service/AudioPolicyInterfaceImpl.cpp \ + service/AudioPolicyClientImpl.cpp endif LOCAL_C_INCLUDES := \ @@ -53,7 +53,15 @@ ifneq ($(USE_LEGACY_AUDIO_POLICY), 1) include $(CLEAR_VARS) LOCAL_SRC_FILES:= \ - AudioPolicyManager.cpp + managerdefault/AudioPolicyManager.cpp \ + managerdefault/ConfigParsingUtils.cpp \ + managerdefault/Devices.cpp \ + managerdefault/Gains.cpp \ + managerdefault/HwModule.cpp \ + managerdefault/IOProfile.cpp \ + managerdefault/Ports.cpp \ + managerdefault/AudioInputDescriptor.cpp \ + managerdefault/AudioOutputDescriptor.cpp LOCAL_SHARED_LIBRARIES := \ libcutils \ @@ -73,7 +81,7 @@ ifneq ($(USE_CUSTOM_AUDIO_POLICY), 1) include $(CLEAR_VARS) LOCAL_SRC_FILES:= \ - AudioPolicyFactory.cpp + manager/AudioPolicyFactory.cpp LOCAL_SHARED_LIBRARIES := \ libaudiopolicymanagerdefault diff --git a/services/audiopolicy/AudioPolicyClientImpl.cpp b/services/audiopolicy/AudioPolicyClientImpl.cpp deleted file mode 100644 index 3e090e9..0000000 --- a/services/audiopolicy/AudioPolicyClientImpl.cpp +++ /dev/null @@ -1,221 +0,0 @@ -/* - * Copyright (C) 2009 The Android Open Source Project - * - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - */ - -#define LOG_TAG "AudioPolicyClientImpl" -//#define LOG_NDEBUG 0 - -#include -#include -#include "AudioPolicyService.h" - -namespace android { - -/* implementation of the client interface from the policy manager */ - -audio_module_handle_t AudioPolicyService::AudioPolicyClient::loadHwModule(const char *name) -{ - sp af = AudioSystem::get_audio_flinger(); - if (af == 0) { - ALOGW("%s: could not get AudioFlinger", __func__); - return 0; - } - - return af->loadHwModule(name); -} - -status_t AudioPolicyService::AudioPolicyClient::openOutput(audio_module_handle_t module, - audio_io_handle_t *output, - audio_config_t *config, - audio_devices_t *devices, - const String8& address, - uint32_t *latencyMs, - audio_output_flags_t flags) -{ - sp af = AudioSystem::get_audio_flinger(); - if (af == 0) { - ALOGW("%s: could not get AudioFlinger", __func__); - return PERMISSION_DENIED; - } - return af->openOutput(module, output, config, devices, address, latencyMs, flags); -} - -audio_io_handle_t AudioPolicyService::AudioPolicyClient::openDuplicateOutput( - audio_io_handle_t output1, - audio_io_handle_t output2) -{ - sp af = AudioSystem::get_audio_flinger(); - if (af == 0) { - ALOGW("%s: could not get AudioFlinger", __func__); - return 0; - } - return af->openDuplicateOutput(output1, output2); -} - -status_t AudioPolicyService::AudioPolicyClient::closeOutput(audio_io_handle_t output) -{ - sp af = AudioSystem::get_audio_flinger(); - if (af == 0) { - return PERMISSION_DENIED; - } - - return af->closeOutput(output); -} - -status_t AudioPolicyService::AudioPolicyClient::suspendOutput(audio_io_handle_t output) -{ - sp af = AudioSystem::get_audio_flinger(); - if (af == 0) { - ALOGW("%s: could not get AudioFlinger", __func__); - return PERMISSION_DENIED; - } - - return af->suspendOutput(output); -} - -status_t AudioPolicyService::AudioPolicyClient::restoreOutput(audio_io_handle_t output) -{ - sp af = AudioSystem::get_audio_flinger(); - if (af == 0) { - ALOGW("%s: could not get AudioFlinger", __func__); - return PERMISSION_DENIED; - } - - return af->restoreOutput(output); -} - -status_t AudioPolicyService::AudioPolicyClient::openInput(audio_module_handle_t module, - audio_io_handle_t *input, - audio_config_t *config, - audio_devices_t *device, - const String8& address, - audio_source_t source, - audio_input_flags_t flags) -{ - sp af = AudioSystem::get_audio_flinger(); - if (af == 0) { - ALOGW("%s: could not get AudioFlinger", __func__); - return PERMISSION_DENIED; - } - - return af->openInput(module, input, config, device, address, source, flags); -} - -status_t AudioPolicyService::AudioPolicyClient::closeInput(audio_io_handle_t input) -{ - sp af = AudioSystem::get_audio_flinger(); - if (af == 0) { - return PERMISSION_DENIED; - } - - return af->closeInput(input); -} - -status_t AudioPolicyService::AudioPolicyClient::setStreamVolume(audio_stream_type_t stream, - float volume, audio_io_handle_t output, - int delay_ms) -{ - return mAudioPolicyService->setStreamVolume(stream, volume, output, - delay_ms); -} - -status_t AudioPolicyService::AudioPolicyClient::invalidateStream(audio_stream_type_t stream) -{ - sp af = AudioSystem::get_audio_flinger(); - if (af == 0) { - return PERMISSION_DENIED; - } - - return af->invalidateStream(stream); -} - -void AudioPolicyService::AudioPolicyClient::setParameters(audio_io_handle_t io_handle, - const String8& keyValuePairs, - int delay_ms) -{ - mAudioPolicyService->setParameters(io_handle, keyValuePairs.string(), delay_ms); -} - -String8 AudioPolicyService::AudioPolicyClient::getParameters(audio_io_handle_t io_handle, - const String8& keys) -{ - String8 result = AudioSystem::getParameters(io_handle, keys); - return result; -} - -status_t AudioPolicyService::AudioPolicyClient::startTone(audio_policy_tone_t tone, - audio_stream_type_t stream) -{ - return mAudioPolicyService->startTone(tone, stream); -} - -status_t AudioPolicyService::AudioPolicyClient::stopTone() -{ - return mAudioPolicyService->stopTone(); -} - -status_t AudioPolicyService::AudioPolicyClient::setVoiceVolume(float volume, int delay_ms) -{ - return mAudioPolicyService->setVoiceVolume(volume, delay_ms); -} - -status_t AudioPolicyService::AudioPolicyClient::moveEffects(int session, - audio_io_handle_t src_output, - audio_io_handle_t dst_output) -{ - sp af = AudioSystem::get_audio_flinger(); - if (af == 0) { - return PERMISSION_DENIED; - } - - return af->moveEffects(session, src_output, dst_output); -} - -status_t AudioPolicyService::AudioPolicyClient::createAudioPatch(const struct audio_patch *patch, - audio_patch_handle_t *handle, - int delayMs) -{ - return mAudioPolicyService->clientCreateAudioPatch(patch, handle, delayMs); -} - -status_t AudioPolicyService::AudioPolicyClient::releaseAudioPatch(audio_patch_handle_t handle, - int delayMs) -{ - return mAudioPolicyService->clientReleaseAudioPatch(handle, delayMs); -} - -status_t AudioPolicyService::AudioPolicyClient::setAudioPortConfig( - const struct audio_port_config *config, - int delayMs) -{ - return mAudioPolicyService->clientSetAudioPortConfig(config, delayMs); -} - -void AudioPolicyService::AudioPolicyClient::onAudioPortListUpdate() -{ - mAudioPolicyService->onAudioPortListUpdate(); -} - -void AudioPolicyService::AudioPolicyClient::onAudioPatchListUpdate() -{ - mAudioPolicyService->onAudioPatchListUpdate(); -} - -audio_unique_id_t AudioPolicyService::AudioPolicyClient::newAudioUniqueId() -{ - return AudioSystem::newAudioUniqueId(); -} - -}; // namespace android diff --git a/services/audiopolicy/AudioPolicyClientImplLegacy.cpp b/services/audiopolicy/AudioPolicyClientImplLegacy.cpp deleted file mode 100644 index a79f8ae..0000000 --- a/services/audiopolicy/AudioPolicyClientImplLegacy.cpp +++ /dev/null @@ -1,316 +0,0 @@ -/* - * Copyright (C) 2009 The Android Open Source Project - * - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - */ - -#define LOG_TAG "AudioPolicyService" -//#define LOG_NDEBUG 0 - -#include "Configuration.h" -#undef __STRICT_ANSI__ -#define __STDINT_LIMITS -#define __STDC_LIMIT_MACROS -#include - -#include -#include -#include -#include -#include -#include -#include -#include "AudioPolicyService.h" -#include "ServiceUtilities.h" -#include -#include -#include -//#include - -#include -#include -#include -#include -#include -#include - - -namespace android { - -/* implementation of the interface to the policy manager */ -extern "C" { - -audio_module_handle_t aps_load_hw_module(void *service __unused, - const char *name) -{ - sp af = AudioSystem::get_audio_flinger(); - if (af == 0) { - ALOGW("%s: could not get AudioFlinger", __func__); - return 0; - } - - return af->loadHwModule(name); -} - -static audio_io_handle_t open_output(audio_module_handle_t module, - audio_devices_t *pDevices, - uint32_t *pSamplingRate, - audio_format_t *pFormat, - audio_channel_mask_t *pChannelMask, - uint32_t *pLatencyMs, - audio_output_flags_t flags, - const audio_offload_info_t *offloadInfo) -{ - sp af = AudioSystem::get_audio_flinger(); - if (af == 0) { - ALOGW("%s: could not get AudioFlinger", __func__); - return AUDIO_IO_HANDLE_NONE; - } - - if (pSamplingRate == NULL || pFormat == NULL || pChannelMask == NULL || - pDevices == NULL || pLatencyMs == NULL) { - return AUDIO_IO_HANDLE_NONE; - } - audio_config_t config = AUDIO_CONFIG_INITIALIZER; - config.sample_rate = *pSamplingRate; - config.format = *pFormat; - config.channel_mask = *pChannelMask; - if (offloadInfo != NULL) { - config.offload_info = *offloadInfo; - } - audio_io_handle_t output = AUDIO_IO_HANDLE_NONE; - status_t status = af->openOutput(module, &output, &config, pDevices, - String8(""), pLatencyMs, flags); - if (status == NO_ERROR) { - *pSamplingRate = config.sample_rate; - *pFormat = config.format; - *pChannelMask = config.channel_mask; - if (offloadInfo != NULL) { - *((audio_offload_info_t *)offloadInfo) = config.offload_info; - } - } - return output; -} - -// deprecated: replaced by aps_open_output_on_module() -audio_io_handle_t aps_open_output(void *service __unused, - audio_devices_t *pDevices, - uint32_t *pSamplingRate, - audio_format_t *pFormat, - audio_channel_mask_t *pChannelMask, - uint32_t *pLatencyMs, - audio_output_flags_t flags) -{ - return open_output((audio_module_handle_t)0, pDevices, pSamplingRate, pFormat, pChannelMask, - pLatencyMs, flags, NULL); -} - -audio_io_handle_t aps_open_output_on_module(void *service __unused, - audio_module_handle_t module, - audio_devices_t *pDevices, - uint32_t *pSamplingRate, - audio_format_t *pFormat, - audio_channel_mask_t *pChannelMask, - uint32_t *pLatencyMs, - audio_output_flags_t flags, - const audio_offload_info_t *offloadInfo) -{ - return open_output(module, pDevices, pSamplingRate, pFormat, pChannelMask, - pLatencyMs, flags, offloadInfo); -} - -audio_io_handle_t aps_open_dup_output(void *service __unused, - audio_io_handle_t output1, - audio_io_handle_t output2) -{ - sp af = AudioSystem::get_audio_flinger(); - if (af == 0) { - ALOGW("%s: could not get AudioFlinger", __func__); - return 0; - } - return af->openDuplicateOutput(output1, output2); -} - -int aps_close_output(void *service __unused, audio_io_handle_t output) -{ - sp af = AudioSystem::get_audio_flinger(); - if (af == 0) { - return PERMISSION_DENIED; - } - - return af->closeOutput(output); -} - -int aps_suspend_output(void *service __unused, audio_io_handle_t output) -{ - sp af = AudioSystem::get_audio_flinger(); - if (af == 0) { - ALOGW("%s: could not get AudioFlinger", __func__); - return PERMISSION_DENIED; - } - - return af->suspendOutput(output); -} - -int aps_restore_output(void *service __unused, audio_io_handle_t output) -{ - sp af = AudioSystem::get_audio_flinger(); - if (af == 0) { - ALOGW("%s: could not get AudioFlinger", __func__); - return PERMISSION_DENIED; - } - - return af->restoreOutput(output); -} - -static audio_io_handle_t open_input(audio_module_handle_t module, - audio_devices_t *pDevices, - uint32_t *pSamplingRate, - audio_format_t *pFormat, - audio_channel_mask_t *pChannelMask) -{ - sp af = AudioSystem::get_audio_flinger(); - if (af == 0) { - ALOGW("%s: could not get AudioFlinger", __func__); - return AUDIO_IO_HANDLE_NONE; - } - - if (pSamplingRate == NULL || pFormat == NULL || pChannelMask == NULL || pDevices == NULL) { - return AUDIO_IO_HANDLE_NONE; - } - - if (((*pDevices & AUDIO_DEVICE_IN_REMOTE_SUBMIX) == AUDIO_DEVICE_IN_REMOTE_SUBMIX) - && !captureAudioOutputAllowed()) { - ALOGE("open_input() permission denied: capture not allowed"); - return AUDIO_IO_HANDLE_NONE; - } - - audio_config_t config = AUDIO_CONFIG_INITIALIZER;; - config.sample_rate = *pSamplingRate; - config.format = *pFormat; - config.channel_mask = *pChannelMask; - audio_io_handle_t input = AUDIO_IO_HANDLE_NONE; - status_t status = af->openInput(module, &input, &config, pDevices, - String8(""), AUDIO_SOURCE_MIC, AUDIO_INPUT_FLAG_FAST /*FIXME*/); - if (status == NO_ERROR) { - *pSamplingRate = config.sample_rate; - *pFormat = config.format; - *pChannelMask = config.channel_mask; - } - return input; -} - - -// deprecated: replaced by aps_open_input_on_module(), and acoustics parameter is ignored -audio_io_handle_t aps_open_input(void *service __unused, - audio_devices_t *pDevices, - uint32_t *pSamplingRate, - audio_format_t *pFormat, - audio_channel_mask_t *pChannelMask, - audio_in_acoustics_t acoustics __unused) -{ - return open_input((audio_module_handle_t)0, pDevices, pSamplingRate, pFormat, pChannelMask); -} - -audio_io_handle_t aps_open_input_on_module(void *service __unused, - audio_module_handle_t module, - audio_devices_t *pDevices, - uint32_t *pSamplingRate, - audio_format_t *pFormat, - audio_channel_mask_t *pChannelMask) -{ - return open_input(module, pDevices, pSamplingRate, pFormat, pChannelMask); -} - -int aps_close_input(void *service __unused, audio_io_handle_t input) -{ - sp af = AudioSystem::get_audio_flinger(); - if (af == 0) { - return PERMISSION_DENIED; - } - - return af->closeInput(input); -} - -int aps_invalidate_stream(void *service __unused, audio_stream_type_t stream) -{ - sp af = AudioSystem::get_audio_flinger(); - if (af == 0) { - return PERMISSION_DENIED; - } - - return af->invalidateStream(stream); -} - -int aps_move_effects(void *service __unused, int session, - audio_io_handle_t src_output, - audio_io_handle_t dst_output) -{ - sp af = AudioSystem::get_audio_flinger(); - if (af == 0) { - return PERMISSION_DENIED; - } - - return af->moveEffects(session, src_output, dst_output); -} - -char * aps_get_parameters(void *service __unused, audio_io_handle_t io_handle, - const char *keys) -{ - String8 result = AudioSystem::getParameters(io_handle, String8(keys)); - return strdup(result.string()); -} - -void aps_set_parameters(void *service, audio_io_handle_t io_handle, - const char *kv_pairs, int delay_ms) -{ - AudioPolicyService *audioPolicyService = (AudioPolicyService *)service; - - audioPolicyService->setParameters(io_handle, kv_pairs, delay_ms); -} - -int aps_set_stream_volume(void *service, audio_stream_type_t stream, - float volume, audio_io_handle_t output, - int delay_ms) -{ - AudioPolicyService *audioPolicyService = (AudioPolicyService *)service; - - return audioPolicyService->setStreamVolume(stream, volume, output, - delay_ms); -} - -int aps_start_tone(void *service, audio_policy_tone_t tone, - audio_stream_type_t stream) -{ - AudioPolicyService *audioPolicyService = (AudioPolicyService *)service; - - return audioPolicyService->startTone(tone, stream); -} - -int aps_stop_tone(void *service) -{ - AudioPolicyService *audioPolicyService = (AudioPolicyService *)service; - - return audioPolicyService->stopTone(); -} - -int aps_set_voice_volume(void *service, float volume, int delay_ms) -{ - AudioPolicyService *audioPolicyService = (AudioPolicyService *)service; - - return audioPolicyService->setVoiceVolume(volume, delay_ms); -} - -}; // extern "C" - -}; // namespace android diff --git a/services/audiopolicy/AudioPolicyEffects.cpp b/services/audiopolicy/AudioPolicyEffects.cpp deleted file mode 100644 index e6ace20..0000000 --- a/services/audiopolicy/AudioPolicyEffects.cpp +++ /dev/null @@ -1,673 +0,0 @@ -/* - * Copyright (C) 2014 The Android Open Source Project - * - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - */ - -#define LOG_TAG "AudioPolicyEffects" -//#define LOG_NDEBUG 0 - -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include "AudioPolicyEffects.h" -#include "ServiceUtilities.h" - -namespace android { - -// ---------------------------------------------------------------------------- -// AudioPolicyEffects Implementation -// ---------------------------------------------------------------------------- - -AudioPolicyEffects::AudioPolicyEffects() -{ - // load automatic audio effect modules - if (access(AUDIO_EFFECT_VENDOR_CONFIG_FILE, R_OK) == 0) { - loadAudioEffectConfig(AUDIO_EFFECT_VENDOR_CONFIG_FILE); - } else if (access(AUDIO_EFFECT_DEFAULT_CONFIG_FILE, R_OK) == 0) { - loadAudioEffectConfig(AUDIO_EFFECT_DEFAULT_CONFIG_FILE); - } -} - - -AudioPolicyEffects::~AudioPolicyEffects() -{ - size_t i = 0; - // release audio input processing resources - for (i = 0; i < mInputSources.size(); i++) { - delete mInputSources.valueAt(i); - } - mInputSources.clear(); - - for (i = 0; i < mInputs.size(); i++) { - mInputs.valueAt(i)->mEffects.clear(); - delete mInputs.valueAt(i); - } - mInputs.clear(); - - // release audio output processing resources - for (i = 0; i < mOutputStreams.size(); i++) { - delete mOutputStreams.valueAt(i); - } - mOutputStreams.clear(); - - for (i = 0; i < mOutputSessions.size(); i++) { - mOutputSessions.valueAt(i)->mEffects.clear(); - delete mOutputSessions.valueAt(i); - } - mOutputSessions.clear(); -} - - -status_t AudioPolicyEffects::addInputEffects(audio_io_handle_t input, - audio_source_t inputSource, - int audioSession) -{ - status_t status = NO_ERROR; - - // create audio pre processors according to input source - audio_source_t aliasSource = (inputSource == AUDIO_SOURCE_HOTWORD) ? - AUDIO_SOURCE_VOICE_RECOGNITION : inputSource; - - Mutex::Autolock _l(mLock); - ssize_t index = mInputSources.indexOfKey(aliasSource); - if (index < 0) { - ALOGV("addInputEffects(): no processing needs to be attached to this source"); - return status; - } - ssize_t idx = mInputs.indexOfKey(input); - EffectVector *inputDesc; - if (idx < 0) { - inputDesc = new EffectVector(audioSession); - mInputs.add(input, inputDesc); - } else { - // EffectVector is existing and we just need to increase ref count - inputDesc = mInputs.valueAt(idx); - } - inputDesc->mRefCount++; - - ALOGV("addInputEffects(): input: %d, refCount: %d", input, inputDesc->mRefCount); - if (inputDesc->mRefCount == 1) { - Vector effects = mInputSources.valueAt(index)->mEffects; - for (size_t i = 0; i < effects.size(); i++) { - EffectDesc *effect = effects[i]; - sp fx = new AudioEffect(NULL, &effect->mUuid, -1, 0, 0, - audioSession, input); - status_t status = fx->initCheck(); - if (status != NO_ERROR && status != ALREADY_EXISTS) { - ALOGW("addInputEffects(): failed to create Fx %s on source %d", - effect->mName, (int32_t)aliasSource); - // fx goes out of scope and strong ref on AudioEffect is released - continue; - } - for (size_t j = 0; j < effect->mParams.size(); j++) { - fx->setParameter(effect->mParams[j]); - } - ALOGV("addInputEffects(): added Fx %s on source: %d", - effect->mName, (int32_t)aliasSource); - inputDesc->mEffects.add(fx); - } - inputDesc->setProcessorEnabled(true); - } - return status; -} - - -status_t AudioPolicyEffects::releaseInputEffects(audio_io_handle_t input) -{ - status_t status = NO_ERROR; - - Mutex::Autolock _l(mLock); - ssize_t index = mInputs.indexOfKey(input); - if (index < 0) { - return status; - } - EffectVector *inputDesc = mInputs.valueAt(index); - inputDesc->mRefCount--; - ALOGV("releaseInputEffects(): input: %d, refCount: %d", input, inputDesc->mRefCount); - if (inputDesc->mRefCount == 0) { - inputDesc->setProcessorEnabled(false); - delete inputDesc; - mInputs.removeItemsAt(index); - ALOGV("releaseInputEffects(): all effects released"); - } - return status; -} - -status_t AudioPolicyEffects::queryDefaultInputEffects(int audioSession, - effect_descriptor_t *descriptors, - uint32_t *count) -{ - status_t status = NO_ERROR; - - Mutex::Autolock _l(mLock); - size_t index; - for (index = 0; index < mInputs.size(); index++) { - if (mInputs.valueAt(index)->mSessionId == audioSession) { - break; - } - } - if (index == mInputs.size()) { - *count = 0; - return BAD_VALUE; - } - Vector< sp > effects = mInputs.valueAt(index)->mEffects; - - for (size_t i = 0; i < effects.size(); i++) { - effect_descriptor_t desc = effects[i]->descriptor(); - if (i < *count) { - descriptors[i] = desc; - } - } - if (effects.size() > *count) { - status = NO_MEMORY; - } - *count = effects.size(); - return status; -} - - -status_t AudioPolicyEffects::queryDefaultOutputSessionEffects(int audioSession, - effect_descriptor_t *descriptors, - uint32_t *count) -{ - status_t status = NO_ERROR; - - Mutex::Autolock _l(mLock); - size_t index; - for (index = 0; index < mOutputSessions.size(); index++) { - if (mOutputSessions.valueAt(index)->mSessionId == audioSession) { - break; - } - } - if (index == mOutputSessions.size()) { - *count = 0; - return BAD_VALUE; - } - Vector< sp > effects = mOutputSessions.valueAt(index)->mEffects; - - for (size_t i = 0; i < effects.size(); i++) { - effect_descriptor_t desc = effects[i]->descriptor(); - if (i < *count) { - descriptors[i] = desc; - } - } - if (effects.size() > *count) { - status = NO_MEMORY; - } - *count = effects.size(); - return status; -} - - -status_t AudioPolicyEffects::addOutputSessionEffects(audio_io_handle_t output, - audio_stream_type_t stream, - int audioSession) -{ - status_t status = NO_ERROR; - - Mutex::Autolock _l(mLock); - // create audio processors according to stream - // FIXME: should we have specific post processing settings for internal streams? - // default to media for now. - if (stream >= AUDIO_STREAM_PUBLIC_CNT) { - stream = AUDIO_STREAM_MUSIC; - } - ssize_t index = mOutputStreams.indexOfKey(stream); - if (index < 0) { - ALOGV("addOutputSessionEffects(): no output processing needed for this stream"); - return NO_ERROR; - } - - ssize_t idx = mOutputSessions.indexOfKey(audioSession); - EffectVector *procDesc; - if (idx < 0) { - procDesc = new EffectVector(audioSession); - mOutputSessions.add(audioSession, procDesc); - } else { - // EffectVector is existing and we just need to increase ref count - procDesc = mOutputSessions.valueAt(idx); - } - procDesc->mRefCount++; - - ALOGV("addOutputSessionEffects(): session: %d, refCount: %d", - audioSession, procDesc->mRefCount); - if (procDesc->mRefCount == 1) { - Vector effects = mOutputStreams.valueAt(index)->mEffects; - for (size_t i = 0; i < effects.size(); i++) { - EffectDesc *effect = effects[i]; - sp fx = new AudioEffect(NULL, &effect->mUuid, 0, 0, 0, - audioSession, output); - status_t status = fx->initCheck(); - if (status != NO_ERROR && status != ALREADY_EXISTS) { - ALOGE("addOutputSessionEffects(): failed to create Fx %s on session %d", - effect->mName, audioSession); - // fx goes out of scope and strong ref on AudioEffect is released - continue; - } - ALOGV("addOutputSessionEffects(): added Fx %s on session: %d for stream: %d", - effect->mName, audioSession, (int32_t)stream); - procDesc->mEffects.add(fx); - } - - procDesc->setProcessorEnabled(true); - } - return status; -} - -status_t AudioPolicyEffects::releaseOutputSessionEffects(audio_io_handle_t output, - audio_stream_type_t stream, - int audioSession) -{ - status_t status = NO_ERROR; - (void) output; // argument not used for now - (void) stream; // argument not used for now - - Mutex::Autolock _l(mLock); - ssize_t index = mOutputSessions.indexOfKey(audioSession); - if (index < 0) { - ALOGV("releaseOutputSessionEffects: no output processing was attached to this stream"); - return NO_ERROR; - } - - EffectVector *procDesc = mOutputSessions.valueAt(index); - procDesc->mRefCount--; - ALOGV("releaseOutputSessionEffects(): session: %d, refCount: %d", - audioSession, procDesc->mRefCount); - if (procDesc->mRefCount == 0) { - procDesc->setProcessorEnabled(false); - procDesc->mEffects.clear(); - delete procDesc; - mOutputSessions.removeItemsAt(index); - ALOGV("releaseOutputSessionEffects(): output processing released from session: %d", - audioSession); - } - return status; -} - - -void AudioPolicyEffects::EffectVector::setProcessorEnabled(bool enabled) -{ - for (size_t i = 0; i < mEffects.size(); i++) { - mEffects.itemAt(i)->setEnabled(enabled); - } -} - - -// ---------------------------------------------------------------------------- -// Audio processing configuration -// ---------------------------------------------------------------------------- - -/*static*/ const char * const AudioPolicyEffects::kInputSourceNames[AUDIO_SOURCE_CNT -1] = { - MIC_SRC_TAG, - VOICE_UL_SRC_TAG, - VOICE_DL_SRC_TAG, - VOICE_CALL_SRC_TAG, - CAMCORDER_SRC_TAG, - VOICE_REC_SRC_TAG, - VOICE_COMM_SRC_TAG -}; - -// returns the audio_source_t enum corresponding to the input source name or -// AUDIO_SOURCE_CNT is no match found -/*static*/ audio_source_t AudioPolicyEffects::inputSourceNameToEnum(const char *name) -{ - int i; - for (i = AUDIO_SOURCE_MIC; i < AUDIO_SOURCE_CNT; i++) { - if (strcmp(name, kInputSourceNames[i - AUDIO_SOURCE_MIC]) == 0) { - ALOGV("inputSourceNameToEnum found source %s %d", name, i); - break; - } - } - return (audio_source_t)i; -} - -const char *AudioPolicyEffects::kStreamNames[AUDIO_STREAM_PUBLIC_CNT+1] = { - AUDIO_STREAM_DEFAULT_TAG, - AUDIO_STREAM_VOICE_CALL_TAG, - AUDIO_STREAM_SYSTEM_TAG, - AUDIO_STREAM_RING_TAG, - AUDIO_STREAM_MUSIC_TAG, - AUDIO_STREAM_ALARM_TAG, - AUDIO_STREAM_NOTIFICATION_TAG, - AUDIO_STREAM_BLUETOOTH_SCO_TAG, - AUDIO_STREAM_ENFORCED_AUDIBLE_TAG, - AUDIO_STREAM_DTMF_TAG, - AUDIO_STREAM_TTS_TAG -}; - -// returns the audio_stream_t enum corresponding to the output stream name or -// AUDIO_STREAM_PUBLIC_CNT is no match found -audio_stream_type_t AudioPolicyEffects::streamNameToEnum(const char *name) -{ - int i; - for (i = AUDIO_STREAM_DEFAULT; i < AUDIO_STREAM_PUBLIC_CNT; i++) { - if (strcmp(name, kStreamNames[i - AUDIO_STREAM_DEFAULT]) == 0) { - ALOGV("streamNameToEnum found stream %s %d", name, i); - break; - } - } - return (audio_stream_type_t)i; -} - -// ---------------------------------------------------------------------------- -// Audio Effect Config parser -// ---------------------------------------------------------------------------- - -size_t AudioPolicyEffects::growParamSize(char *param, - size_t size, - size_t *curSize, - size_t *totSize) -{ - // *curSize is at least sizeof(effect_param_t) + 2 * sizeof(int) - size_t pos = ((*curSize - 1 ) / size + 1) * size; - - if (pos + size > *totSize) { - while (pos + size > *totSize) { - *totSize += ((*totSize + 7) / 8) * 4; - } - param = (char *)realloc(param, *totSize); - } - *curSize = pos + size; - return pos; -} - -size_t AudioPolicyEffects::readParamValue(cnode *node, - char *param, - size_t *curSize, - size_t *totSize) -{ - if (strncmp(node->name, SHORT_TAG, sizeof(SHORT_TAG) + 1) == 0) { - size_t pos = growParamSize(param, sizeof(short), curSize, totSize); - *(short *)((char *)param + pos) = (short)atoi(node->value); - ALOGV("readParamValue() reading short %d", *(short *)((char *)param + pos)); - return sizeof(short); - } else if (strncmp(node->name, INT_TAG, sizeof(INT_TAG) + 1) == 0) { - size_t pos = growParamSize(param, sizeof(int), curSize, totSize); - *(int *)((char *)param + pos) = atoi(node->value); - ALOGV("readParamValue() reading int %d", *(int *)((char *)param + pos)); - return sizeof(int); - } else if (strncmp(node->name, FLOAT_TAG, sizeof(FLOAT_TAG) + 1) == 0) { - size_t pos = growParamSize(param, sizeof(float), curSize, totSize); - *(float *)((char *)param + pos) = (float)atof(node->value); - ALOGV("readParamValue() reading float %f",*(float *)((char *)param + pos)); - return sizeof(float); - } else if (strncmp(node->name, BOOL_TAG, sizeof(BOOL_TAG) + 1) == 0) { - size_t pos = growParamSize(param, sizeof(bool), curSize, totSize); - if (strncmp(node->value, "false", strlen("false") + 1) == 0) { - *(bool *)((char *)param + pos) = false; - } else { - *(bool *)((char *)param + pos) = true; - } - ALOGV("readParamValue() reading bool %s",*(bool *)((char *)param + pos) ? "true" : "false"); - return sizeof(bool); - } else if (strncmp(node->name, STRING_TAG, sizeof(STRING_TAG) + 1) == 0) { - size_t len = strnlen(node->value, EFFECT_STRING_LEN_MAX); - if (*curSize + len + 1 > *totSize) { - *totSize = *curSize + len + 1; - param = (char *)realloc(param, *totSize); - } - strncpy(param + *curSize, node->value, len); - *curSize += len; - param[*curSize] = '\0'; - ALOGV("readParamValue() reading string %s", param + *curSize - len); - return len; - } - ALOGW("readParamValue() unknown param type %s", node->name); - return 0; -} - -effect_param_t *AudioPolicyEffects::loadEffectParameter(cnode *root) -{ - cnode *param; - cnode *value; - size_t curSize = sizeof(effect_param_t); - size_t totSize = sizeof(effect_param_t) + 2 * sizeof(int); - effect_param_t *fx_param = (effect_param_t *)malloc(totSize); - - param = config_find(root, PARAM_TAG); - value = config_find(root, VALUE_TAG); - if (param == NULL && value == NULL) { - // try to parse simple parameter form {int int} - param = root->first_child; - if (param != NULL) { - // Note: that a pair of random strings is read as 0 0 - int *ptr = (int *)fx_param->data; - int *ptr2 = (int *)((char *)param + sizeof(effect_param_t)); - ALOGW("loadEffectParameter() ptr %p ptr2 %p", ptr, ptr2); - *ptr++ = atoi(param->name); - *ptr = atoi(param->value); - fx_param->psize = sizeof(int); - fx_param->vsize = sizeof(int); - return fx_param; - } - } - if (param == NULL || value == NULL) { - ALOGW("loadEffectParameter() invalid parameter description %s", root->name); - goto error; - } - - fx_param->psize = 0; - param = param->first_child; - while (param) { - ALOGV("loadEffectParameter() reading param of type %s", param->name); - size_t size = readParamValue(param, (char *)fx_param, &curSize, &totSize); - if (size == 0) { - goto error; - } - fx_param->psize += size; - param = param->next; - } - - // align start of value field on 32 bit boundary - curSize = ((curSize - 1 ) / sizeof(int) + 1) * sizeof(int); - - fx_param->vsize = 0; - value = value->first_child; - while (value) { - ALOGV("loadEffectParameter() reading value of type %s", value->name); - size_t size = readParamValue(value, (char *)fx_param, &curSize, &totSize); - if (size == 0) { - goto error; - } - fx_param->vsize += size; - value = value->next; - } - - return fx_param; - -error: - delete fx_param; - return NULL; -} - -void AudioPolicyEffects::loadEffectParameters(cnode *root, Vector & params) -{ - cnode *node = root->first_child; - while (node) { - ALOGV("loadEffectParameters() loading param %s", node->name); - effect_param_t *param = loadEffectParameter(node); - if (param == NULL) { - node = node->next; - continue; - } - params.add(param); - node = node->next; - } -} - - -AudioPolicyEffects::EffectDescVector *AudioPolicyEffects::loadEffectConfig( - cnode *root, - const Vector & effects) -{ - cnode *node = root->first_child; - if (node == NULL) { - ALOGW("loadInputSource() empty element %s", root->name); - return NULL; - } - EffectDescVector *desc = new EffectDescVector(); - while (node) { - size_t i; - for (i = 0; i < effects.size(); i++) { - if (strncmp(effects[i]->mName, node->name, EFFECT_STRING_LEN_MAX) == 0) { - ALOGV("loadEffectConfig() found effect %s in list", node->name); - break; - } - } - if (i == effects.size()) { - ALOGV("loadEffectConfig() effect %s not in list", node->name); - node = node->next; - continue; - } - EffectDesc *effect = new EffectDesc(*effects[i]); // deep copy - loadEffectParameters(node, effect->mParams); - ALOGV("loadEffectConfig() adding effect %s uuid %08x", - effect->mName, effect->mUuid.timeLow); - desc->mEffects.add(effect); - node = node->next; - } - if (desc->mEffects.size() == 0) { - ALOGW("loadEffectConfig() no valid effects found in config %s", root->name); - delete desc; - return NULL; - } - return desc; -} - -status_t AudioPolicyEffects::loadInputEffectConfigurations(cnode *root, - const Vector & effects) -{ - cnode *node = config_find(root, PREPROCESSING_TAG); - if (node == NULL) { - return -ENOENT; - } - node = node->first_child; - while (node) { - audio_source_t source = inputSourceNameToEnum(node->name); - if (source == AUDIO_SOURCE_CNT) { - ALOGW("loadInputSources() invalid input source %s", node->name); - node = node->next; - continue; - } - ALOGV("loadInputSources() loading input source %s", node->name); - EffectDescVector *desc = loadEffectConfig(node, effects); - if (desc == NULL) { - node = node->next; - continue; - } - mInputSources.add(source, desc); - node = node->next; - } - return NO_ERROR; -} - -status_t AudioPolicyEffects::loadStreamEffectConfigurations(cnode *root, - const Vector & effects) -{ - cnode *node = config_find(root, OUTPUT_SESSION_PROCESSING_TAG); - if (node == NULL) { - return -ENOENT; - } - node = node->first_child; - while (node) { - audio_stream_type_t stream = streamNameToEnum(node->name); - if (stream == AUDIO_STREAM_PUBLIC_CNT) { - ALOGW("loadStreamEffectConfigurations() invalid output stream %s", node->name); - node = node->next; - continue; - } - ALOGV("loadStreamEffectConfigurations() loading output stream %s", node->name); - EffectDescVector *desc = loadEffectConfig(node, effects); - if (desc == NULL) { - node = node->next; - continue; - } - mOutputStreams.add(stream, desc); - node = node->next; - } - return NO_ERROR; -} - -AudioPolicyEffects::EffectDesc *AudioPolicyEffects::loadEffect(cnode *root) -{ - cnode *node = config_find(root, UUID_TAG); - if (node == NULL) { - return NULL; - } - effect_uuid_t uuid; - if (AudioEffect::stringToGuid(node->value, &uuid) != NO_ERROR) { - ALOGW("loadEffect() invalid uuid %s", node->value); - return NULL; - } - return new EffectDesc(root->name, uuid); -} - -status_t AudioPolicyEffects::loadEffects(cnode *root, Vector & effects) -{ - cnode *node = config_find(root, EFFECTS_TAG); - if (node == NULL) { - return -ENOENT; - } - node = node->first_child; - while (node) { - ALOGV("loadEffects() loading effect %s", node->name); - EffectDesc *effect = loadEffect(node); - if (effect == NULL) { - node = node->next; - continue; - } - effects.add(effect); - node = node->next; - } - return NO_ERROR; -} - -status_t AudioPolicyEffects::loadAudioEffectConfig(const char *path) -{ - cnode *root; - char *data; - - data = (char *)load_file(path, NULL); - if (data == NULL) { - return -ENODEV; - } - root = config_node("", ""); - config_load(root, data); - - Vector effects; - loadEffects(root, effects); - loadInputEffectConfigurations(root, effects); - loadStreamEffectConfigurations(root, effects); - - for (size_t i = 0; i < effects.size(); i++) { - delete effects[i]; - } - - config_free(root); - free(root); - free(data); - - return NO_ERROR; -} - - -}; // namespace android diff --git a/services/audiopolicy/AudioPolicyEffects.h b/services/audiopolicy/AudioPolicyEffects.h deleted file mode 100644 index 3dec437..0000000 --- a/services/audiopolicy/AudioPolicyEffects.h +++ /dev/null @@ -1,196 +0,0 @@ -/* - * Copyright (C) 2014 The Android Open Source Project - * - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - */ - -#ifndef ANDROID_AUDIOPOLICYEFFECTS_H -#define ANDROID_AUDIOPOLICYEFFECTS_H - -#include -#include -#include -#include -#include -#include -#include -#include -#include - -namespace android { - -// ---------------------------------------------------------------------------- - -// AudioPolicyEffects class -// This class will manage all effects attached to input and output streams in -// AudioPolicyService as configured in audio_effects.conf. -class AudioPolicyEffects : public RefBase -{ - -public: - - // The constructor will parse audio_effects.conf - // First it will look whether vendor specific file exists, - // otherwise it will parse the system default file. - AudioPolicyEffects(); - virtual ~AudioPolicyEffects(); - - // NOTE: methods on AudioPolicyEffects should never be called with the AudioPolicyService - // main mutex (mLock) held as they will indirectly call back into AudioPolicyService when - // managing audio effects. - - // Return a list of effect descriptors for default input effects - // associated with audioSession - status_t queryDefaultInputEffects(int audioSession, - effect_descriptor_t *descriptors, - uint32_t *count); - - // Add all input effects associated with this input - // Effects are attached depending on the audio_source_t - status_t addInputEffects(audio_io_handle_t input, - audio_source_t inputSource, - int audioSession); - - // Add all input effects associated to this input - status_t releaseInputEffects(audio_io_handle_t input); - - - // Return a list of effect descriptors for default output effects - // associated with audioSession - status_t queryDefaultOutputSessionEffects(int audioSession, - effect_descriptor_t *descriptors, - uint32_t *count); - - // Add all output effects associated to this output - // Effects are attached depending on the audio_stream_type_t - status_t addOutputSessionEffects(audio_io_handle_t output, - audio_stream_type_t stream, - int audioSession); - - // release all output effects associated with this output stream and audiosession - status_t releaseOutputSessionEffects(audio_io_handle_t output, - audio_stream_type_t stream, - int audioSession); - -private: - - // class to store the description of an effects and its parameters - // as defined in audio_effects.conf - class EffectDesc { - public: - EffectDesc(const char *name, const effect_uuid_t& uuid) : - mName(strdup(name)), - mUuid(uuid) { } - EffectDesc(const EffectDesc& orig) : - mName(strdup(orig.mName)), - mUuid(orig.mUuid) { - // deep copy mParams - for (size_t k = 0; k < orig.mParams.size(); k++) { - effect_param_t *origParam = orig.mParams[k]; - // psize and vsize are rounded up to an int boundary for allocation - size_t origSize = sizeof(effect_param_t) + - ((origParam->psize + 3) & ~3) + - ((origParam->vsize + 3) & ~3); - effect_param_t *dupParam = (effect_param_t *) malloc(origSize); - memcpy(dupParam, origParam, origSize); - // This works because the param buffer allocation is also done by - // multiples of 4 bytes originally. In theory we should memcpy only - // the actual param size, that is without rounding vsize. - mParams.add(dupParam); - } - } - /*virtual*/ ~EffectDesc() { - free(mName); - for (size_t k = 0; k < mParams.size(); k++) { - free(mParams[k]); - } - } - char *mName; - effect_uuid_t mUuid; - Vector mParams; - }; - - // class to store voctor of EffectDesc - class EffectDescVector { - public: - EffectDescVector() {} - /*virtual*/ ~EffectDescVector() { - for (size_t j = 0; j < mEffects.size(); j++) { - delete mEffects[j]; - } - } - Vector mEffects; - }; - - // class to store voctor of AudioEffects - class EffectVector { - public: - EffectVector(int session) : mSessionId(session), mRefCount(0) {} - /*virtual*/ ~EffectVector() {} - - // Enable or disable all effects in effect vector - void setProcessorEnabled(bool enabled); - - const int mSessionId; - // AudioPolicyManager keeps mLock, no need for lock on reference count here - int mRefCount; - Vector< sp >mEffects; - }; - - - static const char * const kInputSourceNames[AUDIO_SOURCE_CNT -1]; - static audio_source_t inputSourceNameToEnum(const char *name); - - static const char *kStreamNames[AUDIO_STREAM_PUBLIC_CNT+1]; //+1 required as streams start from -1 - audio_stream_type_t streamNameToEnum(const char *name); - - // Parse audio_effects.conf - status_t loadAudioEffectConfig(const char *path); - - // Load all effects descriptors in configuration file - status_t loadEffects(cnode *root, Vector & effects); - EffectDesc *loadEffect(cnode *root); - - // Load all automatic effect configurations - status_t loadInputEffectConfigurations(cnode *root, const Vector & effects); - status_t loadStreamEffectConfigurations(cnode *root, const Vector & effects); - EffectDescVector *loadEffectConfig(cnode *root, const Vector & effects); - - // Load all automatic effect parameters - void loadEffectParameters(cnode *root, Vector & params); - effect_param_t *loadEffectParameter(cnode *root); - size_t readParamValue(cnode *node, - char *param, - size_t *curSize, - size_t *totSize); - size_t growParamSize(char *param, - size_t size, - size_t *curSize, - size_t *totSize); - - // protects access to mInputSources, mInputs, mOutputStreams, mOutputSessions - Mutex mLock; - // Automatic input effects are configured per audio_source_t - KeyedVector< audio_source_t, EffectDescVector* > mInputSources; - // Automatic input effects are unique for audio_io_handle_t - KeyedVector< audio_io_handle_t, EffectVector* > mInputs; - - // Automatic output effects are organized per audio_stream_type_t - KeyedVector< audio_stream_type_t, EffectDescVector* > mOutputStreams; - // Automatic output effects are unique for audiosession ID - KeyedVector< int32_t, EffectVector* > mOutputSessions; -}; - -}; // namespace android - -#endif // ANDROID_AUDIOPOLICYEFFECTS_H diff --git a/services/audiopolicy/AudioPolicyFactory.cpp b/services/audiopolicy/AudioPolicyFactory.cpp deleted file mode 100644 index 2ae7bc1..0000000 --- a/services/audiopolicy/AudioPolicyFactory.cpp +++ /dev/null @@ -1,32 +0,0 @@ -/* - * Copyright (C) 2014 The Android Open Source Project - * - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - */ - -#include "AudioPolicyManager.h" - -namespace android { - -extern "C" AudioPolicyInterface* createAudioPolicyManager( - AudioPolicyClientInterface *clientInterface) -{ - return new AudioPolicyManager(clientInterface); -} - -extern "C" void destroyAudioPolicyManager(AudioPolicyInterface *interface) -{ - delete interface; -} - -}; // namespace android diff --git a/services/audiopolicy/AudioPolicyInterfaceImpl.cpp b/services/audiopolicy/AudioPolicyInterfaceImpl.cpp deleted file mode 100644 index e9ff838..0000000 --- a/services/audiopolicy/AudioPolicyInterfaceImpl.cpp +++ /dev/null @@ -1,664 +0,0 @@ -/* - * Copyright (C) 2009 The Android Open Source Project - * - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - */ - -#define LOG_TAG "AudioPolicyIntefaceImpl" -//#define LOG_NDEBUG 0 - -#include -#include "AudioPolicyService.h" -#include "ServiceUtilities.h" - -namespace android { - - -// ---------------------------------------------------------------------------- - -status_t AudioPolicyService::setDeviceConnectionState(audio_devices_t device, - audio_policy_dev_state_t state, - const char *device_address, - const char *device_name) -{ - if (mAudioPolicyManager == NULL) { - return NO_INIT; - } - if (!settingsAllowed()) { - return PERMISSION_DENIED; - } - if (!audio_is_output_device(device) && !audio_is_input_device(device)) { - return BAD_VALUE; - } - if (state != AUDIO_POLICY_DEVICE_STATE_AVAILABLE && - state != AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) { - return BAD_VALUE; - } - - ALOGV("setDeviceConnectionState()"); - Mutex::Autolock _l(mLock); - return mAudioPolicyManager->setDeviceConnectionState(device, state, - device_address, device_name); -} - -audio_policy_dev_state_t AudioPolicyService::getDeviceConnectionState( - audio_devices_t device, - const char *device_address) -{ - if (mAudioPolicyManager == NULL) { - return AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE; - } - return mAudioPolicyManager->getDeviceConnectionState(device, - device_address); -} - -status_t AudioPolicyService::setPhoneState(audio_mode_t state) -{ - if (mAudioPolicyManager == NULL) { - return NO_INIT; - } - if (!settingsAllowed()) { - return PERMISSION_DENIED; - } - if (uint32_t(state) >= AUDIO_MODE_CNT) { - return BAD_VALUE; - } - - ALOGV("setPhoneState()"); - - // TODO: check if it is more appropriate to do it in platform specific policy manager - AudioSystem::setMode(state); - - Mutex::Autolock _l(mLock); - mAudioPolicyManager->setPhoneState(state); - mPhoneState = state; - return NO_ERROR; -} - -audio_mode_t AudioPolicyService::getPhoneState() -{ - Mutex::Autolock _l(mLock); - return mPhoneState; -} - -status_t AudioPolicyService::setForceUse(audio_policy_force_use_t usage, - audio_policy_forced_cfg_t config) -{ - if (mAudioPolicyManager == NULL) { - return NO_INIT; - } - if (!settingsAllowed()) { - return PERMISSION_DENIED; - } - if (usage < 0 || usage >= AUDIO_POLICY_FORCE_USE_CNT) { - return BAD_VALUE; - } - if (config < 0 || config >= AUDIO_POLICY_FORCE_CFG_CNT) { - return BAD_VALUE; - } - ALOGV("setForceUse()"); - Mutex::Autolock _l(mLock); - mAudioPolicyManager->setForceUse(usage, config); - return NO_ERROR; -} - -audio_policy_forced_cfg_t AudioPolicyService::getForceUse(audio_policy_force_use_t usage) -{ - if (mAudioPolicyManager == NULL) { - return AUDIO_POLICY_FORCE_NONE; - } - if (usage < 0 || usage >= AUDIO_POLICY_FORCE_USE_CNT) { - return AUDIO_POLICY_FORCE_NONE; - } - return mAudioPolicyManager->getForceUse(usage); -} - -audio_io_handle_t AudioPolicyService::getOutput(audio_stream_type_t stream, - uint32_t samplingRate, - audio_format_t format, - audio_channel_mask_t channelMask, - audio_output_flags_t flags, - const audio_offload_info_t *offloadInfo) -{ - if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT) { - return AUDIO_IO_HANDLE_NONE; - } - if (mAudioPolicyManager == NULL) { - return AUDIO_IO_HANDLE_NONE; - } - ALOGV("getOutput()"); - Mutex::Autolock _l(mLock); - return mAudioPolicyManager->getOutput(stream, samplingRate, - format, channelMask, flags, offloadInfo); -} - -status_t AudioPolicyService::getOutputForAttr(const audio_attributes_t *attr, - audio_io_handle_t *output, - audio_session_t session, - audio_stream_type_t *stream, - uint32_t samplingRate, - audio_format_t format, - audio_channel_mask_t channelMask, - audio_output_flags_t flags, - const audio_offload_info_t *offloadInfo) -{ - if (mAudioPolicyManager == NULL) { - return NO_INIT; - } - ALOGV("getOutput()"); - Mutex::Autolock _l(mLock); - return mAudioPolicyManager->getOutputForAttr(attr, output, session, stream, samplingRate, - format, channelMask, flags, offloadInfo); -} - -status_t AudioPolicyService::startOutput(audio_io_handle_t output, - audio_stream_type_t stream, - audio_session_t session) -{ - if (uint32_t(stream) >= AUDIO_STREAM_CNT) { - return BAD_VALUE; - } - if (mAudioPolicyManager == NULL) { - return NO_INIT; - } - ALOGV("startOutput()"); - spaudioPolicyEffects; - { - Mutex::Autolock _l(mLock); - audioPolicyEffects = mAudioPolicyEffects; - } - if (audioPolicyEffects != 0) { - // create audio processors according to stream - status_t status = audioPolicyEffects->addOutputSessionEffects(output, stream, session); - if (status != NO_ERROR && status != ALREADY_EXISTS) { - ALOGW("Failed to add effects on session %d", session); - } - } - Mutex::Autolock _l(mLock); - return mAudioPolicyManager->startOutput(output, stream, session); -} - -status_t AudioPolicyService::stopOutput(audio_io_handle_t output, - audio_stream_type_t stream, - audio_session_t session) -{ - if (uint32_t(stream) >= AUDIO_STREAM_CNT) { - return BAD_VALUE; - } - if (mAudioPolicyManager == NULL) { - return NO_INIT; - } - ALOGV("stopOutput()"); - mOutputCommandThread->stopOutputCommand(output, stream, session); - return NO_ERROR; -} - -status_t AudioPolicyService::doStopOutput(audio_io_handle_t output, - audio_stream_type_t stream, - audio_session_t session) -{ - ALOGV("doStopOutput from tid %d", gettid()); - spaudioPolicyEffects; - { - Mutex::Autolock _l(mLock); - audioPolicyEffects = mAudioPolicyEffects; - } - if (audioPolicyEffects != 0) { - // release audio processors from the stream - status_t status = audioPolicyEffects->releaseOutputSessionEffects(output, stream, session); - if (status != NO_ERROR && status != ALREADY_EXISTS) { - ALOGW("Failed to release effects on session %d", session); - } - } - Mutex::Autolock _l(mLock); - return mAudioPolicyManager->stopOutput(output, stream, session); -} - -void AudioPolicyService::releaseOutput(audio_io_handle_t output, - audio_stream_type_t stream, - audio_session_t session) -{ - if (mAudioPolicyManager == NULL) { - return; - } - ALOGV("releaseOutput()"); - mOutputCommandThread->releaseOutputCommand(output, stream, session); -} - -void AudioPolicyService::doReleaseOutput(audio_io_handle_t output, - audio_stream_type_t stream, - audio_session_t session) -{ - ALOGV("doReleaseOutput from tid %d", gettid()); - Mutex::Autolock _l(mLock); - mAudioPolicyManager->releaseOutput(output, stream, session); -} - -status_t AudioPolicyService::getInputForAttr(const audio_attributes_t *attr, - audio_io_handle_t *input, - audio_session_t session, - uint32_t samplingRate, - audio_format_t format, - audio_channel_mask_t channelMask, - audio_input_flags_t flags) -{ - if (mAudioPolicyManager == NULL) { - return NO_INIT; - } - // already checked by client, but double-check in case the client wrapper is bypassed - if (attr->source >= AUDIO_SOURCE_CNT && attr->source != AUDIO_SOURCE_HOTWORD && - attr->source != AUDIO_SOURCE_FM_TUNER) { - return BAD_VALUE; - } - - if (((attr->source == AUDIO_SOURCE_HOTWORD) && !captureHotwordAllowed()) || - ((attr->source == AUDIO_SOURCE_FM_TUNER) && !captureFmTunerAllowed())) { - return BAD_VALUE; - } - spaudioPolicyEffects; - status_t status; - AudioPolicyInterface::input_type_t inputType; - { - Mutex::Autolock _l(mLock); - // the audio_in_acoustics_t parameter is ignored by get_input() - status = mAudioPolicyManager->getInputForAttr(attr, input, session, - samplingRate, format, channelMask, - flags, &inputType); - audioPolicyEffects = mAudioPolicyEffects; - - if (status == NO_ERROR) { - // enforce permission (if any) required for each type of input - switch (inputType) { - case AudioPolicyInterface::API_INPUT_LEGACY: - break; - case AudioPolicyInterface::API_INPUT_MIX_CAPTURE: - if (!captureAudioOutputAllowed()) { - ALOGE("getInputForAttr() permission denied: capture not allowed"); - status = PERMISSION_DENIED; - } - break; - case AudioPolicyInterface::API_INPUT_MIX_EXT_POLICY_REROUTE: - if (!modifyAudioRoutingAllowed()) { - ALOGE("getInputForAttr() permission denied: modify audio routing not allowed"); - status = PERMISSION_DENIED; - } - break; - case AudioPolicyInterface::API_INPUT_INVALID: - default: - LOG_ALWAYS_FATAL("getInputForAttr() encountered an invalid input type %d", - (int)inputType); - } - } - - if (status != NO_ERROR) { - if (status == PERMISSION_DENIED) { - mAudioPolicyManager->releaseInput(*input, session); - } - return status; - } - } - - if (audioPolicyEffects != 0) { - // create audio pre processors according to input source - status_t status = audioPolicyEffects->addInputEffects(*input, attr->source, session); - if (status != NO_ERROR && status != ALREADY_EXISTS) { - ALOGW("Failed to add effects on input %d", *input); - } - } - return NO_ERROR; -} - -status_t AudioPolicyService::startInput(audio_io_handle_t input, - audio_session_t session) -{ - if (mAudioPolicyManager == NULL) { - return NO_INIT; - } - Mutex::Autolock _l(mLock); - - return mAudioPolicyManager->startInput(input, session); -} - -status_t AudioPolicyService::stopInput(audio_io_handle_t input, - audio_session_t session) -{ - if (mAudioPolicyManager == NULL) { - return NO_INIT; - } - Mutex::Autolock _l(mLock); - - return mAudioPolicyManager->stopInput(input, session); -} - -void AudioPolicyService::releaseInput(audio_io_handle_t input, - audio_session_t session) -{ - if (mAudioPolicyManager == NULL) { - return; - } - spaudioPolicyEffects; - { - Mutex::Autolock _l(mLock); - mAudioPolicyManager->releaseInput(input, session); - audioPolicyEffects = mAudioPolicyEffects; - } - if (audioPolicyEffects != 0) { - // release audio processors from the input - status_t status = audioPolicyEffects->releaseInputEffects(input); - if(status != NO_ERROR) { - ALOGW("Failed to release effects on input %d", input); - } - } -} - -status_t AudioPolicyService::initStreamVolume(audio_stream_type_t stream, - int indexMin, - int indexMax) -{ - if (mAudioPolicyManager == NULL) { - return NO_INIT; - } - if (!settingsAllowed()) { - return PERMISSION_DENIED; - } - if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT) { - return BAD_VALUE; - } - Mutex::Autolock _l(mLock); - mAudioPolicyManager->initStreamVolume(stream, indexMin, indexMax); - return NO_ERROR; -} - -status_t AudioPolicyService::setStreamVolumeIndex(audio_stream_type_t stream, - int index, - audio_devices_t device) -{ - if (mAudioPolicyManager == NULL) { - return NO_INIT; - } - if (!settingsAllowed()) { - return PERMISSION_DENIED; - } - if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT) { - return BAD_VALUE; - } - Mutex::Autolock _l(mLock); - return mAudioPolicyManager->setStreamVolumeIndex(stream, - index, - device); -} - -status_t AudioPolicyService::getStreamVolumeIndex(audio_stream_type_t stream, - int *index, - audio_devices_t device) -{ - if (mAudioPolicyManager == NULL) { - return NO_INIT; - } - if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT) { - return BAD_VALUE; - } - Mutex::Autolock _l(mLock); - return mAudioPolicyManager->getStreamVolumeIndex(stream, - index, - device); -} - -uint32_t AudioPolicyService::getStrategyForStream(audio_stream_type_t stream) -{ - if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT) { - return 0; - } - if (mAudioPolicyManager == NULL) { - return 0; - } - return mAudioPolicyManager->getStrategyForStream(stream); -} - -//audio policy: use audio_device_t appropriately - -audio_devices_t AudioPolicyService::getDevicesForStream(audio_stream_type_t stream) -{ - if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT) { - return AUDIO_DEVICE_NONE; - } - if (mAudioPolicyManager == NULL) { - return AUDIO_DEVICE_NONE; - } - return mAudioPolicyManager->getDevicesForStream(stream); -} - -audio_io_handle_t AudioPolicyService::getOutputForEffect(const effect_descriptor_t *desc) -{ - // FIXME change return type to status_t, and return NO_INIT here - if (mAudioPolicyManager == NULL) { - return 0; - } - Mutex::Autolock _l(mLock); - return mAudioPolicyManager->getOutputForEffect(desc); -} - -status_t AudioPolicyService::registerEffect(const effect_descriptor_t *desc, - audio_io_handle_t io, - uint32_t strategy, - int session, - int id) -{ - if (mAudioPolicyManager == NULL) { - return NO_INIT; - } - return mAudioPolicyManager->registerEffect(desc, io, strategy, session, id); -} - -status_t AudioPolicyService::unregisterEffect(int id) -{ - if (mAudioPolicyManager == NULL) { - return NO_INIT; - } - return mAudioPolicyManager->unregisterEffect(id); -} - -status_t AudioPolicyService::setEffectEnabled(int id, bool enabled) -{ - if (mAudioPolicyManager == NULL) { - return NO_INIT; - } - return mAudioPolicyManager->setEffectEnabled(id, enabled); -} - -bool AudioPolicyService::isStreamActive(audio_stream_type_t stream, uint32_t inPastMs) const -{ - if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT) { - return false; - } - if (mAudioPolicyManager == NULL) { - return false; - } - Mutex::Autolock _l(mLock); - return mAudioPolicyManager->isStreamActive(stream, inPastMs); -} - -bool AudioPolicyService::isStreamActiveRemotely(audio_stream_type_t stream, uint32_t inPastMs) const -{ - if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT) { - return false; - } - if (mAudioPolicyManager == NULL) { - return false; - } - Mutex::Autolock _l(mLock); - return mAudioPolicyManager->isStreamActiveRemotely(stream, inPastMs); -} - -bool AudioPolicyService::isSourceActive(audio_source_t source) const -{ - if (mAudioPolicyManager == NULL) { - return false; - } - Mutex::Autolock _l(mLock); - return mAudioPolicyManager->isSourceActive(source); -} - -status_t AudioPolicyService::queryDefaultPreProcessing(int audioSession, - effect_descriptor_t *descriptors, - uint32_t *count) -{ - if (mAudioPolicyManager == NULL) { - *count = 0; - return NO_INIT; - } - spaudioPolicyEffects; - { - Mutex::Autolock _l(mLock); - audioPolicyEffects = mAudioPolicyEffects; - } - if (audioPolicyEffects == 0) { - *count = 0; - return NO_INIT; - } - return audioPolicyEffects->queryDefaultInputEffects(audioSession, descriptors, count); -} - -bool AudioPolicyService::isOffloadSupported(const audio_offload_info_t& info) -{ - if (mAudioPolicyManager == NULL) { - ALOGV("mAudioPolicyManager == NULL"); - return false; - } - - return mAudioPolicyManager->isOffloadSupported(info); -} - -status_t AudioPolicyService::listAudioPorts(audio_port_role_t role, - audio_port_type_t type, - unsigned int *num_ports, - struct audio_port *ports, - unsigned int *generation) -{ - Mutex::Autolock _l(mLock); - if(!modifyAudioRoutingAllowed()) { - return PERMISSION_DENIED; - } - if (mAudioPolicyManager == NULL) { - return NO_INIT; - } - - return mAudioPolicyManager->listAudioPorts(role, type, num_ports, ports, generation); -} - -status_t AudioPolicyService::getAudioPort(struct audio_port *port) -{ - Mutex::Autolock _l(mLock); - if(!modifyAudioRoutingAllowed()) { - return PERMISSION_DENIED; - } - if (mAudioPolicyManager == NULL) { - return NO_INIT; - } - - return mAudioPolicyManager->getAudioPort(port); -} - -status_t AudioPolicyService::createAudioPatch(const struct audio_patch *patch, - audio_patch_handle_t *handle) -{ - Mutex::Autolock _l(mLock); - if(!modifyAudioRoutingAllowed()) { - return PERMISSION_DENIED; - } - if (mAudioPolicyManager == NULL) { - return NO_INIT; - } - return mAudioPolicyManager->createAudioPatch(patch, handle, - IPCThreadState::self()->getCallingUid()); -} - -status_t AudioPolicyService::releaseAudioPatch(audio_patch_handle_t handle) -{ - Mutex::Autolock _l(mLock); - if(!modifyAudioRoutingAllowed()) { - return PERMISSION_DENIED; - } - if (mAudioPolicyManager == NULL) { - return NO_INIT; - } - - return mAudioPolicyManager->releaseAudioPatch(handle, - IPCThreadState::self()->getCallingUid()); -} - -status_t AudioPolicyService::listAudioPatches(unsigned int *num_patches, - struct audio_patch *patches, - unsigned int *generation) -{ - Mutex::Autolock _l(mLock); - if(!modifyAudioRoutingAllowed()) { - return PERMISSION_DENIED; - } - if (mAudioPolicyManager == NULL) { - return NO_INIT; - } - - return mAudioPolicyManager->listAudioPatches(num_patches, patches, generation); -} - -status_t AudioPolicyService::setAudioPortConfig(const struct audio_port_config *config) -{ - Mutex::Autolock _l(mLock); - if(!modifyAudioRoutingAllowed()) { - return PERMISSION_DENIED; - } - if (mAudioPolicyManager == NULL) { - return NO_INIT; - } - - return mAudioPolicyManager->setAudioPortConfig(config); -} - -status_t AudioPolicyService::acquireSoundTriggerSession(audio_session_t *session, - audio_io_handle_t *ioHandle, - audio_devices_t *device) -{ - if (mAudioPolicyManager == NULL) { - return NO_INIT; - } - - return mAudioPolicyManager->acquireSoundTriggerSession(session, ioHandle, device); -} - -status_t AudioPolicyService::releaseSoundTriggerSession(audio_session_t session) -{ - if (mAudioPolicyManager == NULL) { - return NO_INIT; - } - - return mAudioPolicyManager->releaseSoundTriggerSession(session); -} - -status_t AudioPolicyService::registerPolicyMixes(Vector mixes, bool registration) -{ - Mutex::Autolock _l(mLock); - if(!modifyAudioRoutingAllowed()) { - return PERMISSION_DENIED; - } - if (mAudioPolicyManager == NULL) { - return NO_INIT; - } - if (registration) { - return mAudioPolicyManager->registerPolicyMixes(mixes); - } else { - return mAudioPolicyManager->unregisterPolicyMixes(mixes); - } -} - -}; // namespace android diff --git a/services/audiopolicy/AudioPolicyInterfaceImplLegacy.cpp b/services/audiopolicy/AudioPolicyInterfaceImplLegacy.cpp deleted file mode 100644 index 5a91192..0000000 --- a/services/audiopolicy/AudioPolicyInterfaceImplLegacy.cpp +++ /dev/null @@ -1,607 +0,0 @@ -/* - * Copyright (C) 2009 The Android Open Source Project - * - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - */ - -#define LOG_TAG "AudioPolicyService" -//#define LOG_NDEBUG 0 - -#include -#include "AudioPolicyService.h" -#include "ServiceUtilities.h" - -#include -#include -#include -#include - -namespace android { - - -// ---------------------------------------------------------------------------- - -status_t AudioPolicyService::setDeviceConnectionState(audio_devices_t device, - audio_policy_dev_state_t state, - const char *device_address, - const char *device_name __unused) -{ - if (mpAudioPolicy == NULL) { - return NO_INIT; - } - if (!settingsAllowed()) { - return PERMISSION_DENIED; - } - if (!audio_is_output_device(device) && !audio_is_input_device(device)) { - return BAD_VALUE; - } - if (state != AUDIO_POLICY_DEVICE_STATE_AVAILABLE && - state != AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) { - return BAD_VALUE; - } - - ALOGV("setDeviceConnectionState()"); - Mutex::Autolock _l(mLock); - return mpAudioPolicy->set_device_connection_state(mpAudioPolicy, device, - state, device_address); -} - -audio_policy_dev_state_t AudioPolicyService::getDeviceConnectionState( - audio_devices_t device, - const char *device_address) -{ - if (mpAudioPolicy == NULL) { - return AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE; - } - return mpAudioPolicy->get_device_connection_state(mpAudioPolicy, device, - device_address); -} - -status_t AudioPolicyService::setPhoneState(audio_mode_t state) -{ - if (mpAudioPolicy == NULL) { - return NO_INIT; - } - if (!settingsAllowed()) { - return PERMISSION_DENIED; - } - if (uint32_t(state) >= AUDIO_MODE_CNT) { - return BAD_VALUE; - } - - ALOGV("setPhoneState()"); - - // TODO: check if it is more appropriate to do it in platform specific policy manager - AudioSystem::setMode(state); - - Mutex::Autolock _l(mLock); - mpAudioPolicy->set_phone_state(mpAudioPolicy, state); - mPhoneState = state; - return NO_ERROR; -} - -audio_mode_t AudioPolicyService::getPhoneState() -{ - Mutex::Autolock _l(mLock); - return mPhoneState; -} - -status_t AudioPolicyService::setForceUse(audio_policy_force_use_t usage, - audio_policy_forced_cfg_t config) -{ - if (mpAudioPolicy == NULL) { - return NO_INIT; - } - if (!settingsAllowed()) { - return PERMISSION_DENIED; - } - if (usage < 0 || usage >= AUDIO_POLICY_FORCE_USE_CNT) { - return BAD_VALUE; - } - if (config < 0 || config >= AUDIO_POLICY_FORCE_CFG_CNT) { - return BAD_VALUE; - } - ALOGV("setForceUse()"); - Mutex::Autolock _l(mLock); - mpAudioPolicy->set_force_use(mpAudioPolicy, usage, config); - return NO_ERROR; -} - -audio_policy_forced_cfg_t AudioPolicyService::getForceUse(audio_policy_force_use_t usage) -{ - if (mpAudioPolicy == NULL) { - return AUDIO_POLICY_FORCE_NONE; - } - if (usage < 0 || usage >= AUDIO_POLICY_FORCE_USE_CNT) { - return AUDIO_POLICY_FORCE_NONE; - } - return mpAudioPolicy->get_force_use(mpAudioPolicy, usage); -} - -audio_io_handle_t AudioPolicyService::getOutput(audio_stream_type_t stream, - uint32_t samplingRate, - audio_format_t format, - audio_channel_mask_t channelMask, - audio_output_flags_t flags, - const audio_offload_info_t *offloadInfo) -{ - if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT) { - return AUDIO_IO_HANDLE_NONE; - } - if (mpAudioPolicy == NULL) { - return AUDIO_IO_HANDLE_NONE; - } - ALOGV("getOutput()"); - Mutex::Autolock _l(mLock); - return mpAudioPolicy->get_output(mpAudioPolicy, stream, samplingRate, - format, channelMask, flags, offloadInfo); -} - -status_t AudioPolicyService::startOutput(audio_io_handle_t output, - audio_stream_type_t stream, - audio_session_t session) -{ - if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT) { - return BAD_VALUE; - } - if (mpAudioPolicy == NULL) { - return NO_INIT; - } - ALOGV("startOutput()"); - // create audio processors according to stream - spaudioPolicyEffects; - { - Mutex::Autolock _l(mLock); - audioPolicyEffects = mAudioPolicyEffects; - } - if (audioPolicyEffects != 0) { - status_t status = audioPolicyEffects->addOutputSessionEffects(output, stream, session); - if (status != NO_ERROR && status != ALREADY_EXISTS) { - ALOGW("Failed to add effects on session %d", session); - } - } - - Mutex::Autolock _l(mLock); - return mpAudioPolicy->start_output(mpAudioPolicy, output, stream, session); -} - -status_t AudioPolicyService::stopOutput(audio_io_handle_t output, - audio_stream_type_t stream, - audio_session_t session) -{ - if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT) { - return BAD_VALUE; - } - if (mpAudioPolicy == NULL) { - return NO_INIT; - } - ALOGV("stopOutput()"); - mOutputCommandThread->stopOutputCommand(output, stream, session); - return NO_ERROR; -} - -status_t AudioPolicyService::doStopOutput(audio_io_handle_t output, - audio_stream_type_t stream, - audio_session_t session) -{ - ALOGV("doStopOutput from tid %d", gettid()); - // release audio processors from the stream - spaudioPolicyEffects; - { - Mutex::Autolock _l(mLock); - audioPolicyEffects = mAudioPolicyEffects; - } - if (audioPolicyEffects != 0) { - status_t status = audioPolicyEffects->releaseOutputSessionEffects(output, stream, session); - if (status != NO_ERROR && status != ALREADY_EXISTS) { - ALOGW("Failed to release effects on session %d", session); - } - } - Mutex::Autolock _l(mLock); - return mpAudioPolicy->stop_output(mpAudioPolicy, output, stream, session); -} - -void AudioPolicyService::releaseOutput(audio_io_handle_t output, - audio_stream_type_t stream, - audio_session_t session) -{ - if (mpAudioPolicy == NULL) { - return; - } - ALOGV("releaseOutput()"); - mOutputCommandThread->releaseOutputCommand(output, stream, session); -} - -void AudioPolicyService::doReleaseOutput(audio_io_handle_t output, - audio_stream_type_t stream __unused, - audio_session_t session __unused) -{ - ALOGV("doReleaseOutput from tid %d", gettid()); - Mutex::Autolock _l(mLock); - mpAudioPolicy->release_output(mpAudioPolicy, output); -} - -status_t AudioPolicyService::getInputForAttr(const audio_attributes_t *attr, - audio_io_handle_t *input, - audio_session_t session, - uint32_t samplingRate, - audio_format_t format, - audio_channel_mask_t channelMask, - audio_input_flags_t flags __unused) -{ - if (mpAudioPolicy == NULL) { - return NO_INIT; - } - - audio_source_t inputSource = attr->source; - - // already checked by client, but double-check in case the client wrapper is bypassed - if (inputSource >= AUDIO_SOURCE_CNT && inputSource != AUDIO_SOURCE_HOTWORD && - inputSource != AUDIO_SOURCE_FM_TUNER) { - return BAD_VALUE; - } - - if (inputSource == AUDIO_SOURCE_DEFAULT) { - inputSource = AUDIO_SOURCE_MIC; - } - - if (((inputSource == AUDIO_SOURCE_HOTWORD) && !captureHotwordAllowed()) || - ((inputSource == AUDIO_SOURCE_FM_TUNER) && !captureFmTunerAllowed())) { - return BAD_VALUE; - } - - spaudioPolicyEffects; - { - Mutex::Autolock _l(mLock); - // the audio_in_acoustics_t parameter is ignored by get_input() - *input = mpAudioPolicy->get_input(mpAudioPolicy, inputSource, samplingRate, - format, channelMask, (audio_in_acoustics_t) 0); - audioPolicyEffects = mAudioPolicyEffects; - } - if (*input == AUDIO_IO_HANDLE_NONE) { - return INVALID_OPERATION; - } - - if (audioPolicyEffects != 0) { - // create audio pre processors according to input source - status_t status = audioPolicyEffects->addInputEffects(*input, inputSource, session); - if (status != NO_ERROR && status != ALREADY_EXISTS) { - ALOGW("Failed to add effects on input %d", input); - } - } - return NO_ERROR; -} - -status_t AudioPolicyService::startInput(audio_io_handle_t input, - audio_session_t session __unused) -{ - if (mpAudioPolicy == NULL) { - return NO_INIT; - } - Mutex::Autolock _l(mLock); - - return mpAudioPolicy->start_input(mpAudioPolicy, input); -} - -status_t AudioPolicyService::stopInput(audio_io_handle_t input, - audio_session_t session __unused) -{ - if (mpAudioPolicy == NULL) { - return NO_INIT; - } - Mutex::Autolock _l(mLock); - - return mpAudioPolicy->stop_input(mpAudioPolicy, input); -} - -void AudioPolicyService::releaseInput(audio_io_handle_t input, - audio_session_t session __unused) -{ - if (mpAudioPolicy == NULL) { - return; - } - - spaudioPolicyEffects; - { - Mutex::Autolock _l(mLock); - mpAudioPolicy->release_input(mpAudioPolicy, input); - audioPolicyEffects = mAudioPolicyEffects; - } - if (audioPolicyEffects != 0) { - // release audio processors from the input - status_t status = audioPolicyEffects->releaseInputEffects(input); - if(status != NO_ERROR) { - ALOGW("Failed to release effects on input %d", input); - } - } -} - -status_t AudioPolicyService::initStreamVolume(audio_stream_type_t stream, - int indexMin, - int indexMax) -{ - if (mpAudioPolicy == NULL) { - return NO_INIT; - } - if (!settingsAllowed()) { - return PERMISSION_DENIED; - } - if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT) { - return BAD_VALUE; - } - Mutex::Autolock _l(mLock); - mpAudioPolicy->init_stream_volume(mpAudioPolicy, stream, indexMin, indexMax); - return NO_ERROR; -} - -status_t AudioPolicyService::setStreamVolumeIndex(audio_stream_type_t stream, - int index, - audio_devices_t device) -{ - if (mpAudioPolicy == NULL) { - return NO_INIT; - } - if (!settingsAllowed()) { - return PERMISSION_DENIED; - } - if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT) { - return BAD_VALUE; - } - Mutex::Autolock _l(mLock); - if (mpAudioPolicy->set_stream_volume_index_for_device) { - return mpAudioPolicy->set_stream_volume_index_for_device(mpAudioPolicy, - stream, - index, - device); - } else { - return mpAudioPolicy->set_stream_volume_index(mpAudioPolicy, stream, index); - } -} - -status_t AudioPolicyService::getStreamVolumeIndex(audio_stream_type_t stream, - int *index, - audio_devices_t device) -{ - if (mpAudioPolicy == NULL) { - return NO_INIT; - } - if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT) { - return BAD_VALUE; - } - Mutex::Autolock _l(mLock); - if (mpAudioPolicy->get_stream_volume_index_for_device) { - return mpAudioPolicy->get_stream_volume_index_for_device(mpAudioPolicy, - stream, - index, - device); - } else { - return mpAudioPolicy->get_stream_volume_index(mpAudioPolicy, stream, index); - } -} - -uint32_t AudioPolicyService::getStrategyForStream(audio_stream_type_t stream) -{ - if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT) { - return 0; - } - if (mpAudioPolicy == NULL) { - return 0; - } - return mpAudioPolicy->get_strategy_for_stream(mpAudioPolicy, stream); -} - -//audio policy: use audio_device_t appropriately - -audio_devices_t AudioPolicyService::getDevicesForStream(audio_stream_type_t stream) -{ - if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT) { - return AUDIO_DEVICE_NONE; - } - if (mpAudioPolicy == NULL) { - return AUDIO_DEVICE_NONE; - } - return mpAudioPolicy->get_devices_for_stream(mpAudioPolicy, stream); -} - -audio_io_handle_t AudioPolicyService::getOutputForEffect(const effect_descriptor_t *desc) -{ - // FIXME change return type to status_t, and return NO_INIT here - if (mpAudioPolicy == NULL) { - return 0; - } - Mutex::Autolock _l(mLock); - return mpAudioPolicy->get_output_for_effect(mpAudioPolicy, desc); -} - -status_t AudioPolicyService::registerEffect(const effect_descriptor_t *desc, - audio_io_handle_t io, - uint32_t strategy, - int session, - int id) -{ - if (mpAudioPolicy == NULL) { - return NO_INIT; - } - return mpAudioPolicy->register_effect(mpAudioPolicy, desc, io, strategy, session, id); -} - -status_t AudioPolicyService::unregisterEffect(int id) -{ - if (mpAudioPolicy == NULL) { - return NO_INIT; - } - return mpAudioPolicy->unregister_effect(mpAudioPolicy, id); -} - -status_t AudioPolicyService::setEffectEnabled(int id, bool enabled) -{ - if (mpAudioPolicy == NULL) { - return NO_INIT; - } - return mpAudioPolicy->set_effect_enabled(mpAudioPolicy, id, enabled); -} - -bool AudioPolicyService::isStreamActive(audio_stream_type_t stream, uint32_t inPastMs) const -{ - if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT) { - return false; - } - if (mpAudioPolicy == NULL) { - return false; - } - Mutex::Autolock _l(mLock); - return mpAudioPolicy->is_stream_active(mpAudioPolicy, stream, inPastMs); -} - -bool AudioPolicyService::isStreamActiveRemotely(audio_stream_type_t stream, uint32_t inPastMs) const -{ - if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT) { - return false; - } - if (mpAudioPolicy == NULL) { - return false; - } - Mutex::Autolock _l(mLock); - return mpAudioPolicy->is_stream_active_remotely(mpAudioPolicy, stream, inPastMs); -} - -bool AudioPolicyService::isSourceActive(audio_source_t source) const -{ - if (mpAudioPolicy == NULL) { - return false; - } - if (mpAudioPolicy->is_source_active == 0) { - return false; - } - Mutex::Autolock _l(mLock); - return mpAudioPolicy->is_source_active(mpAudioPolicy, source); -} - -status_t AudioPolicyService::queryDefaultPreProcessing(int audioSession, - effect_descriptor_t *descriptors, - uint32_t *count) -{ - if (mpAudioPolicy == NULL) { - *count = 0; - return NO_INIT; - } - spaudioPolicyEffects; - { - Mutex::Autolock _l(mLock); - audioPolicyEffects = mAudioPolicyEffects; - } - if (audioPolicyEffects == 0) { - *count = 0; - return NO_INIT; - } - return audioPolicyEffects->queryDefaultInputEffects(audioSession, descriptors, count); -} - -bool AudioPolicyService::isOffloadSupported(const audio_offload_info_t& info) -{ - if (mpAudioPolicy == NULL) { - ALOGV("mpAudioPolicy == NULL"); - return false; - } - - if (mpAudioPolicy->is_offload_supported == NULL) { - ALOGV("HAL does not implement is_offload_supported"); - return false; - } - - return mpAudioPolicy->is_offload_supported(mpAudioPolicy, &info); -} - -status_t AudioPolicyService::listAudioPorts(audio_port_role_t role __unused, - audio_port_type_t type __unused, - unsigned int *num_ports, - struct audio_port *ports __unused, - unsigned int *generation __unused) -{ - *num_ports = 0; - return INVALID_OPERATION; -} - -status_t AudioPolicyService::getAudioPort(struct audio_port *port __unused) -{ - return INVALID_OPERATION; -} - -status_t AudioPolicyService::createAudioPatch(const struct audio_patch *patch __unused, - audio_patch_handle_t *handle __unused) -{ - return INVALID_OPERATION; -} - -status_t AudioPolicyService::releaseAudioPatch(audio_patch_handle_t handle __unused) -{ - return INVALID_OPERATION; -} - -status_t AudioPolicyService::listAudioPatches(unsigned int *num_patches, - struct audio_patch *patches __unused, - unsigned int *generation __unused) -{ - *num_patches = 0; - return INVALID_OPERATION; -} - -status_t AudioPolicyService::setAudioPortConfig(const struct audio_port_config *config __unused) -{ - return INVALID_OPERATION; -} - -status_t AudioPolicyService::getOutputForAttr(const audio_attributes_t *attr, - audio_io_handle_t *output, - audio_session_t session __unused, - audio_stream_type_t *stream, - uint32_t samplingRate, - audio_format_t format, - audio_channel_mask_t channelMask, - audio_output_flags_t flags, - const audio_offload_info_t *offloadInfo) -{ - if (attr != NULL) { - *stream = audio_attributes_to_stream_type(attr); - } else { - if (*stream == AUDIO_STREAM_DEFAULT) { - return BAD_VALUE; - } - } - *output = getOutput(*stream, samplingRate, format, channelMask, - flags, offloadInfo); - if (*output == AUDIO_IO_HANDLE_NONE) { - return INVALID_OPERATION; - } - return NO_ERROR; -} - -status_t AudioPolicyService::acquireSoundTriggerSession(audio_session_t *session __unused, - audio_io_handle_t *ioHandle __unused, - audio_devices_t *device __unused) -{ - return INVALID_OPERATION; -} - -status_t AudioPolicyService::releaseSoundTriggerSession(audio_session_t session __unused) -{ - return INVALID_OPERATION; -} - -status_t AudioPolicyService::registerPolicyMixes(Vector mixes __unused, - bool registration __unused) -{ - return INVALID_OPERATION; -} - -}; // namespace android diff --git a/services/audiopolicy/AudioPolicyManager.cpp b/services/audiopolicy/AudioPolicyManager.cpp deleted file mode 100644 index 50ea6ff..0000000 --- a/services/audiopolicy/AudioPolicyManager.cpp +++ /dev/null @@ -1,8108 +0,0 @@ -/* - * Copyright (C) 2009 The Android Open Source Project - * - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - */ - -#define LOG_TAG "AudioPolicyManager" -//#define LOG_NDEBUG 0 - -//#define VERY_VERBOSE_LOGGING -#ifdef VERY_VERBOSE_LOGGING -#define ALOGVV ALOGV -#else -#define ALOGVV(a...) do { } while(0) -#endif - -// A device mask for all audio input devices that are considered "virtual" when evaluating -// active inputs in getActiveInput() -#define APM_AUDIO_IN_DEVICE_VIRTUAL_ALL (AUDIO_DEVICE_IN_REMOTE_SUBMIX|AUDIO_DEVICE_IN_FM_TUNER) -// A device mask for all audio output devices that are considered "remote" when evaluating -// active output devices in isStreamActiveRemotely() -#define APM_AUDIO_OUT_DEVICE_REMOTE_ALL AUDIO_DEVICE_OUT_REMOTE_SUBMIX -// A device mask for all audio input and output devices where matching inputs/outputs on device -// type alone is not enough: the address must match too -#define APM_AUDIO_DEVICE_MATCH_ADDRESS_ALL (AUDIO_DEVICE_IN_REMOTE_SUBMIX | \ - AUDIO_DEVICE_OUT_REMOTE_SUBMIX) - -#include -#include - -#include -#include -#include -#include -#include -#include -#include -#include "AudioPolicyManager.h" -#include "audio_policy_conf.h" - -namespace android { - -// ---------------------------------------------------------------------------- -// Definitions for audio_policy.conf file parsing -// ---------------------------------------------------------------------------- - -struct StringToEnum { - const char *name; - uint32_t value; -}; - -#define STRING_TO_ENUM(string) { #string, string } -#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0])) - -const StringToEnum sDeviceNameToEnumTable[] = { - STRING_TO_ENUM(AUDIO_DEVICE_OUT_EARPIECE), - STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPEAKER), - STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPEAKER_SAFE), - STRING_TO_ENUM(AUDIO_DEVICE_OUT_WIRED_HEADSET), - STRING_TO_ENUM(AUDIO_DEVICE_OUT_WIRED_HEADPHONE), - STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_SCO), - STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET), - STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT), - STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_SCO), - STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_A2DP), - STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES), - STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER), - STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_A2DP), - STRING_TO_ENUM(AUDIO_DEVICE_OUT_AUX_DIGITAL), - STRING_TO_ENUM(AUDIO_DEVICE_OUT_HDMI), - STRING_TO_ENUM(AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET), - STRING_TO_ENUM(AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET), - STRING_TO_ENUM(AUDIO_DEVICE_OUT_USB_ACCESSORY), - STRING_TO_ENUM(AUDIO_DEVICE_OUT_USB_DEVICE), - STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_USB), - STRING_TO_ENUM(AUDIO_DEVICE_OUT_REMOTE_SUBMIX), - STRING_TO_ENUM(AUDIO_DEVICE_OUT_TELEPHONY_TX), - STRING_TO_ENUM(AUDIO_DEVICE_OUT_LINE), - STRING_TO_ENUM(AUDIO_DEVICE_OUT_HDMI_ARC), - STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPDIF), - STRING_TO_ENUM(AUDIO_DEVICE_OUT_FM), - STRING_TO_ENUM(AUDIO_DEVICE_OUT_AUX_LINE), - STRING_TO_ENUM(AUDIO_DEVICE_IN_AMBIENT), - STRING_TO_ENUM(AUDIO_DEVICE_IN_BUILTIN_MIC), - STRING_TO_ENUM(AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET), - STRING_TO_ENUM(AUDIO_DEVICE_IN_ALL_SCO), - STRING_TO_ENUM(AUDIO_DEVICE_IN_WIRED_HEADSET), - STRING_TO_ENUM(AUDIO_DEVICE_IN_AUX_DIGITAL), - STRING_TO_ENUM(AUDIO_DEVICE_IN_HDMI), - STRING_TO_ENUM(AUDIO_DEVICE_IN_TELEPHONY_RX), - STRING_TO_ENUM(AUDIO_DEVICE_IN_VOICE_CALL), - STRING_TO_ENUM(AUDIO_DEVICE_IN_BACK_MIC), - STRING_TO_ENUM(AUDIO_DEVICE_IN_REMOTE_SUBMIX), - STRING_TO_ENUM(AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET), - STRING_TO_ENUM(AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET), - STRING_TO_ENUM(AUDIO_DEVICE_IN_USB_ACCESSORY), - STRING_TO_ENUM(AUDIO_DEVICE_IN_USB_DEVICE), - STRING_TO_ENUM(AUDIO_DEVICE_IN_FM_TUNER), - STRING_TO_ENUM(AUDIO_DEVICE_IN_TV_TUNER), - STRING_TO_ENUM(AUDIO_DEVICE_IN_LINE), - STRING_TO_ENUM(AUDIO_DEVICE_IN_SPDIF), - STRING_TO_ENUM(AUDIO_DEVICE_IN_BLUETOOTH_A2DP), - STRING_TO_ENUM(AUDIO_DEVICE_IN_LOOPBACK), -}; - -const StringToEnum sOutputFlagNameToEnumTable[] = { - STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_DIRECT), - STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_PRIMARY), - STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_FAST), - STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_DEEP_BUFFER), - STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD), - STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_NON_BLOCKING), - STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_HW_AV_SYNC), -}; - -const StringToEnum sInputFlagNameToEnumTable[] = { - STRING_TO_ENUM(AUDIO_INPUT_FLAG_FAST), - STRING_TO_ENUM(AUDIO_INPUT_FLAG_HW_HOTWORD), -}; - -const StringToEnum sFormatNameToEnumTable[] = { - STRING_TO_ENUM(AUDIO_FORMAT_PCM_16_BIT), - STRING_TO_ENUM(AUDIO_FORMAT_PCM_8_BIT), - STRING_TO_ENUM(AUDIO_FORMAT_PCM_32_BIT), - STRING_TO_ENUM(AUDIO_FORMAT_PCM_8_24_BIT), - STRING_TO_ENUM(AUDIO_FORMAT_PCM_FLOAT), - STRING_TO_ENUM(AUDIO_FORMAT_PCM_24_BIT_PACKED), - STRING_TO_ENUM(AUDIO_FORMAT_MP3), - STRING_TO_ENUM(AUDIO_FORMAT_AAC), - STRING_TO_ENUM(AUDIO_FORMAT_AAC_MAIN), - STRING_TO_ENUM(AUDIO_FORMAT_AAC_LC), - STRING_TO_ENUM(AUDIO_FORMAT_AAC_SSR), - STRING_TO_ENUM(AUDIO_FORMAT_AAC_LTP), - STRING_TO_ENUM(AUDIO_FORMAT_AAC_HE_V1), - STRING_TO_ENUM(AUDIO_FORMAT_AAC_SCALABLE), - STRING_TO_ENUM(AUDIO_FORMAT_AAC_ERLC), - STRING_TO_ENUM(AUDIO_FORMAT_AAC_LD), - STRING_TO_ENUM(AUDIO_FORMAT_AAC_HE_V2), - STRING_TO_ENUM(AUDIO_FORMAT_AAC_ELD), - STRING_TO_ENUM(AUDIO_FORMAT_VORBIS), - STRING_TO_ENUM(AUDIO_FORMAT_HE_AAC_V1), - STRING_TO_ENUM(AUDIO_FORMAT_HE_AAC_V2), - STRING_TO_ENUM(AUDIO_FORMAT_OPUS), - STRING_TO_ENUM(AUDIO_FORMAT_AC3), - STRING_TO_ENUM(AUDIO_FORMAT_E_AC3), -}; - -const StringToEnum sOutChannelsNameToEnumTable[] = { - STRING_TO_ENUM(AUDIO_CHANNEL_OUT_MONO), - STRING_TO_ENUM(AUDIO_CHANNEL_OUT_STEREO), - STRING_TO_ENUM(AUDIO_CHANNEL_OUT_QUAD), - STRING_TO_ENUM(AUDIO_CHANNEL_OUT_5POINT1), - STRING_TO_ENUM(AUDIO_CHANNEL_OUT_7POINT1), -}; - -const StringToEnum sInChannelsNameToEnumTable[] = { - STRING_TO_ENUM(AUDIO_CHANNEL_IN_MONO), - STRING_TO_ENUM(AUDIO_CHANNEL_IN_STEREO), - STRING_TO_ENUM(AUDIO_CHANNEL_IN_FRONT_BACK), -}; - -const StringToEnum sGainModeNameToEnumTable[] = { - STRING_TO_ENUM(AUDIO_GAIN_MODE_JOINT), - STRING_TO_ENUM(AUDIO_GAIN_MODE_CHANNELS), - STRING_TO_ENUM(AUDIO_GAIN_MODE_RAMP), -}; - - -uint32_t AudioPolicyManager::stringToEnum(const struct StringToEnum *table, - size_t size, - const char *name) -{ - for (size_t i = 0; i < size; i++) { - if (strcmp(table[i].name, name) == 0) { - ALOGV("stringToEnum() found %s", table[i].name); - return table[i].value; - } - } - return 0; -} - -const char *AudioPolicyManager::enumToString(const struct StringToEnum *table, - size_t size, - uint32_t value) -{ - for (size_t i = 0; i < size; i++) { - if (table[i].value == value) { - return table[i].name; - } - } - return ""; -} - -bool AudioPolicyManager::stringToBool(const char *value) -{ - return ((strcasecmp("true", value) == 0) || (strcmp("1", value) == 0)); -} - - -// ---------------------------------------------------------------------------- -// AudioPolicyInterface implementation -// ---------------------------------------------------------------------------- - -status_t AudioPolicyManager::setDeviceConnectionState(audio_devices_t device, - audio_policy_dev_state_t state, - const char *device_address, - const char *device_name) -{ - return setDeviceConnectionStateInt(device, state, device_address, device_name); -} - -status_t AudioPolicyManager::setDeviceConnectionStateInt(audio_devices_t device, - audio_policy_dev_state_t state, - const char *device_address, - const char *device_name) -{ - ALOGV("setDeviceConnectionStateInt() device: 0x%X, state %d, address %s name %s", -- device, state, device_address, device_name); - - // connect/disconnect only 1 device at a time - if (!audio_is_output_device(device) && !audio_is_input_device(device)) return BAD_VALUE; - - sp devDesc = getDeviceDescriptor(device, device_address, device_name); - - // handle output devices - if (audio_is_output_device(device)) { - SortedVector outputs; - - ssize_t index = mAvailableOutputDevices.indexOf(devDesc); - - // save a copy of the opened output descriptors before any output is opened or closed - // by checkOutputsForDevice(). This will be needed by checkOutputForAllStrategies() - mPreviousOutputs = mOutputs; - switch (state) - { - // handle output device connection - case AUDIO_POLICY_DEVICE_STATE_AVAILABLE: { - if (index >= 0) { - ALOGW("setDeviceConnectionState() device already connected: %x", device); - return INVALID_OPERATION; - } - ALOGV("setDeviceConnectionState() connecting device %x", device); - - // register new device as available - index = mAvailableOutputDevices.add(devDesc); - if (index >= 0) { - sp module = getModuleForDevice(device); - if (module == 0) { - ALOGD("setDeviceConnectionState() could not find HW module for device %08x", - device); - mAvailableOutputDevices.remove(devDesc); - return INVALID_OPERATION; - } - mAvailableOutputDevices[index]->attach(module); - } else { - return NO_MEMORY; - } - - if (checkOutputsForDevice(devDesc, state, outputs, devDesc->mAddress) != NO_ERROR) { - mAvailableOutputDevices.remove(devDesc); - return INVALID_OPERATION; - } - // outputs should never be empty here - ALOG_ASSERT(outputs.size() != 0, "setDeviceConnectionState():" - "checkOutputsForDevice() returned no outputs but status OK"); - ALOGV("setDeviceConnectionState() checkOutputsForDevice() returned %zu outputs", - outputs.size()); - - - // Set connect to HALs - AudioParameter param = AudioParameter(devDesc->mAddress); - param.addInt(String8(AUDIO_PARAMETER_DEVICE_CONNECT), device); - mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString()); - - } break; - // handle output device disconnection - case AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE: { - if (index < 0) { - ALOGW("setDeviceConnectionState() device not connected: %x", device); - return INVALID_OPERATION; - } - - ALOGV("setDeviceConnectionState() disconnecting output device %x", device); - - // Send Disconnect to HALs - AudioParameter param = AudioParameter(devDesc->mAddress); - param.addInt(String8(AUDIO_PARAMETER_DEVICE_DISCONNECT), device); - mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString()); - - // remove device from available output devices - mAvailableOutputDevices.remove(devDesc); - - checkOutputsForDevice(devDesc, state, outputs, devDesc->mAddress); - } break; - - default: - ALOGE("setDeviceConnectionState() invalid state: %x", state); - return BAD_VALUE; - } - - // checkA2dpSuspend must run before checkOutputForAllStrategies so that A2DP - // output is suspended before any tracks are moved to it - checkA2dpSuspend(); - checkOutputForAllStrategies(); - // outputs must be closed after checkOutputForAllStrategies() is executed - if (!outputs.isEmpty()) { - for (size_t i = 0; i < outputs.size(); i++) { - sp desc = mOutputs.valueFor(outputs[i]); - // close unused outputs after device disconnection or direct outputs that have been - // opened by checkOutputsForDevice() to query dynamic parameters - if ((state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) || - (((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) && - (desc->mDirectOpenCount == 0))) { - closeOutput(outputs[i]); - } - } - // check again after closing A2DP output to reset mA2dpSuspended if needed - checkA2dpSuspend(); - } - - updateDevicesAndOutputs(); - if (mPhoneState == AUDIO_MODE_IN_CALL) { - audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/); - updateCallRouting(newDevice); - } - for (size_t i = 0; i < mOutputs.size(); i++) { - audio_io_handle_t output = mOutputs.keyAt(i); - if ((mPhoneState != AUDIO_MODE_IN_CALL) || (output != mPrimaryOutput)) { - audio_devices_t newDevice = getNewOutputDevice(mOutputs.keyAt(i), - true /*fromCache*/); - // do not force device change on duplicated output because if device is 0, it will - // also force a device 0 for the two outputs it is duplicated to which may override - // a valid device selection on those outputs. - bool force = !mOutputs.valueAt(i)->isDuplicated() - && (!deviceDistinguishesOnAddress(device) - // always force when disconnecting (a non-duplicated device) - || (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE)); - setOutputDevice(output, newDevice, force, 0); - } - } - - mpClientInterface->onAudioPortListUpdate(); - return NO_ERROR; - } // end if is output device - - // handle input devices - if (audio_is_input_device(device)) { - SortedVector inputs; - - ssize_t index = mAvailableInputDevices.indexOf(devDesc); - switch (state) - { - // handle input device connection - case AUDIO_POLICY_DEVICE_STATE_AVAILABLE: { - if (index >= 0) { - ALOGW("setDeviceConnectionState() device already connected: %d", device); - return INVALID_OPERATION; - } - sp module = getModuleForDevice(device); - if (module == NULL) { - ALOGW("setDeviceConnectionState(): could not find HW module for device %08x", - device); - return INVALID_OPERATION; - } - if (checkInputsForDevice(device, state, inputs, devDesc->mAddress) != NO_ERROR) { - return INVALID_OPERATION; - } - - index = mAvailableInputDevices.add(devDesc); - if (index >= 0) { - mAvailableInputDevices[index]->attach(module); - } else { - return NO_MEMORY; - } - - // Set connect to HALs - AudioParameter param = AudioParameter(devDesc->mAddress); - param.addInt(String8(AUDIO_PARAMETER_DEVICE_CONNECT), device); - mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString()); - - } break; - - // handle input device disconnection - case AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE: { - if (index < 0) { - ALOGW("setDeviceConnectionState() device not connected: %d", device); - return INVALID_OPERATION; - } - - ALOGV("setDeviceConnectionState() disconnecting input device %x", device); - - // Set Disconnect to HALs - AudioParameter param = AudioParameter(devDesc->mAddress); - param.addInt(String8(AUDIO_PARAMETER_DEVICE_DISCONNECT), device); - mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString()); - - checkInputsForDevice(device, state, inputs, devDesc->mAddress); - mAvailableInputDevices.remove(devDesc); - - } break; - - default: - ALOGE("setDeviceConnectionState() invalid state: %x", state); - return BAD_VALUE; - } - - closeAllInputs(); - - if (mPhoneState == AUDIO_MODE_IN_CALL) { - audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/); - updateCallRouting(newDevice); - } - - mpClientInterface->onAudioPortListUpdate(); - return NO_ERROR; - } // end if is input device - - ALOGW("setDeviceConnectionState() invalid device: %x", device); - return BAD_VALUE; -} - -audio_policy_dev_state_t AudioPolicyManager::getDeviceConnectionState(audio_devices_t device, - const char *device_address) -{ - sp devDesc = getDeviceDescriptor(device, device_address, ""); - DeviceVector *deviceVector; - - if (audio_is_output_device(device)) { - deviceVector = &mAvailableOutputDevices; - } else if (audio_is_input_device(device)) { - deviceVector = &mAvailableInputDevices; - } else { - ALOGW("getDeviceConnectionState() invalid device type %08x", device); - return AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE; - } - - ssize_t index = deviceVector->indexOf(devDesc); - if (index >= 0) { - return AUDIO_POLICY_DEVICE_STATE_AVAILABLE; - } else { - return AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE; - } -} - -sp AudioPolicyManager::getDeviceDescriptor( - const audio_devices_t device, - const char *device_address, - const char *device_name) -{ - String8 address = (device_address == NULL) ? String8("") : String8(device_address); - // handle legacy remote submix case where the address was not always specified - if (deviceDistinguishesOnAddress(device) && (address.length() == 0)) { - address = String8("0"); - } - - for (size_t i = 0; i < mHwModules.size(); i++) { - if (mHwModules[i]->mHandle == 0) { - continue; - } - DeviceVector deviceList = - mHwModules[i]->mDeclaredDevices.getDevicesFromTypeAddr(device, address); - if (!deviceList.isEmpty()) { - return deviceList.itemAt(0); - } - deviceList = mHwModules[i]->mDeclaredDevices.getDevicesFromType(device); - if (!deviceList.isEmpty()) { - return deviceList.itemAt(0); - } - } - - sp devDesc = - new DeviceDescriptor(String8(device_name != NULL ? device_name : ""), device); - devDesc->mAddress = address; - return devDesc; -} - -void AudioPolicyManager::updateCallRouting(audio_devices_t rxDevice, int delayMs) -{ - bool createTxPatch = false; - struct audio_patch patch; - patch.num_sources = 1; - patch.num_sinks = 1; - status_t status; - audio_patch_handle_t afPatchHandle; - DeviceVector deviceList; - - audio_devices_t txDevice = getDeviceAndMixForInputSource(AUDIO_SOURCE_VOICE_COMMUNICATION); - ALOGV("updateCallRouting device rxDevice %08x txDevice %08x", rxDevice, txDevice); - - // release existing RX patch if any - if (mCallRxPatch != 0) { - mpClientInterface->releaseAudioPatch(mCallRxPatch->mAfPatchHandle, 0); - mCallRxPatch.clear(); - } - // release TX patch if any - if (mCallTxPatch != 0) { - mpClientInterface->releaseAudioPatch(mCallTxPatch->mAfPatchHandle, 0); - mCallTxPatch.clear(); - } - - // If the RX device is on the primary HW module, then use legacy routing method for voice calls - // via setOutputDevice() on primary output. - // Otherwise, create two audio patches for TX and RX path. - if (availablePrimaryOutputDevices() & rxDevice) { - setOutputDevice(mPrimaryOutput, rxDevice, true, delayMs); - // If the TX device is also on the primary HW module, setOutputDevice() will take care - // of it due to legacy implementation. If not, create a patch. - if ((availablePrimaryInputDevices() & txDevice & ~AUDIO_DEVICE_BIT_IN) - == AUDIO_DEVICE_NONE) { - createTxPatch = true; - } - } else { - // create RX path audio patch - deviceList = mAvailableOutputDevices.getDevicesFromType(rxDevice); - ALOG_ASSERT(!deviceList.isEmpty(), - "updateCallRouting() selected device not in output device list"); - sp rxSinkDeviceDesc = deviceList.itemAt(0); - deviceList = mAvailableInputDevices.getDevicesFromType(AUDIO_DEVICE_IN_TELEPHONY_RX); - ALOG_ASSERT(!deviceList.isEmpty(), - "updateCallRouting() no telephony RX device"); - sp rxSourceDeviceDesc = deviceList.itemAt(0); - - rxSourceDeviceDesc->toAudioPortConfig(&patch.sources[0]); - rxSinkDeviceDesc->toAudioPortConfig(&patch.sinks[0]); - - // request to reuse existing output stream if one is already opened to reach the RX device - SortedVector outputs = - getOutputsForDevice(rxDevice, mOutputs); - audio_io_handle_t output = selectOutput(outputs, - AUDIO_OUTPUT_FLAG_NONE, - AUDIO_FORMAT_INVALID); - if (output != AUDIO_IO_HANDLE_NONE) { - sp outputDesc = mOutputs.valueFor(output); - ALOG_ASSERT(!outputDesc->isDuplicated(), - "updateCallRouting() RX device output is duplicated"); - outputDesc->toAudioPortConfig(&patch.sources[1]); - patch.num_sources = 2; - } - - afPatchHandle = AUDIO_PATCH_HANDLE_NONE; - status = mpClientInterface->createAudioPatch(&patch, &afPatchHandle, 0); - ALOGW_IF(status != NO_ERROR, "updateCallRouting() error %d creating RX audio patch", - status); - if (status == NO_ERROR) { - mCallRxPatch = new AudioPatch((audio_patch_handle_t)nextUniqueId(), - &patch, mUidCached); - mCallRxPatch->mAfPatchHandle = afPatchHandle; - mCallRxPatch->mUid = mUidCached; - } - createTxPatch = true; - } - if (createTxPatch) { - - struct audio_patch patch; - patch.num_sources = 1; - patch.num_sinks = 1; - deviceList = mAvailableInputDevices.getDevicesFromType(txDevice); - ALOG_ASSERT(!deviceList.isEmpty(), - "updateCallRouting() selected device not in input device list"); - sp txSourceDeviceDesc = deviceList.itemAt(0); - txSourceDeviceDesc->toAudioPortConfig(&patch.sources[0]); - deviceList = mAvailableOutputDevices.getDevicesFromType(AUDIO_DEVICE_OUT_TELEPHONY_TX); - ALOG_ASSERT(!deviceList.isEmpty(), - "updateCallRouting() no telephony TX device"); - sp txSinkDeviceDesc = deviceList.itemAt(0); - txSinkDeviceDesc->toAudioPortConfig(&patch.sinks[0]); - - SortedVector outputs = - getOutputsForDevice(AUDIO_DEVICE_OUT_TELEPHONY_TX, mOutputs); - audio_io_handle_t output = selectOutput(outputs, - AUDIO_OUTPUT_FLAG_NONE, - AUDIO_FORMAT_INVALID); - // request to reuse existing output stream if one is already opened to reach the TX - // path output device - if (output != AUDIO_IO_HANDLE_NONE) { - sp outputDesc = mOutputs.valueFor(output); - ALOG_ASSERT(!outputDesc->isDuplicated(), - "updateCallRouting() RX device output is duplicated"); - outputDesc->toAudioPortConfig(&patch.sources[1]); - patch.num_sources = 2; - } - - afPatchHandle = AUDIO_PATCH_HANDLE_NONE; - status = mpClientInterface->createAudioPatch(&patch, &afPatchHandle, 0); - ALOGW_IF(status != NO_ERROR, "setPhoneState() error %d creating TX audio patch", - status); - if (status == NO_ERROR) { - mCallTxPatch = new AudioPatch((audio_patch_handle_t)nextUniqueId(), - &patch, mUidCached); - mCallTxPatch->mAfPatchHandle = afPatchHandle; - mCallTxPatch->mUid = mUidCached; - } - } -} - -void AudioPolicyManager::setPhoneState(audio_mode_t state) -{ - ALOGV("setPhoneState() state %d", state); - if (state < 0 || state >= AUDIO_MODE_CNT) { - ALOGW("setPhoneState() invalid state %d", state); - return; - } - - if (state == mPhoneState ) { - ALOGW("setPhoneState() setting same state %d", state); - return; - } - - // if leaving call state, handle special case of active streams - // pertaining to sonification strategy see handleIncallSonification() - if (isInCall()) { - ALOGV("setPhoneState() in call state management: new state is %d", state); - for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) { - if (stream == AUDIO_STREAM_PATCH) { - continue; - } - handleIncallSonification((audio_stream_type_t)stream, false, true); - } - - // force reevaluating accessibility routing when call starts - mpClientInterface->invalidateStream(AUDIO_STREAM_ACCESSIBILITY); - } - - // store previous phone state for management of sonification strategy below - int oldState = mPhoneState; - mPhoneState = state; - bool force = false; - - // are we entering or starting a call - if (!isStateInCall(oldState) && isStateInCall(state)) { - ALOGV(" Entering call in setPhoneState()"); - // force routing command to audio hardware when starting a call - // even if no device change is needed - force = true; - for (int j = 0; j < DEVICE_CATEGORY_CNT; j++) { - mStreams[AUDIO_STREAM_DTMF].mVolumeCurve[j] = - sVolumeProfiles[AUDIO_STREAM_VOICE_CALL][j]; - } - } else if (isStateInCall(oldState) && !isStateInCall(state)) { - ALOGV(" Exiting call in setPhoneState()"); - // force routing command to audio hardware when exiting a call - // even if no device change is needed - force = true; - for (int j = 0; j < DEVICE_CATEGORY_CNT; j++) { - mStreams[AUDIO_STREAM_DTMF].mVolumeCurve[j] = - sVolumeProfiles[AUDIO_STREAM_DTMF][j]; - } - } else if (isStateInCall(state) && (state != oldState)) { - ALOGV(" Switching between telephony and VoIP in setPhoneState()"); - // force routing command to audio hardware when switching between telephony and VoIP - // even if no device change is needed - force = true; - } - - // check for device and output changes triggered by new phone state - checkA2dpSuspend(); - checkOutputForAllStrategies(); - updateDevicesAndOutputs(); - - sp hwOutputDesc = mOutputs.valueFor(mPrimaryOutput); - - int delayMs = 0; - if (isStateInCall(state)) { - nsecs_t sysTime = systemTime(); - for (size_t i = 0; i < mOutputs.size(); i++) { - sp desc = mOutputs.valueAt(i); - // mute media and sonification strategies and delay device switch by the largest - // latency of any output where either strategy is active. - // This avoid sending the ring tone or music tail into the earpiece or headset. - if ((desc->isStrategyActive(STRATEGY_MEDIA, - SONIFICATION_HEADSET_MUSIC_DELAY, - sysTime) || - desc->isStrategyActive(STRATEGY_SONIFICATION, - SONIFICATION_HEADSET_MUSIC_DELAY, - sysTime)) && - (delayMs < (int)desc->mLatency*2)) { - delayMs = desc->mLatency*2; - } - setStrategyMute(STRATEGY_MEDIA, true, mOutputs.keyAt(i)); - setStrategyMute(STRATEGY_MEDIA, false, mOutputs.keyAt(i), MUTE_TIME_MS, - getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/)); - setStrategyMute(STRATEGY_SONIFICATION, true, mOutputs.keyAt(i)); - setStrategyMute(STRATEGY_SONIFICATION, false, mOutputs.keyAt(i), MUTE_TIME_MS, - getDeviceForStrategy(STRATEGY_SONIFICATION, true /*fromCache*/)); - } - } - - // Note that despite the fact that getNewOutputDevice() is called on the primary output, - // the device returned is not necessarily reachable via this output - audio_devices_t rxDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/); - // force routing command to audio hardware when ending call - // even if no device change is needed - if (isStateInCall(oldState) && rxDevice == AUDIO_DEVICE_NONE) { - rxDevice = hwOutputDesc->device(); - } - - if (state == AUDIO_MODE_IN_CALL) { - updateCallRouting(rxDevice, delayMs); - } else if (oldState == AUDIO_MODE_IN_CALL) { - if (mCallRxPatch != 0) { - mpClientInterface->releaseAudioPatch(mCallRxPatch->mAfPatchHandle, 0); - mCallRxPatch.clear(); - } - if (mCallTxPatch != 0) { - mpClientInterface->releaseAudioPatch(mCallTxPatch->mAfPatchHandle, 0); - mCallTxPatch.clear(); - } - setOutputDevice(mPrimaryOutput, rxDevice, force, 0); - } else { - setOutputDevice(mPrimaryOutput, rxDevice, force, 0); - } - // if entering in call state, handle special case of active streams - // pertaining to sonification strategy see handleIncallSonification() - if (isStateInCall(state)) { - ALOGV("setPhoneState() in call state management: new state is %d", state); - for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) { - if (stream == AUDIO_STREAM_PATCH) { - continue; - } - handleIncallSonification((audio_stream_type_t)stream, true, true); - } - } - - // Flag that ringtone volume must be limited to music volume until we exit MODE_RINGTONE - if (state == AUDIO_MODE_RINGTONE && - isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY)) { - mLimitRingtoneVolume = true; - } else { - mLimitRingtoneVolume = false; - } -} - -void AudioPolicyManager::setForceUse(audio_policy_force_use_t usage, - audio_policy_forced_cfg_t config) -{ - ALOGV("setForceUse() usage %d, config %d, mPhoneState %d", usage, config, mPhoneState); - - bool forceVolumeReeval = false; - switch(usage) { - case AUDIO_POLICY_FORCE_FOR_COMMUNICATION: - if (config != AUDIO_POLICY_FORCE_SPEAKER && config != AUDIO_POLICY_FORCE_BT_SCO && - config != AUDIO_POLICY_FORCE_NONE) { - ALOGW("setForceUse() invalid config %d for FOR_COMMUNICATION", config); - return; - } - forceVolumeReeval = true; - mForceUse[usage] = config; - break; - case AUDIO_POLICY_FORCE_FOR_MEDIA: - if (config != AUDIO_POLICY_FORCE_HEADPHONES && config != AUDIO_POLICY_FORCE_BT_A2DP && - config != AUDIO_POLICY_FORCE_WIRED_ACCESSORY && - config != AUDIO_POLICY_FORCE_ANALOG_DOCK && - config != AUDIO_POLICY_FORCE_DIGITAL_DOCK && config != AUDIO_POLICY_FORCE_NONE && - config != AUDIO_POLICY_FORCE_NO_BT_A2DP && config != AUDIO_POLICY_FORCE_SPEAKER ) { - ALOGW("setForceUse() invalid config %d for FOR_MEDIA", config); - return; - } - mForceUse[usage] = config; - break; - case AUDIO_POLICY_FORCE_FOR_RECORD: - if (config != AUDIO_POLICY_FORCE_BT_SCO && config != AUDIO_POLICY_FORCE_WIRED_ACCESSORY && - config != AUDIO_POLICY_FORCE_NONE) { - ALOGW("setForceUse() invalid config %d for FOR_RECORD", config); - return; - } - mForceUse[usage] = config; - break; - case AUDIO_POLICY_FORCE_FOR_DOCK: - if (config != AUDIO_POLICY_FORCE_NONE && config != AUDIO_POLICY_FORCE_BT_CAR_DOCK && - config != AUDIO_POLICY_FORCE_BT_DESK_DOCK && - config != AUDIO_POLICY_FORCE_WIRED_ACCESSORY && - config != AUDIO_POLICY_FORCE_ANALOG_DOCK && - config != AUDIO_POLICY_FORCE_DIGITAL_DOCK) { - ALOGW("setForceUse() invalid config %d for FOR_DOCK", config); - } - forceVolumeReeval = true; - mForceUse[usage] = config; - break; - case AUDIO_POLICY_FORCE_FOR_SYSTEM: - if (config != AUDIO_POLICY_FORCE_NONE && - config != AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) { - ALOGW("setForceUse() invalid config %d for FOR_SYSTEM", config); - } - forceVolumeReeval = true; - mForceUse[usage] = config; - break; - case AUDIO_POLICY_FORCE_FOR_HDMI_SYSTEM_AUDIO: - if (config != AUDIO_POLICY_FORCE_NONE && - config != AUDIO_POLICY_FORCE_HDMI_SYSTEM_AUDIO_ENFORCED) { - ALOGW("setForceUse() invalid config %d forHDMI_SYSTEM_AUDIO", config); - } - mForceUse[usage] = config; - break; - default: - ALOGW("setForceUse() invalid usage %d", usage); - break; - } - - // check for device and output changes triggered by new force usage - checkA2dpSuspend(); - checkOutputForAllStrategies(); - updateDevicesAndOutputs(); - if (mPhoneState == AUDIO_MODE_IN_CALL) { - audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, true /*fromCache*/); - updateCallRouting(newDevice); - } - for (size_t i = 0; i < mOutputs.size(); i++) { - audio_io_handle_t output = mOutputs.keyAt(i); - audio_devices_t newDevice = getNewOutputDevice(output, true /*fromCache*/); - if ((mPhoneState != AUDIO_MODE_IN_CALL) || (output != mPrimaryOutput)) { - setOutputDevice(output, newDevice, (newDevice != AUDIO_DEVICE_NONE)); - } - if (forceVolumeReeval && (newDevice != AUDIO_DEVICE_NONE)) { - applyStreamVolumes(output, newDevice, 0, true); - } - } - - audio_io_handle_t activeInput = getActiveInput(); - if (activeInput != 0) { - setInputDevice(activeInput, getNewInputDevice(activeInput)); - } - -} - -audio_policy_forced_cfg_t AudioPolicyManager::getForceUse(audio_policy_force_use_t usage) -{ - return mForceUse[usage]; -} - -void AudioPolicyManager::setSystemProperty(const char* property, const char* value) -{ - ALOGV("setSystemProperty() property %s, value %s", property, value); -} - -// Find a direct output profile compatible with the parameters passed, even if the input flags do -// not explicitly request a direct output -sp AudioPolicyManager::getProfileForDirectOutput( - audio_devices_t device, - uint32_t samplingRate, - audio_format_t format, - audio_channel_mask_t channelMask, - audio_output_flags_t flags) -{ - for (size_t i = 0; i < mHwModules.size(); i++) { - if (mHwModules[i]->mHandle == 0) { - continue; - } - for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++) { - sp profile = mHwModules[i]->mOutputProfiles[j]; - bool found = profile->isCompatibleProfile(device, String8(""), samplingRate, - NULL /*updatedSamplingRate*/, format, channelMask, - flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD ? - AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD : AUDIO_OUTPUT_FLAG_DIRECT); - if (found && (mAvailableOutputDevices.types() & profile->mSupportedDevices.types())) { - return profile; - } - } - } - return 0; -} - -audio_io_handle_t AudioPolicyManager::getOutput(audio_stream_type_t stream, - uint32_t samplingRate, - audio_format_t format, - audio_channel_mask_t channelMask, - audio_output_flags_t flags, - const audio_offload_info_t *offloadInfo) -{ - routing_strategy strategy = getStrategy(stream); - audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/); - ALOGV("getOutput() device %d, stream %d, samplingRate %d, format %x, channelMask %x, flags %x", - device, stream, samplingRate, format, channelMask, flags); - - return getOutputForDevice(device, AUDIO_SESSION_ALLOCATE, - stream, samplingRate,format, channelMask, - flags, offloadInfo); -} - -status_t AudioPolicyManager::getOutputForAttr(const audio_attributes_t *attr, - audio_io_handle_t *output, - audio_session_t session, - audio_stream_type_t *stream, - uint32_t samplingRate, - audio_format_t format, - audio_channel_mask_t channelMask, - audio_output_flags_t flags, - const audio_offload_info_t *offloadInfo) -{ - audio_attributes_t attributes; - if (attr != NULL) { - if (!isValidAttributes(attr)) { - ALOGE("getOutputForAttr() invalid attributes: usage=%d content=%d flags=0x%x tags=[%s]", - attr->usage, attr->content_type, attr->flags, - attr->tags); - return BAD_VALUE; - } - attributes = *attr; - } else { - if (*stream < AUDIO_STREAM_MIN || *stream >= AUDIO_STREAM_PUBLIC_CNT) { - ALOGE("getOutputForAttr(): invalid stream type"); - return BAD_VALUE; - } - stream_type_to_audio_attributes(*stream, &attributes); - } - - for (size_t i = 0; i < mPolicyMixes.size(); i++) { - sp desc; - if (mPolicyMixes[i]->mMix.mMixType == MIX_TYPE_PLAYERS) { - for (size_t j = 0; j < mPolicyMixes[i]->mMix.mCriteria.size(); j++) { - if ((RULE_MATCH_ATTRIBUTE_USAGE == mPolicyMixes[i]->mMix.mCriteria[j].mRule && - mPolicyMixes[i]->mMix.mCriteria[j].mAttr.mUsage == attributes.usage) || - (RULE_EXCLUDE_ATTRIBUTE_USAGE == mPolicyMixes[i]->mMix.mCriteria[j].mRule && - mPolicyMixes[i]->mMix.mCriteria[j].mAttr.mUsage != attributes.usage)) { - desc = mPolicyMixes[i]->mOutput; - break; - } - if (strncmp(attributes.tags, "addr=", strlen("addr=")) == 0 && - strncmp(attributes.tags + strlen("addr="), - mPolicyMixes[i]->mMix.mRegistrationId.string(), - AUDIO_ATTRIBUTES_TAGS_MAX_SIZE - strlen("addr=") - 1) == 0) { - desc = mPolicyMixes[i]->mOutput; - break; - } - } - } else if (mPolicyMixes[i]->mMix.mMixType == MIX_TYPE_RECORDERS) { - if (attributes.usage == AUDIO_USAGE_VIRTUAL_SOURCE && - strncmp(attributes.tags, "addr=", strlen("addr=")) == 0 && - strncmp(attributes.tags + strlen("addr="), - mPolicyMixes[i]->mMix.mRegistrationId.string(), - AUDIO_ATTRIBUTES_TAGS_MAX_SIZE - strlen("addr=") - 1) == 0) { - desc = mPolicyMixes[i]->mOutput; - } - } - if (desc != 0) { - if (!audio_is_linear_pcm(format)) { - return BAD_VALUE; - } - desc->mPolicyMix = &mPolicyMixes[i]->mMix; - *stream = streamTypefromAttributesInt(&attributes); - *output = desc->mIoHandle; - ALOGV("getOutputForAttr() returns output %d", *output); - return NO_ERROR; - } - } - if (attributes.usage == AUDIO_USAGE_VIRTUAL_SOURCE) { - ALOGW("getOutputForAttr() no policy mix found for usage AUDIO_USAGE_VIRTUAL_SOURCE"); - return BAD_VALUE; - } - - ALOGV("getOutputForAttr() usage=%d, content=%d, tag=%s flags=%08x", - attributes.usage, attributes.content_type, attributes.tags, attributes.flags); - - routing_strategy strategy = (routing_strategy) getStrategyForAttr(&attributes); - audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/); - - if ((attributes.flags & AUDIO_FLAG_HW_AV_SYNC) != 0) { - flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_HW_AV_SYNC); - } - - ALOGV("getOutputForAttr() device 0x%x, samplingRate %d, format %x, channelMask %x, flags %x", - device, samplingRate, format, channelMask, flags); - - *stream = streamTypefromAttributesInt(&attributes); - *output = getOutputForDevice(device, session, *stream, - samplingRate, format, channelMask, - flags, offloadInfo); - if (*output == AUDIO_IO_HANDLE_NONE) { - return INVALID_OPERATION; - } - return NO_ERROR; -} - -audio_io_handle_t AudioPolicyManager::getOutputForDevice( - audio_devices_t device, - audio_session_t session __unused, - audio_stream_type_t stream, - uint32_t samplingRate, - audio_format_t format, - audio_channel_mask_t channelMask, - audio_output_flags_t flags, - const audio_offload_info_t *offloadInfo) -{ - audio_io_handle_t output = AUDIO_IO_HANDLE_NONE; - uint32_t latency = 0; - status_t status; - -#ifdef AUDIO_POLICY_TEST - if (mCurOutput != 0) { - ALOGV("getOutput() test output mCurOutput %d, samplingRate %d, format %d, channelMask %x, mDirectOutput %d", - mCurOutput, mTestSamplingRate, mTestFormat, mTestChannels, mDirectOutput); - - if (mTestOutputs[mCurOutput] == 0) { - ALOGV("getOutput() opening test output"); - sp outputDesc = new AudioOutputDescriptor(NULL); - outputDesc->mDevice = mTestDevice; - outputDesc->mLatency = mTestLatencyMs; - outputDesc->mFlags = - (audio_output_flags_t)(mDirectOutput ? AUDIO_OUTPUT_FLAG_DIRECT : 0); - outputDesc->mRefCount[stream] = 0; - audio_config_t config = AUDIO_CONFIG_INITIALIZER; - config.sample_rate = mTestSamplingRate; - config.channel_mask = mTestChannels; - config.format = mTestFormat; - if (offloadInfo != NULL) { - config.offload_info = *offloadInfo; - } - status = mpClientInterface->openOutput(0, - &mTestOutputs[mCurOutput], - &config, - &outputDesc->mDevice, - String8(""), - &outputDesc->mLatency, - outputDesc->mFlags); - if (status == NO_ERROR) { - outputDesc->mSamplingRate = config.sample_rate; - outputDesc->mFormat = config.format; - outputDesc->mChannelMask = config.channel_mask; - AudioParameter outputCmd = AudioParameter(); - outputCmd.addInt(String8("set_id"),mCurOutput); - mpClientInterface->setParameters(mTestOutputs[mCurOutput],outputCmd.toString()); - addOutput(mTestOutputs[mCurOutput], outputDesc); - } - } - return mTestOutputs[mCurOutput]; - } -#endif //AUDIO_POLICY_TEST - - // open a direct output if required by specified parameters - //force direct flag if offload flag is set: offloading implies a direct output stream - // and all common behaviors are driven by checking only the direct flag - // this should normally be set appropriately in the policy configuration file - if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) { - flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT); - } - if ((flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) != 0) { - flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT); - } - // only allow deep buffering for music stream type - if (stream != AUDIO_STREAM_MUSIC) { - flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER); - } - - sp profile; - - // skip direct output selection if the request can obviously be attached to a mixed output - // and not explicitly requested - if (((flags & AUDIO_OUTPUT_FLAG_DIRECT) == 0) && - audio_is_linear_pcm(format) && samplingRate <= MAX_MIXER_SAMPLING_RATE && - audio_channel_count_from_out_mask(channelMask) <= 2) { - goto non_direct_output; - } - - // Do not allow offloading if one non offloadable effect is enabled. This prevents from - // creating an offloaded track and tearing it down immediately after start when audioflinger - // detects there is an active non offloadable effect. - // FIXME: We should check the audio session here but we do not have it in this context. - // This may prevent offloading in rare situations where effects are left active by apps - // in the background. - - if (((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) == 0) || - !isNonOffloadableEffectEnabled()) { - profile = getProfileForDirectOutput(device, - samplingRate, - format, - channelMask, - (audio_output_flags_t)flags); - } - - if (profile != 0) { - sp outputDesc = NULL; - - for (size_t i = 0; i < mOutputs.size(); i++) { - sp desc = mOutputs.valueAt(i); - if (!desc->isDuplicated() && (profile == desc->mProfile)) { - outputDesc = desc; - // reuse direct output if currently open and configured with same parameters - if ((samplingRate == outputDesc->mSamplingRate) && - (format == outputDesc->mFormat) && - (channelMask == outputDesc->mChannelMask)) { - outputDesc->mDirectOpenCount++; - ALOGV("getOutput() reusing direct output %d", mOutputs.keyAt(i)); - return mOutputs.keyAt(i); - } - } - } - // close direct output if currently open and configured with different parameters - if (outputDesc != NULL) { - closeOutput(outputDesc->mIoHandle); - } - outputDesc = new AudioOutputDescriptor(profile); - outputDesc->mDevice = device; - outputDesc->mLatency = 0; - outputDesc->mFlags =(audio_output_flags_t) (outputDesc->mFlags | flags); - audio_config_t config = AUDIO_CONFIG_INITIALIZER; - config.sample_rate = samplingRate; - config.channel_mask = channelMask; - config.format = format; - if (offloadInfo != NULL) { - config.offload_info = *offloadInfo; - } - status = mpClientInterface->openOutput(profile->mModule->mHandle, - &output, - &config, - &outputDesc->mDevice, - String8(""), - &outputDesc->mLatency, - outputDesc->mFlags); - - // only accept an output with the requested parameters - if (status != NO_ERROR || - (samplingRate != 0 && samplingRate != config.sample_rate) || - (format != AUDIO_FORMAT_DEFAULT && format != config.format) || - (channelMask != 0 && channelMask != config.channel_mask)) { - ALOGV("getOutput() failed opening direct output: output %d samplingRate %d %d," - "format %d %d, channelMask %04x %04x", output, samplingRate, - outputDesc->mSamplingRate, format, outputDesc->mFormat, channelMask, - outputDesc->mChannelMask); - if (output != AUDIO_IO_HANDLE_NONE) { - mpClientInterface->closeOutput(output); - } - // fall back to mixer output if possible when the direct output could not be open - if (audio_is_linear_pcm(format) && samplingRate <= MAX_MIXER_SAMPLING_RATE) { - goto non_direct_output; - } - return AUDIO_IO_HANDLE_NONE; - } - outputDesc->mSamplingRate = config.sample_rate; - outputDesc->mChannelMask = config.channel_mask; - outputDesc->mFormat = config.format; - outputDesc->mRefCount[stream] = 0; - outputDesc->mStopTime[stream] = 0; - outputDesc->mDirectOpenCount = 1; - - audio_io_handle_t srcOutput = getOutputForEffect(); - addOutput(output, outputDesc); - audio_io_handle_t dstOutput = getOutputForEffect(); - if (dstOutput == output) { - mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, srcOutput, dstOutput); - } - mPreviousOutputs = mOutputs; - ALOGV("getOutput() returns new direct output %d", output); - mpClientInterface->onAudioPortListUpdate(); - return output; - } - -non_direct_output: - - // ignoring channel mask due to downmix capability in mixer - - // open a non direct output - - // for non direct outputs, only PCM is supported - if (audio_is_linear_pcm(format)) { - // get which output is suitable for the specified stream. The actual - // routing change will happen when startOutput() will be called - SortedVector outputs = getOutputsForDevice(device, mOutputs); - - // at this stage we should ignore the DIRECT flag as no direct output could be found earlier - flags = (audio_output_flags_t)(flags & ~AUDIO_OUTPUT_FLAG_DIRECT); - output = selectOutput(outputs, flags, format); - } - ALOGW_IF((output == 0), "getOutput() could not find output for stream %d, samplingRate %d," - "format %d, channels %x, flags %x", stream, samplingRate, format, channelMask, flags); - - ALOGV("getOutput() returns output %d", output); - - return output; -} - -audio_io_handle_t AudioPolicyManager::selectOutput(const SortedVector& outputs, - audio_output_flags_t flags, - audio_format_t format) -{ - // select one output among several that provide a path to a particular device or set of - // devices (the list was previously build by getOutputsForDevice()). - // The priority is as follows: - // 1: the output with the highest number of requested policy flags - // 2: the primary output - // 3: the first output in the list - - if (outputs.size() == 0) { - return 0; - } - if (outputs.size() == 1) { - return outputs[0]; - } - - int maxCommonFlags = 0; - audio_io_handle_t outputFlags = 0; - audio_io_handle_t outputPrimary = 0; - - for (size_t i = 0; i < outputs.size(); i++) { - sp outputDesc = mOutputs.valueFor(outputs[i]); - if (!outputDesc->isDuplicated()) { - // if a valid format is specified, skip output if not compatible - if (format != AUDIO_FORMAT_INVALID) { - if (outputDesc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) { - if (format != outputDesc->mFormat) { - continue; - } - } else if (!audio_is_linear_pcm(format)) { - continue; - } - } - - int commonFlags = popcount(outputDesc->mProfile->mFlags & flags); - if (commonFlags > maxCommonFlags) { - outputFlags = outputs[i]; - maxCommonFlags = commonFlags; - ALOGV("selectOutput() commonFlags for output %d, %04x", outputs[i], commonFlags); - } - if (outputDesc->mProfile->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY) { - outputPrimary = outputs[i]; - } - } - } - - if (outputFlags != 0) { - return outputFlags; - } - if (outputPrimary != 0) { - return outputPrimary; - } - - return outputs[0]; -} - -status_t AudioPolicyManager::startOutput(audio_io_handle_t output, - audio_stream_type_t stream, - audio_session_t session) -{ - ALOGV("startOutput() output %d, stream %d, session %d", output, stream, session); - ssize_t index = mOutputs.indexOfKey(output); - if (index < 0) { - ALOGW("startOutput() unknown output %d", output); - return BAD_VALUE; - } - - // cannot start playback of STREAM_TTS if any other output is being used - uint32_t beaconMuteLatency = 0; - if (stream == AUDIO_STREAM_TTS) { - ALOGV("\t found BEACON stream"); - if (isAnyOutputActive(AUDIO_STREAM_TTS /*streamToIgnore*/)) { - return INVALID_OPERATION; - } else { - beaconMuteLatency = handleEventForBeacon(STARTING_BEACON); - } - } else { - // some playback other than beacon starts - beaconMuteLatency = handleEventForBeacon(STARTING_OUTPUT); - } - - sp outputDesc = mOutputs.valueAt(index); - - // increment usage count for this stream on the requested output: - // NOTE that the usage count is the same for duplicated output and hardware output which is - // necessary for a correct control of hardware output routing by startOutput() and stopOutput() - outputDesc->changeRefCount(stream, 1); - - if (outputDesc->mRefCount[stream] == 1) { - // starting an output being rerouted? - audio_devices_t newDevice; - if (outputDesc->mPolicyMix != NULL) { - newDevice = AUDIO_DEVICE_OUT_REMOTE_SUBMIX; - } else { - newDevice = getNewOutputDevice(output, false /*fromCache*/); - } - routing_strategy strategy = getStrategy(stream); - bool shouldWait = (strategy == STRATEGY_SONIFICATION) || - (strategy == STRATEGY_SONIFICATION_RESPECTFUL) || - (beaconMuteLatency > 0); - uint32_t waitMs = beaconMuteLatency; - bool force = false; - for (size_t i = 0; i < mOutputs.size(); i++) { - sp desc = mOutputs.valueAt(i); - if (desc != outputDesc) { - // force a device change if any other output is managed by the same hw - // module and has a current device selection that differs from selected device. - // In this case, the audio HAL must receive the new device selection so that it can - // change the device currently selected by the other active output. - if (outputDesc->sharesHwModuleWith(desc) && - desc->device() != newDevice) { - force = true; - } - // wait for audio on other active outputs to be presented when starting - // a notification so that audio focus effect can propagate, or that a mute/unmute - // event occurred for beacon - uint32_t latency = desc->latency(); - if (shouldWait && desc->isActive(latency * 2) && (waitMs < latency)) { - waitMs = latency; - } - } - } - uint32_t muteWaitMs = setOutputDevice(output, newDevice, force); - - // handle special case for sonification while in call - if (isInCall()) { - handleIncallSonification(stream, true, false); - } - - // apply volume rules for current stream and device if necessary - checkAndSetVolume(stream, - mStreams[stream].getVolumeIndex(newDevice), - output, - newDevice); - - // update the outputs if starting an output with a stream that can affect notification - // routing - handleNotificationRoutingForStream(stream); - - // Automatically enable the remote submix input when output is started on a re routing mix - // of type MIX_TYPE_RECORDERS - if (audio_is_remote_submix_device(newDevice) && outputDesc->mPolicyMix != NULL && - outputDesc->mPolicyMix->mMixType == MIX_TYPE_RECORDERS) { - setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX, - AUDIO_POLICY_DEVICE_STATE_AVAILABLE, - outputDesc->mPolicyMix->mRegistrationId, - "remote-submix"); - } - - // force reevaluating accessibility routing when ringtone or alarm starts - if (strategy == STRATEGY_SONIFICATION) { - mpClientInterface->invalidateStream(AUDIO_STREAM_ACCESSIBILITY); - } - - if (waitMs > muteWaitMs) { - usleep((waitMs - muteWaitMs) * 2 * 1000); - } - } - return NO_ERROR; -} - - -status_t AudioPolicyManager::stopOutput(audio_io_handle_t output, - audio_stream_type_t stream, - audio_session_t session) -{ - ALOGV("stopOutput() output %d, stream %d, session %d", output, stream, session); - ssize_t index = mOutputs.indexOfKey(output); - if (index < 0) { - ALOGW("stopOutput() unknown output %d", output); - return BAD_VALUE; - } - - sp outputDesc = mOutputs.valueAt(index); - - // always handle stream stop, check which stream type is stopping - handleEventForBeacon(stream == AUDIO_STREAM_TTS ? STOPPING_BEACON : STOPPING_OUTPUT); - - // handle special case for sonification while in call - if (isInCall()) { - handleIncallSonification(stream, false, false); - } - - if (outputDesc->mRefCount[stream] > 0) { - // decrement usage count of this stream on the output - outputDesc->changeRefCount(stream, -1); - // store time at which the stream was stopped - see isStreamActive() - if (outputDesc->mRefCount[stream] == 0) { - // Automatically disable the remote submix input when output is stopped on a - // re routing mix of type MIX_TYPE_RECORDERS - if (audio_is_remote_submix_device(outputDesc->mDevice) && - outputDesc->mPolicyMix != NULL && - outputDesc->mPolicyMix->mMixType == MIX_TYPE_RECORDERS) { - setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX, - AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE, - outputDesc->mPolicyMix->mRegistrationId, - "remote-submix"); - } - - outputDesc->mStopTime[stream] = systemTime(); - audio_devices_t newDevice = getNewOutputDevice(output, false /*fromCache*/); - // delay the device switch by twice the latency because stopOutput() is executed when - // the track stop() command is received and at that time the audio track buffer can - // still contain data that needs to be drained. The latency only covers the audio HAL - // and kernel buffers. Also the latency does not always include additional delay in the - // audio path (audio DSP, CODEC ...) - setOutputDevice(output, newDevice, false, outputDesc->mLatency*2); - - // force restoring the device selection on other active outputs if it differs from the - // one being selected for this output - for (size_t i = 0; i < mOutputs.size(); i++) { - audio_io_handle_t curOutput = mOutputs.keyAt(i); - sp desc = mOutputs.valueAt(i); - if (curOutput != output && - desc->isActive() && - outputDesc->sharesHwModuleWith(desc) && - (newDevice != desc->device())) { - setOutputDevice(curOutput, - getNewOutputDevice(curOutput, false /*fromCache*/), - true, - outputDesc->mLatency*2); - } - } - // update the outputs if stopping one with a stream that can affect notification routing - handleNotificationRoutingForStream(stream); - } - return NO_ERROR; - } else { - ALOGW("stopOutput() refcount is already 0 for output %d", output); - return INVALID_OPERATION; - } -} - -void AudioPolicyManager::releaseOutput(audio_io_handle_t output, - audio_stream_type_t stream __unused, - audio_session_t session __unused) -{ - ALOGV("releaseOutput() %d", output); - ssize_t index = mOutputs.indexOfKey(output); - if (index < 0) { - ALOGW("releaseOutput() releasing unknown output %d", output); - return; - } - -#ifdef AUDIO_POLICY_TEST - int testIndex = testOutputIndex(output); - if (testIndex != 0) { - sp outputDesc = mOutputs.valueAt(index); - if (outputDesc->isActive()) { - mpClientInterface->closeOutput(output); - mOutputs.removeItem(output); - mTestOutputs[testIndex] = 0; - } - return; - } -#endif //AUDIO_POLICY_TEST - - sp desc = mOutputs.valueAt(index); - if (desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) { - if (desc->mDirectOpenCount <= 0) { - ALOGW("releaseOutput() invalid open count %d for output %d", - desc->mDirectOpenCount, output); - return; - } - if (--desc->mDirectOpenCount == 0) { - closeOutput(output); - // If effects where present on the output, audioflinger moved them to the primary - // output by default: move them back to the appropriate output. - audio_io_handle_t dstOutput = getOutputForEffect(); - if (dstOutput != mPrimaryOutput) { - mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, mPrimaryOutput, dstOutput); - } - mpClientInterface->onAudioPortListUpdate(); - } - } -} - - -status_t AudioPolicyManager::getInputForAttr(const audio_attributes_t *attr, - audio_io_handle_t *input, - audio_session_t session, - uint32_t samplingRate, - audio_format_t format, - audio_channel_mask_t channelMask, - audio_input_flags_t flags, - input_type_t *inputType) -{ - ALOGV("getInputForAttr() source %d, samplingRate %d, format %d, channelMask %x," - "session %d, flags %#x", - attr->source, samplingRate, format, channelMask, session, flags); - - *input = AUDIO_IO_HANDLE_NONE; - *inputType = API_INPUT_INVALID; - audio_devices_t device; - // handle legacy remote submix case where the address was not always specified - String8 address = String8(""); - bool isSoundTrigger = false; - audio_source_t inputSource = attr->source; - audio_source_t halInputSource; - AudioMix *policyMix = NULL; - - if (inputSource == AUDIO_SOURCE_DEFAULT) { - inputSource = AUDIO_SOURCE_MIC; - } - halInputSource = inputSource; - - if (inputSource == AUDIO_SOURCE_REMOTE_SUBMIX && - strncmp(attr->tags, "addr=", strlen("addr=")) == 0) { - device = AUDIO_DEVICE_IN_REMOTE_SUBMIX; - address = String8(attr->tags + strlen("addr=")); - ssize_t index = mPolicyMixes.indexOfKey(address); - if (index < 0) { - ALOGW("getInputForAttr() no policy for address %s", address.string()); - return BAD_VALUE; - } - if (mPolicyMixes[index]->mMix.mMixType != MIX_TYPE_PLAYERS) { - ALOGW("getInputForAttr() bad policy mix type for address %s", address.string()); - return BAD_VALUE; - } - policyMix = &mPolicyMixes[index]->mMix; - *inputType = API_INPUT_MIX_EXT_POLICY_REROUTE; - } else { - device = getDeviceAndMixForInputSource(inputSource, &policyMix); - if (device == AUDIO_DEVICE_NONE) { - ALOGW("getInputForAttr() could not find device for source %d", inputSource); - return BAD_VALUE; - } - if (policyMix != NULL) { - address = policyMix->mRegistrationId; - if (policyMix->mMixType == MIX_TYPE_RECORDERS) { - // there is an external policy, but this input is attached to a mix of recorders, - // meaning it receives audio injected into the framework, so the recorder doesn't - // know about it and is therefore considered "legacy" - *inputType = API_INPUT_LEGACY; - } else { - // recording a mix of players defined by an external policy, we're rerouting for - // an external policy - *inputType = API_INPUT_MIX_EXT_POLICY_REROUTE; - } - } else if (audio_is_remote_submix_device(device)) { - address = String8("0"); - *inputType = API_INPUT_MIX_CAPTURE; - } else { - *inputType = API_INPUT_LEGACY; - } - // adapt channel selection to input source - switch (inputSource) { - case AUDIO_SOURCE_VOICE_UPLINK: - channelMask = AUDIO_CHANNEL_IN_VOICE_UPLINK; - break; - case AUDIO_SOURCE_VOICE_DOWNLINK: - channelMask = AUDIO_CHANNEL_IN_VOICE_DNLINK; - break; - case AUDIO_SOURCE_VOICE_CALL: - channelMask = AUDIO_CHANNEL_IN_VOICE_UPLINK | AUDIO_CHANNEL_IN_VOICE_DNLINK; - break; - default: - break; - } - if (inputSource == AUDIO_SOURCE_HOTWORD) { - ssize_t index = mSoundTriggerSessions.indexOfKey(session); - if (index >= 0) { - *input = mSoundTriggerSessions.valueFor(session); - isSoundTrigger = true; - flags = (audio_input_flags_t)(flags | AUDIO_INPUT_FLAG_HW_HOTWORD); - ALOGV("SoundTrigger capture on session %d input %d", session, *input); - } else { - halInputSource = AUDIO_SOURCE_VOICE_RECOGNITION; - } - } - } - - sp profile = getInputProfile(device, address, - samplingRate, format, channelMask, - flags); - if (profile == 0) { - //retry without flags - audio_input_flags_t log_flags = flags; - flags = AUDIO_INPUT_FLAG_NONE; - profile = getInputProfile(device, address, - samplingRate, format, channelMask, - flags); - if (profile == 0) { - ALOGW("getInputForAttr() could not find profile for device 0x%X, samplingRate %u," - "format %#x, channelMask 0x%X, flags %#x", - device, samplingRate, format, channelMask, log_flags); - return BAD_VALUE; - } - } - - if (profile->mModule->mHandle == 0) { - ALOGE("getInputForAttr(): HW module %s not opened", profile->mModule->mName); - return NO_INIT; - } - - audio_config_t config = AUDIO_CONFIG_INITIALIZER; - config.sample_rate = samplingRate; - config.channel_mask = channelMask; - config.format = format; - - status_t status = mpClientInterface->openInput(profile->mModule->mHandle, - input, - &config, - &device, - address, - halInputSource, - flags); - - // only accept input with the exact requested set of parameters - if (status != NO_ERROR || *input == AUDIO_IO_HANDLE_NONE || - (samplingRate != config.sample_rate) || - (format != config.format) || - (channelMask != config.channel_mask)) { - ALOGW("getInputForAttr() failed opening input: samplingRate %d, format %d, channelMask %x", - samplingRate, format, channelMask); - if (*input != AUDIO_IO_HANDLE_NONE) { - mpClientInterface->closeInput(*input); - } - return BAD_VALUE; - } - - sp inputDesc = new AudioInputDescriptor(profile); - inputDesc->mInputSource = inputSource; - inputDesc->mRefCount = 0; - inputDesc->mOpenRefCount = 1; - inputDesc->mSamplingRate = samplingRate; - inputDesc->mFormat = format; - inputDesc->mChannelMask = channelMask; - inputDesc->mDevice = device; - inputDesc->mSessions.add(session); - inputDesc->mIsSoundTrigger = isSoundTrigger; - inputDesc->mPolicyMix = policyMix; - - ALOGV("getInputForAttr() returns input type = %d", inputType); - - addInput(*input, inputDesc); - mpClientInterface->onAudioPortListUpdate(); - return NO_ERROR; -} - -status_t AudioPolicyManager::startInput(audio_io_handle_t input, - audio_session_t session) -{ - ALOGV("startInput() input %d", input); - ssize_t index = mInputs.indexOfKey(input); - if (index < 0) { - ALOGW("startInput() unknown input %d", input); - return BAD_VALUE; - } - sp inputDesc = mInputs.valueAt(index); - - index = inputDesc->mSessions.indexOf(session); - if (index < 0) { - ALOGW("startInput() unknown session %d on input %d", session, input); - return BAD_VALUE; - } - - // virtual input devices are compatible with other input devices - if (!isVirtualInputDevice(inputDesc->mDevice)) { - - // for a non-virtual input device, check if there is another (non-virtual) active input - audio_io_handle_t activeInput = getActiveInput(); - if (activeInput != 0 && activeInput != input) { - - // If the already active input uses AUDIO_SOURCE_HOTWORD then it is closed, - // otherwise the active input continues and the new input cannot be started. - sp activeDesc = mInputs.valueFor(activeInput); - if (activeDesc->mInputSource == AUDIO_SOURCE_HOTWORD) { - ALOGW("startInput(%d) preempting low-priority input %d", input, activeInput); - stopInput(activeInput, activeDesc->mSessions.itemAt(0)); - releaseInput(activeInput, activeDesc->mSessions.itemAt(0)); - } else { - ALOGE("startInput(%d) failed: other input %d already started", input, activeInput); - return INVALID_OPERATION; - } - } - } - - if (inputDesc->mRefCount == 0) { - if (activeInputsCount() == 0) { - SoundTrigger::setCaptureState(true); - } - setInputDevice(input, getNewInputDevice(input), true /* force */); - - // automatically enable the remote submix output when input is started if not - // used by a policy mix of type MIX_TYPE_RECORDERS - // For remote submix (a virtual device), we open only one input per capture request. - if (audio_is_remote_submix_device(inputDesc->mDevice)) { - String8 address = String8(""); - if (inputDesc->mPolicyMix == NULL) { - address = String8("0"); - } else if (inputDesc->mPolicyMix->mMixType == MIX_TYPE_PLAYERS) { - address = inputDesc->mPolicyMix->mRegistrationId; - } - if (address != "") { - setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, - AUDIO_POLICY_DEVICE_STATE_AVAILABLE, - address, "remote-submix"); - } - } - } - - ALOGV("AudioPolicyManager::startInput() input source = %d", inputDesc->mInputSource); - - inputDesc->mRefCount++; - return NO_ERROR; -} - -status_t AudioPolicyManager::stopInput(audio_io_handle_t input, - audio_session_t session) -{ - ALOGV("stopInput() input %d", input); - ssize_t index = mInputs.indexOfKey(input); - if (index < 0) { - ALOGW("stopInput() unknown input %d", input); - return BAD_VALUE; - } - sp inputDesc = mInputs.valueAt(index); - - index = inputDesc->mSessions.indexOf(session); - if (index < 0) { - ALOGW("stopInput() unknown session %d on input %d", session, input); - return BAD_VALUE; - } - - if (inputDesc->mRefCount == 0) { - ALOGW("stopInput() input %d already stopped", input); - return INVALID_OPERATION; - } - - inputDesc->mRefCount--; - if (inputDesc->mRefCount == 0) { - - // automatically disable the remote submix output when input is stopped if not - // used by a policy mix of type MIX_TYPE_RECORDERS - if (audio_is_remote_submix_device(inputDesc->mDevice)) { - String8 address = String8(""); - if (inputDesc->mPolicyMix == NULL) { - address = String8("0"); - } else if (inputDesc->mPolicyMix->mMixType == MIX_TYPE_PLAYERS) { - address = inputDesc->mPolicyMix->mRegistrationId; - } - if (address != "") { - setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, - AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE, - address, "remote-submix"); - } - } - - resetInputDevice(input); - - if (activeInputsCount() == 0) { - SoundTrigger::setCaptureState(false); - } - } - return NO_ERROR; -} - -void AudioPolicyManager::releaseInput(audio_io_handle_t input, - audio_session_t session) -{ - ALOGV("releaseInput() %d", input); - ssize_t index = mInputs.indexOfKey(input); - if (index < 0) { - ALOGW("releaseInput() releasing unknown input %d", input); - return; - } - sp inputDesc = mInputs.valueAt(index); - ALOG_ASSERT(inputDesc != 0); - - index = inputDesc->mSessions.indexOf(session); - if (index < 0) { - ALOGW("releaseInput() unknown session %d on input %d", session, input); - return; - } - inputDesc->mSessions.remove(session); - if (inputDesc->mOpenRefCount == 0) { - ALOGW("releaseInput() invalid open ref count %d", inputDesc->mOpenRefCount); - return; - } - inputDesc->mOpenRefCount--; - if (inputDesc->mOpenRefCount > 0) { - ALOGV("releaseInput() exit > 0"); - return; - } - - closeInput(input); - mpClientInterface->onAudioPortListUpdate(); - ALOGV("releaseInput() exit"); -} - -void AudioPolicyManager::closeAllInputs() { - bool patchRemoved = false; - - for(size_t input_index = 0; input_index < mInputs.size(); input_index++) { - sp inputDesc = mInputs.valueAt(input_index); - ssize_t patch_index = mAudioPatches.indexOfKey(inputDesc->mPatchHandle); - if (patch_index >= 0) { - sp patchDesc = mAudioPatches.valueAt(patch_index); - status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0); - mAudioPatches.removeItemsAt(patch_index); - patchRemoved = true; - } - mpClientInterface->closeInput(mInputs.keyAt(input_index)); - } - mInputs.clear(); - nextAudioPortGeneration(); - - if (patchRemoved) { - mpClientInterface->onAudioPatchListUpdate(); - } -} - -void AudioPolicyManager::initStreamVolume(audio_stream_type_t stream, - int indexMin, - int indexMax) -{ - ALOGV("initStreamVolume() stream %d, min %d, max %d", stream , indexMin, indexMax); - if (indexMin < 0 || indexMin >= indexMax) { - ALOGW("initStreamVolume() invalid index limits for stream %d, min %d, max %d", stream , indexMin, indexMax); - return; - } - mStreams[stream].mIndexMin = indexMin; - mStreams[stream].mIndexMax = indexMax; - //FIXME: AUDIO_STREAM_ACCESSIBILITY volume follows AUDIO_STREAM_MUSIC for now - if (stream == AUDIO_STREAM_MUSIC) { - mStreams[AUDIO_STREAM_ACCESSIBILITY].mIndexMin = indexMin; - mStreams[AUDIO_STREAM_ACCESSIBILITY].mIndexMax = indexMax; - } -} - -status_t AudioPolicyManager::setStreamVolumeIndex(audio_stream_type_t stream, - int index, - audio_devices_t device) -{ - - if ((index < mStreams[stream].mIndexMin) || (index > mStreams[stream].mIndexMax)) { - return BAD_VALUE; - } - if (!audio_is_output_device(device)) { - return BAD_VALUE; - } - - // Force max volume if stream cannot be muted - if (!mStreams[stream].mCanBeMuted) index = mStreams[stream].mIndexMax; - - ALOGV("setStreamVolumeIndex() stream %d, device %04x, index %d", - stream, device, index); - - // if device is AUDIO_DEVICE_OUT_DEFAULT set default value and - // clear all device specific values - if (device == AUDIO_DEVICE_OUT_DEFAULT) { - mStreams[stream].mIndexCur.clear(); - } - mStreams[stream].mIndexCur.add(device, index); - - // update volume on all outputs whose current device is also selected by the same - // strategy as the device specified by the caller - audio_devices_t strategyDevice = getDeviceForStrategy(getStrategy(stream), true /*fromCache*/); - - - //FIXME: AUDIO_STREAM_ACCESSIBILITY volume follows AUDIO_STREAM_MUSIC for now - audio_devices_t accessibilityDevice = AUDIO_DEVICE_NONE; - if (stream == AUDIO_STREAM_MUSIC) { - mStreams[AUDIO_STREAM_ACCESSIBILITY].mIndexCur.add(device, index); - accessibilityDevice = getDeviceForStrategy(STRATEGY_ACCESSIBILITY, true /*fromCache*/); - } - if ((device != AUDIO_DEVICE_OUT_DEFAULT) && - (device & (strategyDevice | accessibilityDevice)) == 0) { - return NO_ERROR; - } - status_t status = NO_ERROR; - for (size_t i = 0; i < mOutputs.size(); i++) { - audio_devices_t curDevice = - getDeviceForVolume(mOutputs.valueAt(i)->device()); - if ((device == AUDIO_DEVICE_OUT_DEFAULT) || ((curDevice & strategyDevice) != 0)) { - status_t volStatus = checkAndSetVolume(stream, index, mOutputs.keyAt(i), curDevice); - if (volStatus != NO_ERROR) { - status = volStatus; - } - } - if ((device == AUDIO_DEVICE_OUT_DEFAULT) || ((curDevice & accessibilityDevice) != 0)) { - status_t volStatus = checkAndSetVolume(AUDIO_STREAM_ACCESSIBILITY, - index, mOutputs.keyAt(i), curDevice); - } - } - return status; -} - -status_t AudioPolicyManager::getStreamVolumeIndex(audio_stream_type_t stream, - int *index, - audio_devices_t device) -{ - if (index == NULL) { - return BAD_VALUE; - } - if (!audio_is_output_device(device)) { - return BAD_VALUE; - } - // if device is AUDIO_DEVICE_OUT_DEFAULT, return volume for device corresponding to - // the strategy the stream belongs to. - if (device == AUDIO_DEVICE_OUT_DEFAULT) { - device = getDeviceForStrategy(getStrategy(stream), true /*fromCache*/); - } - device = getDeviceForVolume(device); - - *index = mStreams[stream].getVolumeIndex(device); - ALOGV("getStreamVolumeIndex() stream %d device %08x index %d", stream, device, *index); - return NO_ERROR; -} - -audio_io_handle_t AudioPolicyManager::selectOutputForEffects( - const SortedVector& outputs) -{ - // select one output among several suitable for global effects. - // The priority is as follows: - // 1: An offloaded output. If the effect ends up not being offloadable, - // AudioFlinger will invalidate the track and the offloaded output - // will be closed causing the effect to be moved to a PCM output. - // 2: A deep buffer output - // 3: the first output in the list - - if (outputs.size() == 0) { - return 0; - } - - audio_io_handle_t outputOffloaded = 0; - audio_io_handle_t outputDeepBuffer = 0; - - for (size_t i = 0; i < outputs.size(); i++) { - sp desc = mOutputs.valueFor(outputs[i]); - ALOGV("selectOutputForEffects outputs[%zu] flags %x", i, desc->mFlags); - if ((desc->mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) { - outputOffloaded = outputs[i]; - } - if ((desc->mFlags & AUDIO_OUTPUT_FLAG_DEEP_BUFFER) != 0) { - outputDeepBuffer = outputs[i]; - } - } - - ALOGV("selectOutputForEffects outputOffloaded %d outputDeepBuffer %d", - outputOffloaded, outputDeepBuffer); - if (outputOffloaded != 0) { - return outputOffloaded; - } - if (outputDeepBuffer != 0) { - return outputDeepBuffer; - } - - return outputs[0]; -} - -audio_io_handle_t AudioPolicyManager::getOutputForEffect(const effect_descriptor_t *desc) -{ - // apply simple rule where global effects are attached to the same output as MUSIC streams - - routing_strategy strategy = getStrategy(AUDIO_STREAM_MUSIC); - audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/); - SortedVector dstOutputs = getOutputsForDevice(device, mOutputs); - - audio_io_handle_t output = selectOutputForEffects(dstOutputs); - ALOGV("getOutputForEffect() got output %d for fx %s flags %x", - output, (desc == NULL) ? "unspecified" : desc->name, (desc == NULL) ? 0 : desc->flags); - - return output; -} - -status_t AudioPolicyManager::registerEffect(const effect_descriptor_t *desc, - audio_io_handle_t io, - uint32_t strategy, - int session, - int id) -{ - ssize_t index = mOutputs.indexOfKey(io); - if (index < 0) { - index = mInputs.indexOfKey(io); - if (index < 0) { - ALOGW("registerEffect() unknown io %d", io); - return INVALID_OPERATION; - } - } - - if (mTotalEffectsMemory + desc->memoryUsage > getMaxEffectsMemory()) { - ALOGW("registerEffect() memory limit exceeded for Fx %s, Memory %d KB", - desc->name, desc->memoryUsage); - return INVALID_OPERATION; - } - mTotalEffectsMemory += desc->memoryUsage; - ALOGV("registerEffect() effect %s, io %d, strategy %d session %d id %d", - desc->name, io, strategy, session, id); - ALOGV("registerEffect() memory %d, total memory %d", desc->memoryUsage, mTotalEffectsMemory); - - sp effectDesc = new EffectDescriptor(); - memcpy (&effectDesc->mDesc, desc, sizeof(effect_descriptor_t)); - effectDesc->mIo = io; - effectDesc->mStrategy = (routing_strategy)strategy; - effectDesc->mSession = session; - effectDesc->mEnabled = false; - - mEffects.add(id, effectDesc); - - return NO_ERROR; -} - -status_t AudioPolicyManager::unregisterEffect(int id) -{ - ssize_t index = mEffects.indexOfKey(id); - if (index < 0) { - ALOGW("unregisterEffect() unknown effect ID %d", id); - return INVALID_OPERATION; - } - - sp effectDesc = mEffects.valueAt(index); - - setEffectEnabled(effectDesc, false); - - if (mTotalEffectsMemory < effectDesc->mDesc.memoryUsage) { - ALOGW("unregisterEffect() memory %d too big for total %d", - effectDesc->mDesc.memoryUsage, mTotalEffectsMemory); - effectDesc->mDesc.memoryUsage = mTotalEffectsMemory; - } - mTotalEffectsMemory -= effectDesc->mDesc.memoryUsage; - ALOGV("unregisterEffect() effect %s, ID %d, memory %d total memory %d", - effectDesc->mDesc.name, id, effectDesc->mDesc.memoryUsage, mTotalEffectsMemory); - - mEffects.removeItem(id); - - return NO_ERROR; -} - -status_t AudioPolicyManager::setEffectEnabled(int id, bool enabled) -{ - ssize_t index = mEffects.indexOfKey(id); - if (index < 0) { - ALOGW("unregisterEffect() unknown effect ID %d", id); - return INVALID_OPERATION; - } - - return setEffectEnabled(mEffects.valueAt(index), enabled); -} - -status_t AudioPolicyManager::setEffectEnabled(const sp& effectDesc, bool enabled) -{ - if (enabled == effectDesc->mEnabled) { - ALOGV("setEffectEnabled(%s) effect already %s", - enabled?"true":"false", enabled?"enabled":"disabled"); - return INVALID_OPERATION; - } - - if (enabled) { - if (mTotalEffectsCpuLoad + effectDesc->mDesc.cpuLoad > getMaxEffectsCpuLoad()) { - ALOGW("setEffectEnabled(true) CPU Load limit exceeded for Fx %s, CPU %f MIPS", - effectDesc->mDesc.name, (float)effectDesc->mDesc.cpuLoad/10); - return INVALID_OPERATION; - } - mTotalEffectsCpuLoad += effectDesc->mDesc.cpuLoad; - ALOGV("setEffectEnabled(true) total CPU %d", mTotalEffectsCpuLoad); - } else { - if (mTotalEffectsCpuLoad < effectDesc->mDesc.cpuLoad) { - ALOGW("setEffectEnabled(false) CPU load %d too high for total %d", - effectDesc->mDesc.cpuLoad, mTotalEffectsCpuLoad); - effectDesc->mDesc.cpuLoad = mTotalEffectsCpuLoad; - } - mTotalEffectsCpuLoad -= effectDesc->mDesc.cpuLoad; - ALOGV("setEffectEnabled(false) total CPU %d", mTotalEffectsCpuLoad); - } - effectDesc->mEnabled = enabled; - return NO_ERROR; -} - -bool AudioPolicyManager::isNonOffloadableEffectEnabled() -{ - for (size_t i = 0; i < mEffects.size(); i++) { - sp effectDesc = mEffects.valueAt(i); - if (effectDesc->mEnabled && (effectDesc->mStrategy == STRATEGY_MEDIA) && - ((effectDesc->mDesc.flags & EFFECT_FLAG_OFFLOAD_SUPPORTED) == 0)) { - ALOGV("isNonOffloadableEffectEnabled() non offloadable effect %s enabled on session %d", - effectDesc->mDesc.name, effectDesc->mSession); - return true; - } - } - return false; -} - -bool AudioPolicyManager::isStreamActive(audio_stream_type_t stream, uint32_t inPastMs) const -{ - nsecs_t sysTime = systemTime(); - for (size_t i = 0; i < mOutputs.size(); i++) { - const sp outputDesc = mOutputs.valueAt(i); - if (outputDesc->isStreamActive(stream, inPastMs, sysTime)) { - return true; - } - } - return false; -} - -bool AudioPolicyManager::isStreamActiveRemotely(audio_stream_type_t stream, - uint32_t inPastMs) const -{ - nsecs_t sysTime = systemTime(); - for (size_t i = 0; i < mOutputs.size(); i++) { - const sp outputDesc = mOutputs.valueAt(i); - if (((outputDesc->device() & APM_AUDIO_OUT_DEVICE_REMOTE_ALL) != 0) && - outputDesc->isStreamActive(stream, inPastMs, sysTime)) { - // do not consider re routing (when the output is going to a dynamic policy) - // as "remote playback" - if (outputDesc->mPolicyMix == NULL) { - return true; - } - } - } - return false; -} - -bool AudioPolicyManager::isSourceActive(audio_source_t source) const -{ - for (size_t i = 0; i < mInputs.size(); i++) { - const sp inputDescriptor = mInputs.valueAt(i); - if (inputDescriptor->mRefCount == 0) { - continue; - } - if (inputDescriptor->mInputSource == (int)source) { - return true; - } - // AUDIO_SOURCE_HOTWORD is equivalent to AUDIO_SOURCE_VOICE_RECOGNITION only if it - // corresponds to an active capture triggered by a hardware hotword recognition - if ((source == AUDIO_SOURCE_VOICE_RECOGNITION) && - (inputDescriptor->mInputSource == AUDIO_SOURCE_HOTWORD)) { - // FIXME: we should not assume that the first session is the active one and keep - // activity count per session. Same in startInput(). - ssize_t index = mSoundTriggerSessions.indexOfKey(inputDescriptor->mSessions.itemAt(0)); - if (index >= 0) { - return true; - } - } - } - return false; -} - -// Register a list of custom mixes with their attributes and format. -// When a mix is registered, corresponding input and output profiles are -// added to the remote submix hw module. The profile contains only the -// parameters (sampling rate, format...) specified by the mix. -// The corresponding input remote submix device is also connected. -// -// When a remote submix device is connected, the address is checked to select the -// appropriate profile and the corresponding input or output stream is opened. -// -// When capture starts, getInputForAttr() will: -// - 1 look for a mix matching the address passed in attribtutes tags if any -// - 2 if none found, getDeviceForInputSource() will: -// - 2.1 look for a mix matching the attributes source -// - 2.2 if none found, default to device selection by policy rules -// At this time, the corresponding output remote submix device is also connected -// and active playback use cases can be transferred to this mix if needed when reconnecting -// after AudioTracks are invalidated -// -// When playback starts, getOutputForAttr() will: -// - 1 look for a mix matching the address passed in attribtutes tags if any -// - 2 if none found, look for a mix matching the attributes usage -// - 3 if none found, default to device and output selection by policy rules. - -status_t AudioPolicyManager::registerPolicyMixes(Vector mixes) -{ - sp module; - for (size_t i = 0; i < mHwModules.size(); i++) { - if (strcmp(AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX, mHwModules[i]->mName) == 0 && - mHwModules[i]->mHandle != 0) { - module = mHwModules[i]; - break; - } - } - - if (module == 0) { - return INVALID_OPERATION; - } - - ALOGV("registerPolicyMixes() num mixes %d", mixes.size()); - - for (size_t i = 0; i < mixes.size(); i++) { - String8 address = mixes[i].mRegistrationId; - ssize_t index = mPolicyMixes.indexOfKey(address); - if (index >= 0) { - ALOGE("registerPolicyMixes(): mix for address %s already registered", address.string()); - continue; - } - audio_config_t outputConfig = mixes[i].mFormat; - audio_config_t inputConfig = mixes[i].mFormat; - // NOTE: audio flinger mixer does not support mono output: configure remote submix HAL in - // stereo and let audio flinger do the channel conversion if needed. - outputConfig.channel_mask = AUDIO_CHANNEL_OUT_STEREO; - inputConfig.channel_mask = AUDIO_CHANNEL_IN_STEREO; - module->addOutputProfile(address, &outputConfig, - AUDIO_DEVICE_OUT_REMOTE_SUBMIX, address); - module->addInputProfile(address, &inputConfig, - AUDIO_DEVICE_IN_REMOTE_SUBMIX, address); - sp policyMix = new AudioPolicyMix(); - policyMix->mMix = mixes[i]; - mPolicyMixes.add(address, policyMix); - if (mixes[i].mMixType == MIX_TYPE_PLAYERS) { - setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX, - AUDIO_POLICY_DEVICE_STATE_AVAILABLE, - address.string(), "remote-submix"); - } else { - setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, - AUDIO_POLICY_DEVICE_STATE_AVAILABLE, - address.string(), "remote-submix"); - } - } - return NO_ERROR; -} - -status_t AudioPolicyManager::unregisterPolicyMixes(Vector mixes) -{ - sp module; - for (size_t i = 0; i < mHwModules.size(); i++) { - if (strcmp(AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX, mHwModules[i]->mName) == 0 && - mHwModules[i]->mHandle != 0) { - module = mHwModules[i]; - break; - } - } - - if (module == 0) { - return INVALID_OPERATION; - } - - ALOGV("unregisterPolicyMixes() num mixes %d", mixes.size()); - - for (size_t i = 0; i < mixes.size(); i++) { - String8 address = mixes[i].mRegistrationId; - ssize_t index = mPolicyMixes.indexOfKey(address); - if (index < 0) { - ALOGE("unregisterPolicyMixes(): mix for address %s not registered", address.string()); - continue; - } - - mPolicyMixes.removeItemsAt(index); - - if (getDeviceConnectionState(AUDIO_DEVICE_IN_REMOTE_SUBMIX, address.string()) == - AUDIO_POLICY_DEVICE_STATE_AVAILABLE) - { - setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX, - AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE, - address.string(), "remote-submix"); - } - - if (getDeviceConnectionState(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, address.string()) == - AUDIO_POLICY_DEVICE_STATE_AVAILABLE) - { - setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, - AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE, - address.string(), "remote-submix"); - } - module->removeOutputProfile(address); - module->removeInputProfile(address); - } - return NO_ERROR; -} - - -status_t AudioPolicyManager::dump(int fd) -{ - const size_t SIZE = 256; - char buffer[SIZE]; - String8 result; - - snprintf(buffer, SIZE, "\nAudioPolicyManager Dump: %p\n", this); - result.append(buffer); - - snprintf(buffer, SIZE, " Primary Output: %d\n", mPrimaryOutput); - result.append(buffer); - snprintf(buffer, SIZE, " Phone state: %d\n", mPhoneState); - result.append(buffer); - snprintf(buffer, SIZE, " Force use for communications %d\n", - mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION]); - result.append(buffer); - snprintf(buffer, SIZE, " Force use for media %d\n", mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA]); - result.append(buffer); - snprintf(buffer, SIZE, " Force use for record %d\n", mForceUse[AUDIO_POLICY_FORCE_FOR_RECORD]); - result.append(buffer); - snprintf(buffer, SIZE, " Force use for dock %d\n", mForceUse[AUDIO_POLICY_FORCE_FOR_DOCK]); - result.append(buffer); - snprintf(buffer, SIZE, " Force use for system %d\n", mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM]); - result.append(buffer); - snprintf(buffer, SIZE, " Force use for hdmi system audio %d\n", - mForceUse[AUDIO_POLICY_FORCE_FOR_HDMI_SYSTEM_AUDIO]); - result.append(buffer); - - snprintf(buffer, SIZE, " Available output devices:\n"); - result.append(buffer); - write(fd, result.string(), result.size()); - for (size_t i = 0; i < mAvailableOutputDevices.size(); i++) { - mAvailableOutputDevices[i]->dump(fd, 2, i); - } - snprintf(buffer, SIZE, "\n Available input devices:\n"); - write(fd, buffer, strlen(buffer)); - for (size_t i = 0; i < mAvailableInputDevices.size(); i++) { - mAvailableInputDevices[i]->dump(fd, 2, i); - } - - snprintf(buffer, SIZE, "\nHW Modules dump:\n"); - write(fd, buffer, strlen(buffer)); - for (size_t i = 0; i < mHwModules.size(); i++) { - snprintf(buffer, SIZE, "- HW Module %zu:\n", i + 1); - write(fd, buffer, strlen(buffer)); - mHwModules[i]->dump(fd); - } - - snprintf(buffer, SIZE, "\nOutputs dump:\n"); - write(fd, buffer, strlen(buffer)); - for (size_t i = 0; i < mOutputs.size(); i++) { - snprintf(buffer, SIZE, "- Output %d dump:\n", mOutputs.keyAt(i)); - write(fd, buffer, strlen(buffer)); - mOutputs.valueAt(i)->dump(fd); - } - - snprintf(buffer, SIZE, "\nInputs dump:\n"); - write(fd, buffer, strlen(buffer)); - for (size_t i = 0; i < mInputs.size(); i++) { - snprintf(buffer, SIZE, "- Input %d dump:\n", mInputs.keyAt(i)); - write(fd, buffer, strlen(buffer)); - mInputs.valueAt(i)->dump(fd); - } - - snprintf(buffer, SIZE, "\nStreams dump:\n"); - write(fd, buffer, strlen(buffer)); - snprintf(buffer, SIZE, - " Stream Can be muted Index Min Index Max Index Cur [device : index]...\n"); - write(fd, buffer, strlen(buffer)); - for (size_t i = 0; i < AUDIO_STREAM_CNT; i++) { - snprintf(buffer, SIZE, " %02zu ", i); - write(fd, buffer, strlen(buffer)); - mStreams[i].dump(fd); - } - - snprintf(buffer, SIZE, "\nTotal Effects CPU: %f MIPS, Total Effects memory: %d KB\n", - (float)mTotalEffectsCpuLoad/10, mTotalEffectsMemory); - write(fd, buffer, strlen(buffer)); - - snprintf(buffer, SIZE, "Registered effects:\n"); - write(fd, buffer, strlen(buffer)); - for (size_t i = 0; i < mEffects.size(); i++) { - snprintf(buffer, SIZE, "- Effect %d dump:\n", mEffects.keyAt(i)); - write(fd, buffer, strlen(buffer)); - mEffects.valueAt(i)->dump(fd); - } - - snprintf(buffer, SIZE, "\nAudio Patches:\n"); - write(fd, buffer, strlen(buffer)); - for (size_t i = 0; i < mAudioPatches.size(); i++) { - mAudioPatches[i]->dump(fd, 2, i); - } - - return NO_ERROR; -} - -// This function checks for the parameters which can be offloaded. -// This can be enhanced depending on the capability of the DSP and policy -// of the system. -bool AudioPolicyManager::isOffloadSupported(const audio_offload_info_t& offloadInfo) -{ - ALOGV("isOffloadSupported: SR=%u, CM=0x%x, Format=0x%x, StreamType=%d," - " BitRate=%u, duration=%" PRId64 " us, has_video=%d", - offloadInfo.sample_rate, offloadInfo.channel_mask, - offloadInfo.format, - offloadInfo.stream_type, offloadInfo.bit_rate, offloadInfo.duration_us, - offloadInfo.has_video); - - // Check if offload has been disabled - char propValue[PROPERTY_VALUE_MAX]; - if (property_get("audio.offload.disable", propValue, "0")) { - if (atoi(propValue) != 0) { - ALOGV("offload disabled by audio.offload.disable=%s", propValue ); - return false; - } - } - - // Check if stream type is music, then only allow offload as of now. - if (offloadInfo.stream_type != AUDIO_STREAM_MUSIC) - { - ALOGV("isOffloadSupported: stream_type != MUSIC, returning false"); - return false; - } - - //TODO: enable audio offloading with video when ready - if (offloadInfo.has_video) - { - ALOGV("isOffloadSupported: has_video == true, returning false"); - return false; - } - - //If duration is less than minimum value defined in property, return false - if (property_get("audio.offload.min.duration.secs", propValue, NULL)) { - if (offloadInfo.duration_us < (atoi(propValue) * 1000000 )) { - ALOGV("Offload denied by duration < audio.offload.min.duration.secs(=%s)", propValue); - return false; - } - } else if (offloadInfo.duration_us < OFFLOAD_DEFAULT_MIN_DURATION_SECS * 1000000) { - ALOGV("Offload denied by duration < default min(=%u)", OFFLOAD_DEFAULT_MIN_DURATION_SECS); - return false; - } - - // Do not allow offloading if one non offloadable effect is enabled. This prevents from - // creating an offloaded track and tearing it down immediately after start when audioflinger - // detects there is an active non offloadable effect. - // FIXME: We should check the audio session here but we do not have it in this context. - // This may prevent offloading in rare situations where effects are left active by apps - // in the background. - if (isNonOffloadableEffectEnabled()) { - return false; - } - - // See if there is a profile to support this. - // AUDIO_DEVICE_NONE - sp profile = getProfileForDirectOutput(AUDIO_DEVICE_NONE /*ignore device */, - offloadInfo.sample_rate, - offloadInfo.format, - offloadInfo.channel_mask, - AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD); - ALOGV("isOffloadSupported() profile %sfound", profile != 0 ? "" : "NOT "); - return (profile != 0); -} - -status_t AudioPolicyManager::listAudioPorts(audio_port_role_t role, - audio_port_type_t type, - unsigned int *num_ports, - struct audio_port *ports, - unsigned int *generation) -{ - if (num_ports == NULL || (*num_ports != 0 && ports == NULL) || - generation == NULL) { - return BAD_VALUE; - } - ALOGV("listAudioPorts() role %d type %d num_ports %d ports %p", role, type, *num_ports, ports); - if (ports == NULL) { - *num_ports = 0; - } - - size_t portsWritten = 0; - size_t portsMax = *num_ports; - *num_ports = 0; - if (type == AUDIO_PORT_TYPE_NONE || type == AUDIO_PORT_TYPE_DEVICE) { - if (role == AUDIO_PORT_ROLE_SINK || role == AUDIO_PORT_ROLE_NONE) { - for (size_t i = 0; - i < mAvailableOutputDevices.size() && portsWritten < portsMax; i++) { - mAvailableOutputDevices[i]->toAudioPort(&ports[portsWritten++]); - } - *num_ports += mAvailableOutputDevices.size(); - } - if (role == AUDIO_PORT_ROLE_SOURCE || role == AUDIO_PORT_ROLE_NONE) { - for (size_t i = 0; - i < mAvailableInputDevices.size() && portsWritten < portsMax; i++) { - mAvailableInputDevices[i]->toAudioPort(&ports[portsWritten++]); - } - *num_ports += mAvailableInputDevices.size(); - } - } - if (type == AUDIO_PORT_TYPE_NONE || type == AUDIO_PORT_TYPE_MIX) { - if (role == AUDIO_PORT_ROLE_SINK || role == AUDIO_PORT_ROLE_NONE) { - for (size_t i = 0; i < mInputs.size() && portsWritten < portsMax; i++) { - mInputs[i]->toAudioPort(&ports[portsWritten++]); - } - *num_ports += mInputs.size(); - } - if (role == AUDIO_PORT_ROLE_SOURCE || role == AUDIO_PORT_ROLE_NONE) { - size_t numOutputs = 0; - for (size_t i = 0; i < mOutputs.size(); i++) { - if (!mOutputs[i]->isDuplicated()) { - numOutputs++; - if (portsWritten < portsMax) { - mOutputs[i]->toAudioPort(&ports[portsWritten++]); - } - } - } - *num_ports += numOutputs; - } - } - *generation = curAudioPortGeneration(); - ALOGV("listAudioPorts() got %zu ports needed %d", portsWritten, *num_ports); - return NO_ERROR; -} - -status_t AudioPolicyManager::getAudioPort(struct audio_port *port __unused) -{ - return NO_ERROR; -} - -sp AudioPolicyManager::getOutputFromId( - audio_port_handle_t id) const -{ - sp outputDesc = NULL; - for (size_t i = 0; i < mOutputs.size(); i++) { - outputDesc = mOutputs.valueAt(i); - if (outputDesc->mId == id) { - break; - } - } - return outputDesc; -} - -sp AudioPolicyManager::getInputFromId( - audio_port_handle_t id) const -{ - sp inputDesc = NULL; - for (size_t i = 0; i < mInputs.size(); i++) { - inputDesc = mInputs.valueAt(i); - if (inputDesc->mId == id) { - break; - } - } - return inputDesc; -} - -sp AudioPolicyManager::getModuleForDevice( - audio_devices_t device) const -{ - sp module; - - for (size_t i = 0; i < mHwModules.size(); i++) { - if (mHwModules[i]->mHandle == 0) { - continue; - } - if (audio_is_output_device(device)) { - for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++) - { - if (mHwModules[i]->mOutputProfiles[j]->mSupportedDevices.types() & device) { - return mHwModules[i]; - } - } - } else { - for (size_t j = 0; j < mHwModules[i]->mInputProfiles.size(); j++) { - if (mHwModules[i]->mInputProfiles[j]->mSupportedDevices.types() & - device & ~AUDIO_DEVICE_BIT_IN) { - return mHwModules[i]; - } - } - } - } - return module; -} - -sp AudioPolicyManager::getModuleFromName(const char *name) const -{ - sp module; - - for (size_t i = 0; i < mHwModules.size(); i++) - { - if (strcmp(mHwModules[i]->mName, name) == 0) { - return mHwModules[i]; - } - } - return module; -} - -audio_devices_t AudioPolicyManager::availablePrimaryOutputDevices() -{ - sp outputDesc = mOutputs.valueFor(mPrimaryOutput); - audio_devices_t devices = outputDesc->mProfile->mSupportedDevices.types(); - return devices & mAvailableOutputDevices.types(); -} - -audio_devices_t AudioPolicyManager::availablePrimaryInputDevices() -{ - audio_module_handle_t primaryHandle = - mOutputs.valueFor(mPrimaryOutput)->mProfile->mModule->mHandle; - audio_devices_t devices = AUDIO_DEVICE_NONE; - for (size_t i = 0; i < mAvailableInputDevices.size(); i++) { - if (mAvailableInputDevices[i]->mModule->mHandle == primaryHandle) { - devices |= mAvailableInputDevices[i]->mDeviceType; - } - } - return devices; -} - -status_t AudioPolicyManager::createAudioPatch(const struct audio_patch *patch, - audio_patch_handle_t *handle, - uid_t uid) -{ - ALOGV("createAudioPatch()"); - - if (handle == NULL || patch == NULL) { - return BAD_VALUE; - } - ALOGV("createAudioPatch() num sources %d num sinks %d", patch->num_sources, patch->num_sinks); - - if (patch->num_sources == 0 || patch->num_sources > AUDIO_PATCH_PORTS_MAX || - patch->num_sinks == 0 || patch->num_sinks > AUDIO_PATCH_PORTS_MAX) { - return BAD_VALUE; - } - // only one source per audio patch supported for now - if (patch->num_sources > 1) { - return INVALID_OPERATION; - } - - if (patch->sources[0].role != AUDIO_PORT_ROLE_SOURCE) { - return INVALID_OPERATION; - } - for (size_t i = 0; i < patch->num_sinks; i++) { - if (patch->sinks[i].role != AUDIO_PORT_ROLE_SINK) { - return INVALID_OPERATION; - } - } - - sp patchDesc; - ssize_t index = mAudioPatches.indexOfKey(*handle); - - ALOGV("createAudioPatch source id %d role %d type %d", patch->sources[0].id, - patch->sources[0].role, - patch->sources[0].type); -#if LOG_NDEBUG == 0 - for (size_t i = 0; i < patch->num_sinks; i++) { - ALOGV("createAudioPatch sink %d: id %d role %d type %d", i, patch->sinks[i].id, - patch->sinks[i].role, - patch->sinks[i].type); - } -#endif - - if (index >= 0) { - patchDesc = mAudioPatches.valueAt(index); - ALOGV("createAudioPatch() mUidCached %d patchDesc->mUid %d uid %d", - mUidCached, patchDesc->mUid, uid); - if (patchDesc->mUid != mUidCached && uid != patchDesc->mUid) { - return INVALID_OPERATION; - } - } else { - *handle = 0; - } - - if (patch->sources[0].type == AUDIO_PORT_TYPE_MIX) { - sp outputDesc = getOutputFromId(patch->sources[0].id); - if (outputDesc == NULL) { - ALOGV("createAudioPatch() output not found for id %d", patch->sources[0].id); - return BAD_VALUE; - } - ALOG_ASSERT(!outputDesc->isDuplicated(),"duplicated output %d in source in ports", - outputDesc->mIoHandle); - if (patchDesc != 0) { - if (patchDesc->mPatch.sources[0].id != patch->sources[0].id) { - ALOGV("createAudioPatch() source id differs for patch current id %d new id %d", - patchDesc->mPatch.sources[0].id, patch->sources[0].id); - return BAD_VALUE; - } - } - DeviceVector devices; - for (size_t i = 0; i < patch->num_sinks; i++) { - // Only support mix to devices connection - // TODO add support for mix to mix connection - if (patch->sinks[i].type != AUDIO_PORT_TYPE_DEVICE) { - ALOGV("createAudioPatch() source mix but sink is not a device"); - return INVALID_OPERATION; - } - sp devDesc = - mAvailableOutputDevices.getDeviceFromId(patch->sinks[i].id); - if (devDesc == 0) { - ALOGV("createAudioPatch() out device not found for id %d", patch->sinks[i].id); - return BAD_VALUE; - } - - if (!outputDesc->mProfile->isCompatibleProfile(devDesc->mDeviceType, - devDesc->mAddress, - patch->sources[0].sample_rate, - NULL, // updatedSamplingRate - patch->sources[0].format, - patch->sources[0].channel_mask, - AUDIO_OUTPUT_FLAG_NONE /*FIXME*/)) { - ALOGV("createAudioPatch() profile not supported for device %08x", - devDesc->mDeviceType); - return INVALID_OPERATION; - } - devices.add(devDesc); - } - if (devices.size() == 0) { - return INVALID_OPERATION; - } - - // TODO: reconfigure output format and channels here - ALOGV("createAudioPatch() setting device %08x on output %d", - devices.types(), outputDesc->mIoHandle); - setOutputDevice(outputDesc->mIoHandle, devices.types(), true, 0, handle); - index = mAudioPatches.indexOfKey(*handle); - if (index >= 0) { - if (patchDesc != 0 && patchDesc != mAudioPatches.valueAt(index)) { - ALOGW("createAudioPatch() setOutputDevice() did not reuse the patch provided"); - } - patchDesc = mAudioPatches.valueAt(index); - patchDesc->mUid = uid; - ALOGV("createAudioPatch() success"); - } else { - ALOGW("createAudioPatch() setOutputDevice() failed to create a patch"); - return INVALID_OPERATION; - } - } else if (patch->sources[0].type == AUDIO_PORT_TYPE_DEVICE) { - if (patch->sinks[0].type == AUDIO_PORT_TYPE_MIX) { - // input device to input mix connection - // only one sink supported when connecting an input device to a mix - if (patch->num_sinks > 1) { - return INVALID_OPERATION; - } - sp inputDesc = getInputFromId(patch->sinks[0].id); - if (inputDesc == NULL) { - return BAD_VALUE; - } - if (patchDesc != 0) { - if (patchDesc->mPatch.sinks[0].id != patch->sinks[0].id) { - return BAD_VALUE; - } - } - sp devDesc = - mAvailableInputDevices.getDeviceFromId(patch->sources[0].id); - if (devDesc == 0) { - return BAD_VALUE; - } - - if (!inputDesc->mProfile->isCompatibleProfile(devDesc->mDeviceType, - devDesc->mAddress, - patch->sinks[0].sample_rate, - NULL, /*updatedSampleRate*/ - patch->sinks[0].format, - patch->sinks[0].channel_mask, - // FIXME for the parameter type, - // and the NONE - (audio_output_flags_t) - AUDIO_INPUT_FLAG_NONE)) { - return INVALID_OPERATION; - } - // TODO: reconfigure output format and channels here - ALOGV("createAudioPatch() setting device %08x on output %d", - devDesc->mDeviceType, inputDesc->mIoHandle); - setInputDevice(inputDesc->mIoHandle, devDesc->mDeviceType, true, handle); - index = mAudioPatches.indexOfKey(*handle); - if (index >= 0) { - if (patchDesc != 0 && patchDesc != mAudioPatches.valueAt(index)) { - ALOGW("createAudioPatch() setInputDevice() did not reuse the patch provided"); - } - patchDesc = mAudioPatches.valueAt(index); - patchDesc->mUid = uid; - ALOGV("createAudioPatch() success"); - } else { - ALOGW("createAudioPatch() setInputDevice() failed to create a patch"); - return INVALID_OPERATION; - } - } else if (patch->sinks[0].type == AUDIO_PORT_TYPE_DEVICE) { - // device to device connection - if (patchDesc != 0) { - if (patchDesc->mPatch.sources[0].id != patch->sources[0].id) { - return BAD_VALUE; - } - } - sp srcDeviceDesc = - mAvailableInputDevices.getDeviceFromId(patch->sources[0].id); - if (srcDeviceDesc == 0) { - return BAD_VALUE; - } - - //update source and sink with our own data as the data passed in the patch may - // be incomplete. - struct audio_patch newPatch = *patch; - srcDeviceDesc->toAudioPortConfig(&newPatch.sources[0], &patch->sources[0]); - - for (size_t i = 0; i < patch->num_sinks; i++) { - if (patch->sinks[i].type != AUDIO_PORT_TYPE_DEVICE) { - ALOGV("createAudioPatch() source device but one sink is not a device"); - return INVALID_OPERATION; - } - - sp sinkDeviceDesc = - mAvailableOutputDevices.getDeviceFromId(patch->sinks[i].id); - if (sinkDeviceDesc == 0) { - return BAD_VALUE; - } - sinkDeviceDesc->toAudioPortConfig(&newPatch.sinks[i], &patch->sinks[i]); - - if (srcDeviceDesc->mModule != sinkDeviceDesc->mModule) { - // only one sink supported when connected devices across HW modules - if (patch->num_sinks > 1) { - return INVALID_OPERATION; - } - SortedVector outputs = - getOutputsForDevice(sinkDeviceDesc->mDeviceType, - mOutputs); - // if the sink device is reachable via an opened output stream, request to go via - // this output stream by adding a second source to the patch description - audio_io_handle_t output = selectOutput(outputs, - AUDIO_OUTPUT_FLAG_NONE, - AUDIO_FORMAT_INVALID); - if (output != AUDIO_IO_HANDLE_NONE) { - sp outputDesc = mOutputs.valueFor(output); - if (outputDesc->isDuplicated()) { - return INVALID_OPERATION; - } - outputDesc->toAudioPortConfig(&newPatch.sources[1], &patch->sources[0]); - newPatch.num_sources = 2; - } - } - } - // TODO: check from routing capabilities in config file and other conflicting patches - - audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE; - if (index >= 0) { - afPatchHandle = patchDesc->mAfPatchHandle; - } - - status_t status = mpClientInterface->createAudioPatch(&newPatch, - &afPatchHandle, - 0); - ALOGV("createAudioPatch() patch panel returned %d patchHandle %d", - status, afPatchHandle); - if (status == NO_ERROR) { - if (index < 0) { - patchDesc = new AudioPatch((audio_patch_handle_t)nextUniqueId(), - &newPatch, uid); - addAudioPatch(patchDesc->mHandle, patchDesc); - } else { - patchDesc->mPatch = newPatch; - } - patchDesc->mAfPatchHandle = afPatchHandle; - *handle = patchDesc->mHandle; - nextAudioPortGeneration(); - mpClientInterface->onAudioPatchListUpdate(); - } else { - ALOGW("createAudioPatch() patch panel could not connect device patch, error %d", - status); - return INVALID_OPERATION; - } - } else { - return BAD_VALUE; - } - } else { - return BAD_VALUE; - } - return NO_ERROR; -} - -status_t AudioPolicyManager::releaseAudioPatch(audio_patch_handle_t handle, - uid_t uid) -{ - ALOGV("releaseAudioPatch() patch %d", handle); - - ssize_t index = mAudioPatches.indexOfKey(handle); - - if (index < 0) { - return BAD_VALUE; - } - sp patchDesc = mAudioPatches.valueAt(index); - ALOGV("releaseAudioPatch() mUidCached %d patchDesc->mUid %d uid %d", - mUidCached, patchDesc->mUid, uid); - if (patchDesc->mUid != mUidCached && uid != patchDesc->mUid) { - return INVALID_OPERATION; - } - - struct audio_patch *patch = &patchDesc->mPatch; - patchDesc->mUid = mUidCached; - if (patch->sources[0].type == AUDIO_PORT_TYPE_MIX) { - sp outputDesc = getOutputFromId(patch->sources[0].id); - if (outputDesc == NULL) { - ALOGV("releaseAudioPatch() output not found for id %d", patch->sources[0].id); - return BAD_VALUE; - } - - setOutputDevice(outputDesc->mIoHandle, - getNewOutputDevice(outputDesc->mIoHandle, true /*fromCache*/), - true, - 0, - NULL); - } else if (patch->sources[0].type == AUDIO_PORT_TYPE_DEVICE) { - if (patch->sinks[0].type == AUDIO_PORT_TYPE_MIX) { - sp inputDesc = getInputFromId(patch->sinks[0].id); - if (inputDesc == NULL) { - ALOGV("releaseAudioPatch() input not found for id %d", patch->sinks[0].id); - return BAD_VALUE; - } - setInputDevice(inputDesc->mIoHandle, - getNewInputDevice(inputDesc->mIoHandle), - true, - NULL); - } else if (patch->sinks[0].type == AUDIO_PORT_TYPE_DEVICE) { - audio_patch_handle_t afPatchHandle = patchDesc->mAfPatchHandle; - status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0); - ALOGV("releaseAudioPatch() patch panel returned %d patchHandle %d", - status, patchDesc->mAfPatchHandle); - removeAudioPatch(patchDesc->mHandle); - nextAudioPortGeneration(); - mpClientInterface->onAudioPatchListUpdate(); - } else { - return BAD_VALUE; - } - } else { - return BAD_VALUE; - } - return NO_ERROR; -} - -status_t AudioPolicyManager::listAudioPatches(unsigned int *num_patches, - struct audio_patch *patches, - unsigned int *generation) -{ - if (num_patches == NULL || (*num_patches != 0 && patches == NULL) || - generation == NULL) { - return BAD_VALUE; - } - ALOGV("listAudioPatches() num_patches %d patches %p available patches %zu", - *num_patches, patches, mAudioPatches.size()); - if (patches == NULL) { - *num_patches = 0; - } - - size_t patchesWritten = 0; - size_t patchesMax = *num_patches; - for (size_t i = 0; - i < mAudioPatches.size() && patchesWritten < patchesMax; i++) { - patches[patchesWritten] = mAudioPatches[i]->mPatch; - patches[patchesWritten++].id = mAudioPatches[i]->mHandle; - ALOGV("listAudioPatches() patch %zu num_sources %d num_sinks %d", - i, mAudioPatches[i]->mPatch.num_sources, mAudioPatches[i]->mPatch.num_sinks); - } - *num_patches = mAudioPatches.size(); - - *generation = curAudioPortGeneration(); - ALOGV("listAudioPatches() got %zu patches needed %d", patchesWritten, *num_patches); - return NO_ERROR; -} - -status_t AudioPolicyManager::setAudioPortConfig(const struct audio_port_config *config) -{ - ALOGV("setAudioPortConfig()"); - - if (config == NULL) { - return BAD_VALUE; - } - ALOGV("setAudioPortConfig() on port handle %d", config->id); - // Only support gain configuration for now - if (config->config_mask != AUDIO_PORT_CONFIG_GAIN) { - return INVALID_OPERATION; - } - - sp audioPortConfig; - if (config->type == AUDIO_PORT_TYPE_MIX) { - if (config->role == AUDIO_PORT_ROLE_SOURCE) { - sp outputDesc = getOutputFromId(config->id); - if (outputDesc == NULL) { - return BAD_VALUE; - } - ALOG_ASSERT(!outputDesc->isDuplicated(), - "setAudioPortConfig() called on duplicated output %d", - outputDesc->mIoHandle); - audioPortConfig = outputDesc; - } else if (config->role == AUDIO_PORT_ROLE_SINK) { - sp inputDesc = getInputFromId(config->id); - if (inputDesc == NULL) { - return BAD_VALUE; - } - audioPortConfig = inputDesc; - } else { - return BAD_VALUE; - } - } else if (config->type == AUDIO_PORT_TYPE_DEVICE) { - sp deviceDesc; - if (config->role == AUDIO_PORT_ROLE_SOURCE) { - deviceDesc = mAvailableInputDevices.getDeviceFromId(config->id); - } else if (config->role == AUDIO_PORT_ROLE_SINK) { - deviceDesc = mAvailableOutputDevices.getDeviceFromId(config->id); - } else { - return BAD_VALUE; - } - if (deviceDesc == NULL) { - return BAD_VALUE; - } - audioPortConfig = deviceDesc; - } else { - return BAD_VALUE; - } - - struct audio_port_config backupConfig; - status_t status = audioPortConfig->applyAudioPortConfig(config, &backupConfig); - if (status == NO_ERROR) { - struct audio_port_config newConfig; - audioPortConfig->toAudioPortConfig(&newConfig, config); - status = mpClientInterface->setAudioPortConfig(&newConfig, 0); - } - if (status != NO_ERROR) { - audioPortConfig->applyAudioPortConfig(&backupConfig); - } - - return status; -} - -void AudioPolicyManager::clearAudioPatches(uid_t uid) -{ - for (ssize_t i = (ssize_t)mAudioPatches.size() - 1; i >= 0; i--) { - sp patchDesc = mAudioPatches.valueAt(i); - if (patchDesc->mUid == uid) { - releaseAudioPatch(mAudioPatches.keyAt(i), uid); - } - } -} - -status_t AudioPolicyManager::acquireSoundTriggerSession(audio_session_t *session, - audio_io_handle_t *ioHandle, - audio_devices_t *device) -{ - *session = (audio_session_t)mpClientInterface->newAudioUniqueId(); - *ioHandle = (audio_io_handle_t)mpClientInterface->newAudioUniqueId(); - *device = getDeviceAndMixForInputSource(AUDIO_SOURCE_HOTWORD); - - mSoundTriggerSessions.add(*session, *ioHandle); - - return NO_ERROR; -} - -status_t AudioPolicyManager::releaseSoundTriggerSession(audio_session_t session) -{ - ssize_t index = mSoundTriggerSessions.indexOfKey(session); - if (index < 0) { - ALOGW("acquireSoundTriggerSession() session %d not registered", session); - return BAD_VALUE; - } - - mSoundTriggerSessions.removeItem(session); - return NO_ERROR; -} - -status_t AudioPolicyManager::addAudioPatch(audio_patch_handle_t handle, - const sp& patch) -{ - ssize_t index = mAudioPatches.indexOfKey(handle); - - if (index >= 0) { - ALOGW("addAudioPatch() patch %d already in", handle); - return ALREADY_EXISTS; - } - mAudioPatches.add(handle, patch); - ALOGV("addAudioPatch() handle %d af handle %d num_sources %d num_sinks %d source handle %d" - "sink handle %d", - handle, patch->mAfPatchHandle, patch->mPatch.num_sources, patch->mPatch.num_sinks, - patch->mPatch.sources[0].id, patch->mPatch.sinks[0].id); - return NO_ERROR; -} - -status_t AudioPolicyManager::removeAudioPatch(audio_patch_handle_t handle) -{ - ssize_t index = mAudioPatches.indexOfKey(handle); - - if (index < 0) { - ALOGW("removeAudioPatch() patch %d not in", handle); - return ALREADY_EXISTS; - } - ALOGV("removeAudioPatch() handle %d af handle %d", handle, - mAudioPatches.valueAt(index)->mAfPatchHandle); - mAudioPatches.removeItemsAt(index); - return NO_ERROR; -} - -// ---------------------------------------------------------------------------- -// AudioPolicyManager -// ---------------------------------------------------------------------------- - -uint32_t AudioPolicyManager::nextUniqueId() -{ - return android_atomic_inc(&mNextUniqueId); -} - -uint32_t AudioPolicyManager::nextAudioPortGeneration() -{ - return android_atomic_inc(&mAudioPortGeneration); -} - -int32_t volatile AudioPolicyManager::mNextUniqueId = 1; - -AudioPolicyManager::AudioPolicyManager(AudioPolicyClientInterface *clientInterface) - : -#ifdef AUDIO_POLICY_TEST - Thread(false), -#endif //AUDIO_POLICY_TEST - mPrimaryOutput((audio_io_handle_t)0), - mPhoneState(AUDIO_MODE_NORMAL), - mLimitRingtoneVolume(false), mLastVoiceVolume(-1.0f), - mTotalEffectsCpuLoad(0), mTotalEffectsMemory(0), - mA2dpSuspended(false), - mSpeakerDrcEnabled(false), - mAudioPortGeneration(1), - mBeaconMuteRefCount(0), - mBeaconPlayingRefCount(0), - mBeaconMuted(false) -{ - mUidCached = getuid(); - mpClientInterface = clientInterface; - - for (int i = 0; i < AUDIO_POLICY_FORCE_USE_CNT; i++) { - mForceUse[i] = AUDIO_POLICY_FORCE_NONE; - } - - mDefaultOutputDevice = new DeviceDescriptor(String8("Speaker"), AUDIO_DEVICE_OUT_SPEAKER); - if (loadAudioPolicyConfig(AUDIO_POLICY_VENDOR_CONFIG_FILE) != NO_ERROR) { - if (loadAudioPolicyConfig(AUDIO_POLICY_CONFIG_FILE) != NO_ERROR) { - ALOGE("could not load audio policy configuration file, setting defaults"); - defaultAudioPolicyConfig(); - } - } - // mAvailableOutputDevices and mAvailableInputDevices now contain all attached devices - - // must be done after reading the policy - initializeVolumeCurves(); - - // open all output streams needed to access attached devices - audio_devices_t outputDeviceTypes = mAvailableOutputDevices.types(); - audio_devices_t inputDeviceTypes = mAvailableInputDevices.types() & ~AUDIO_DEVICE_BIT_IN; - for (size_t i = 0; i < mHwModules.size(); i++) { - mHwModules[i]->mHandle = mpClientInterface->loadHwModule(mHwModules[i]->mName); - if (mHwModules[i]->mHandle == 0) { - ALOGW("could not open HW module %s", mHwModules[i]->mName); - continue; - } - // open all output streams needed to access attached devices - // except for direct output streams that are only opened when they are actually - // required by an app. - // This also validates mAvailableOutputDevices list - for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++) - { - const sp outProfile = mHwModules[i]->mOutputProfiles[j]; - - if (outProfile->mSupportedDevices.isEmpty()) { - ALOGW("Output profile contains no device on module %s", mHwModules[i]->mName); - continue; - } - - if ((outProfile->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) { - continue; - } - audio_devices_t profileType = outProfile->mSupportedDevices.types(); - if ((profileType & mDefaultOutputDevice->mDeviceType) != AUDIO_DEVICE_NONE) { - profileType = mDefaultOutputDevice->mDeviceType; - } else { - // chose first device present in mSupportedDevices also part of - // outputDeviceTypes - for (size_t k = 0; k < outProfile->mSupportedDevices.size(); k++) { - profileType = outProfile->mSupportedDevices[k]->mDeviceType; - if ((profileType & outputDeviceTypes) != 0) { - break; - } - } - } - if ((profileType & outputDeviceTypes) == 0) { - continue; - } - sp outputDesc = new AudioOutputDescriptor(outProfile); - - outputDesc->mDevice = profileType; - audio_config_t config = AUDIO_CONFIG_INITIALIZER; - config.sample_rate = outputDesc->mSamplingRate; - config.channel_mask = outputDesc->mChannelMask; - config.format = outputDesc->mFormat; - audio_io_handle_t output = AUDIO_IO_HANDLE_NONE; - status_t status = mpClientInterface->openOutput(outProfile->mModule->mHandle, - &output, - &config, - &outputDesc->mDevice, - String8(""), - &outputDesc->mLatency, - outputDesc->mFlags); - - if (status != NO_ERROR) { - ALOGW("Cannot open output stream for device %08x on hw module %s", - outputDesc->mDevice, - mHwModules[i]->mName); - } else { - outputDesc->mSamplingRate = config.sample_rate; - outputDesc->mChannelMask = config.channel_mask; - outputDesc->mFormat = config.format; - - for (size_t k = 0; k < outProfile->mSupportedDevices.size(); k++) { - audio_devices_t type = outProfile->mSupportedDevices[k]->mDeviceType; - ssize_t index = - mAvailableOutputDevices.indexOf(outProfile->mSupportedDevices[k]); - // give a valid ID to an attached device once confirmed it is reachable - if (index >= 0 && !mAvailableOutputDevices[index]->isAttached()) { - mAvailableOutputDevices[index]->attach(mHwModules[i]); - } - } - if (mPrimaryOutput == 0 && - outProfile->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY) { - mPrimaryOutput = output; - } - addOutput(output, outputDesc); - setOutputDevice(output, - outputDesc->mDevice, - true); - } - } - // open input streams needed to access attached devices to validate - // mAvailableInputDevices list - for (size_t j = 0; j < mHwModules[i]->mInputProfiles.size(); j++) - { - const sp inProfile = mHwModules[i]->mInputProfiles[j]; - - if (inProfile->mSupportedDevices.isEmpty()) { - ALOGW("Input profile contains no device on module %s", mHwModules[i]->mName); - continue; - } - // chose first device present in mSupportedDevices also part of - // inputDeviceTypes - audio_devices_t profileType = AUDIO_DEVICE_NONE; - for (size_t k = 0; k < inProfile->mSupportedDevices.size(); k++) { - profileType = inProfile->mSupportedDevices[k]->mDeviceType; - if (profileType & inputDeviceTypes) { - break; - } - } - if ((profileType & inputDeviceTypes) == 0) { - continue; - } - sp inputDesc = new AudioInputDescriptor(inProfile); - - inputDesc->mInputSource = AUDIO_SOURCE_MIC; - inputDesc->mDevice = profileType; - - // find the address - DeviceVector inputDevices = mAvailableInputDevices.getDevicesFromType(profileType); - // the inputs vector must be of size 1, but we don't want to crash here - String8 address = inputDevices.size() > 0 ? inputDevices.itemAt(0)->mAddress - : String8(""); - ALOGV(" for input device 0x%x using address %s", profileType, address.string()); - ALOGE_IF(inputDevices.size() == 0, "Input device list is empty!"); - - audio_config_t config = AUDIO_CONFIG_INITIALIZER; - config.sample_rate = inputDesc->mSamplingRate; - config.channel_mask = inputDesc->mChannelMask; - config.format = inputDesc->mFormat; - audio_io_handle_t input = AUDIO_IO_HANDLE_NONE; - status_t status = mpClientInterface->openInput(inProfile->mModule->mHandle, - &input, - &config, - &inputDesc->mDevice, - address, - AUDIO_SOURCE_MIC, - AUDIO_INPUT_FLAG_NONE); - - if (status == NO_ERROR) { - for (size_t k = 0; k < inProfile->mSupportedDevices.size(); k++) { - audio_devices_t type = inProfile->mSupportedDevices[k]->mDeviceType; - ssize_t index = - mAvailableInputDevices.indexOf(inProfile->mSupportedDevices[k]); - // give a valid ID to an attached device once confirmed it is reachable - if (index >= 0 && !mAvailableInputDevices[index]->isAttached()) { - mAvailableInputDevices[index]->attach(mHwModules[i]); - } - } - mpClientInterface->closeInput(input); - } else { - ALOGW("Cannot open input stream for device %08x on hw module %s", - inputDesc->mDevice, - mHwModules[i]->mName); - } - } - } - // make sure all attached devices have been allocated a unique ID - for (size_t i = 0; i < mAvailableOutputDevices.size();) { - if (!mAvailableOutputDevices[i]->isAttached()) { - ALOGW("Input device %08x unreachable", mAvailableOutputDevices[i]->mDeviceType); - mAvailableOutputDevices.remove(mAvailableOutputDevices[i]); - continue; - } - i++; - } - for (size_t i = 0; i < mAvailableInputDevices.size();) { - if (!mAvailableInputDevices[i]->isAttached()) { - ALOGW("Input device %08x unreachable", mAvailableInputDevices[i]->mDeviceType); - mAvailableInputDevices.remove(mAvailableInputDevices[i]); - continue; - } - i++; - } - // make sure default device is reachable - if (mAvailableOutputDevices.indexOf(mDefaultOutputDevice) < 0) { - ALOGE("Default device %08x is unreachable", mDefaultOutputDevice->mDeviceType); - } - - ALOGE_IF((mPrimaryOutput == 0), "Failed to open primary output"); - - updateDevicesAndOutputs(); - -#ifdef AUDIO_POLICY_TEST - if (mPrimaryOutput != 0) { - AudioParameter outputCmd = AudioParameter(); - outputCmd.addInt(String8("set_id"), 0); - mpClientInterface->setParameters(mPrimaryOutput, outputCmd.toString()); - - mTestDevice = AUDIO_DEVICE_OUT_SPEAKER; - mTestSamplingRate = 44100; - mTestFormat = AUDIO_FORMAT_PCM_16_BIT; - mTestChannels = AUDIO_CHANNEL_OUT_STEREO; - mTestLatencyMs = 0; - mCurOutput = 0; - mDirectOutput = false; - for (int i = 0; i < NUM_TEST_OUTPUTS; i++) { - mTestOutputs[i] = 0; - } - - const size_t SIZE = 256; - char buffer[SIZE]; - snprintf(buffer, SIZE, "AudioPolicyManagerTest"); - run(buffer, ANDROID_PRIORITY_AUDIO); - } -#endif //AUDIO_POLICY_TEST -} - -AudioPolicyManager::~AudioPolicyManager() -{ -#ifdef AUDIO_POLICY_TEST - exit(); -#endif //AUDIO_POLICY_TEST - for (size_t i = 0; i < mOutputs.size(); i++) { - mpClientInterface->closeOutput(mOutputs.keyAt(i)); - } - for (size_t i = 0; i < mInputs.size(); i++) { - mpClientInterface->closeInput(mInputs.keyAt(i)); - } - mAvailableOutputDevices.clear(); - mAvailableInputDevices.clear(); - mOutputs.clear(); - mInputs.clear(); - mHwModules.clear(); -} - -status_t AudioPolicyManager::initCheck() -{ - return (mPrimaryOutput == 0) ? NO_INIT : NO_ERROR; -} - -#ifdef AUDIO_POLICY_TEST -bool AudioPolicyManager::threadLoop() -{ - ALOGV("entering threadLoop()"); - while (!exitPending()) - { - String8 command; - int valueInt; - String8 value; - - Mutex::Autolock _l(mLock); - mWaitWorkCV.waitRelative(mLock, milliseconds(50)); - - command = mpClientInterface->getParameters(0, String8("test_cmd_policy")); - AudioParameter param = AudioParameter(command); - - if (param.getInt(String8("test_cmd_policy"), valueInt) == NO_ERROR && - valueInt != 0) { - ALOGV("Test command %s received", command.string()); - String8 target; - if (param.get(String8("target"), target) != NO_ERROR) { - target = "Manager"; - } - if (param.getInt(String8("test_cmd_policy_output"), valueInt) == NO_ERROR) { - param.remove(String8("test_cmd_policy_output")); - mCurOutput = valueInt; - } - if (param.get(String8("test_cmd_policy_direct"), value) == NO_ERROR) { - param.remove(String8("test_cmd_policy_direct")); - if (value == "false") { - mDirectOutput = false; - } else if (value == "true") { - mDirectOutput = true; - } - } - if (param.getInt(String8("test_cmd_policy_input"), valueInt) == NO_ERROR) { - param.remove(String8("test_cmd_policy_input")); - mTestInput = valueInt; - } - - if (param.get(String8("test_cmd_policy_format"), value) == NO_ERROR) { - param.remove(String8("test_cmd_policy_format")); - int format = AUDIO_FORMAT_INVALID; - if (value == "PCM 16 bits") { - format = AUDIO_FORMAT_PCM_16_BIT; - } else if (value == "PCM 8 bits") { - format = AUDIO_FORMAT_PCM_8_BIT; - } else if (value == "Compressed MP3") { - format = AUDIO_FORMAT_MP3; - } - if (format != AUDIO_FORMAT_INVALID) { - if (target == "Manager") { - mTestFormat = format; - } else if (mTestOutputs[mCurOutput] != 0) { - AudioParameter outputParam = AudioParameter(); - outputParam.addInt(String8("format"), format); - mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString()); - } - } - } - if (param.get(String8("test_cmd_policy_channels"), value) == NO_ERROR) { - param.remove(String8("test_cmd_policy_channels")); - int channels = 0; - - if (value == "Channels Stereo") { - channels = AUDIO_CHANNEL_OUT_STEREO; - } else if (value == "Channels Mono") { - channels = AUDIO_CHANNEL_OUT_MONO; - } - if (channels != 0) { - if (target == "Manager") { - mTestChannels = channels; - } else if (mTestOutputs[mCurOutput] != 0) { - AudioParameter outputParam = AudioParameter(); - outputParam.addInt(String8("channels"), channels); - mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString()); - } - } - } - if (param.getInt(String8("test_cmd_policy_sampleRate"), valueInt) == NO_ERROR) { - param.remove(String8("test_cmd_policy_sampleRate")); - if (valueInt >= 0 && valueInt <= 96000) { - int samplingRate = valueInt; - if (target == "Manager") { - mTestSamplingRate = samplingRate; - } else if (mTestOutputs[mCurOutput] != 0) { - AudioParameter outputParam = AudioParameter(); - outputParam.addInt(String8("sampling_rate"), samplingRate); - mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString()); - } - } - } - - if (param.get(String8("test_cmd_policy_reopen"), value) == NO_ERROR) { - param.remove(String8("test_cmd_policy_reopen")); - - sp outputDesc = mOutputs.valueFor(mPrimaryOutput); - mpClientInterface->closeOutput(mPrimaryOutput); - - audio_module_handle_t moduleHandle = outputDesc->mModule->mHandle; - - mOutputs.removeItem(mPrimaryOutput); - - sp outputDesc = new AudioOutputDescriptor(NULL); - outputDesc->mDevice = AUDIO_DEVICE_OUT_SPEAKER; - audio_config_t config = AUDIO_CONFIG_INITIALIZER; - config.sample_rate = outputDesc->mSamplingRate; - config.channel_mask = outputDesc->mChannelMask; - config.format = outputDesc->mFormat; - status_t status = mpClientInterface->openOutput(moduleHandle, - &mPrimaryOutput, - &config, - &outputDesc->mDevice, - String8(""), - &outputDesc->mLatency, - outputDesc->mFlags); - if (status != NO_ERROR) { - ALOGE("Failed to reopen hardware output stream, " - "samplingRate: %d, format %d, channels %d", - outputDesc->mSamplingRate, outputDesc->mFormat, outputDesc->mChannelMask); - } else { - outputDesc->mSamplingRate = config.sample_rate; - outputDesc->mChannelMask = config.channel_mask; - outputDesc->mFormat = config.format; - AudioParameter outputCmd = AudioParameter(); - outputCmd.addInt(String8("set_id"), 0); - mpClientInterface->setParameters(mPrimaryOutput, outputCmd.toString()); - addOutput(mPrimaryOutput, outputDesc); - } - } - - - mpClientInterface->setParameters(0, String8("test_cmd_policy=")); - } - } - return false; -} - -void AudioPolicyManager::exit() -{ - { - AutoMutex _l(mLock); - requestExit(); - mWaitWorkCV.signal(); - } - requestExitAndWait(); -} - -int AudioPolicyManager::testOutputIndex(audio_io_handle_t output) -{ - for (int i = 0; i < NUM_TEST_OUTPUTS; i++) { - if (output == mTestOutputs[i]) return i; - } - return 0; -} -#endif //AUDIO_POLICY_TEST - -// --- - -void AudioPolicyManager::addOutput(audio_io_handle_t output, sp outputDesc) -{ - outputDesc->mIoHandle = output; - outputDesc->mId = nextUniqueId(); - mOutputs.add(output, outputDesc); - nextAudioPortGeneration(); -} - -void AudioPolicyManager::addInput(audio_io_handle_t input, sp inputDesc) -{ - inputDesc->mIoHandle = input; - inputDesc->mId = nextUniqueId(); - mInputs.add(input, inputDesc); - nextAudioPortGeneration(); -} - -void AudioPolicyManager::findIoHandlesByAddress(sp desc /*in*/, - const audio_devices_t device /*in*/, - const String8 address /*in*/, - SortedVector& outputs /*out*/) { - sp devDesc = - desc->mProfile->mSupportedDevices.getDevice(device, address); - if (devDesc != 0) { - ALOGV("findIoHandlesByAddress(): adding opened output %d on same address %s", - desc->mIoHandle, address.string()); - outputs.add(desc->mIoHandle); - } -} - -status_t AudioPolicyManager::checkOutputsForDevice(const sp devDesc, - audio_policy_dev_state_t state, - SortedVector& outputs, - const String8 address) -{ - audio_devices_t device = devDesc->mDeviceType; - sp desc; - // erase all current sample rates, formats and channel masks - devDesc->clearCapabilities(); - - if (state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) { - // first list already open outputs that can be routed to this device - for (size_t i = 0; i < mOutputs.size(); i++) { - desc = mOutputs.valueAt(i); - if (!desc->isDuplicated() && (desc->mProfile->mSupportedDevices.types() & device)) { - if (!deviceDistinguishesOnAddress(device)) { - ALOGV("checkOutputsForDevice(): adding opened output %d", mOutputs.keyAt(i)); - outputs.add(mOutputs.keyAt(i)); - } else { - ALOGV(" checking address match due to device 0x%x", device); - findIoHandlesByAddress(desc, device, address, outputs); - } - } - } - // then look for output profiles that can be routed to this device - SortedVector< sp > profiles; - for (size_t i = 0; i < mHwModules.size(); i++) - { - if (mHwModules[i]->mHandle == 0) { - continue; - } - for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++) - { - sp profile = mHwModules[i]->mOutputProfiles[j]; - if (profile->mSupportedDevices.types() & device) { - if (!deviceDistinguishesOnAddress(device) || - address == profile->mSupportedDevices[0]->mAddress) { - profiles.add(profile); - ALOGV("checkOutputsForDevice(): adding profile %zu from module %zu", j, i); - } - } - } - } - - ALOGV(" found %d profiles, %d outputs", profiles.size(), outputs.size()); - - if (profiles.isEmpty() && outputs.isEmpty()) { - ALOGW("checkOutputsForDevice(): No output available for device %04x", device); - return BAD_VALUE; - } - - // open outputs for matching profiles if needed. Direct outputs are also opened to - // query for dynamic parameters and will be closed later by setDeviceConnectionState() - for (ssize_t profile_index = 0; profile_index < (ssize_t)profiles.size(); profile_index++) { - sp profile = profiles[profile_index]; - - // nothing to do if one output is already opened for this profile - size_t j; - for (j = 0; j < outputs.size(); j++) { - desc = mOutputs.valueFor(outputs.itemAt(j)); - if (!desc->isDuplicated() && desc->mProfile == profile) { - // matching profile: save the sample rates, format and channel masks supported - // by the profile in our device descriptor - devDesc->importAudioPort(profile); - break; - } - } - if (j != outputs.size()) { - continue; - } - - ALOGV("opening output for device %08x with params %s profile %p", - device, address.string(), profile.get()); - desc = new AudioOutputDescriptor(profile); - desc->mDevice = device; - audio_config_t config = AUDIO_CONFIG_INITIALIZER; - config.sample_rate = desc->mSamplingRate; - config.channel_mask = desc->mChannelMask; - config.format = desc->mFormat; - config.offload_info.sample_rate = desc->mSamplingRate; - config.offload_info.channel_mask = desc->mChannelMask; - config.offload_info.format = desc->mFormat; - audio_io_handle_t output = AUDIO_IO_HANDLE_NONE; - status_t status = mpClientInterface->openOutput(profile->mModule->mHandle, - &output, - &config, - &desc->mDevice, - address, - &desc->mLatency, - desc->mFlags); - if (status == NO_ERROR) { - desc->mSamplingRate = config.sample_rate; - desc->mChannelMask = config.channel_mask; - desc->mFormat = config.format; - - // Here is where the out_set_parameters() for card & device gets called - if (!address.isEmpty()) { - char *param = audio_device_address_to_parameter(device, address); - mpClientInterface->setParameters(output, String8(param)); - free(param); - } - - // Here is where we step through and resolve any "dynamic" fields - String8 reply; - char *value; - if (profile->mSamplingRates[0] == 0) { - reply = mpClientInterface->getParameters(output, - String8(AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES)); - ALOGV("checkOutputsForDevice() supported sampling rates %s", - reply.string()); - value = strpbrk((char *)reply.string(), "="); - if (value != NULL) { - profile->loadSamplingRates(value + 1); - } - } - if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) { - reply = mpClientInterface->getParameters(output, - String8(AUDIO_PARAMETER_STREAM_SUP_FORMATS)); - ALOGV("checkOutputsForDevice() supported formats %s", - reply.string()); - value = strpbrk((char *)reply.string(), "="); - if (value != NULL) { - profile->loadFormats(value + 1); - } - } - if (profile->mChannelMasks[0] == 0) { - reply = mpClientInterface->getParameters(output, - String8(AUDIO_PARAMETER_STREAM_SUP_CHANNELS)); - ALOGV("checkOutputsForDevice() supported channel masks %s", - reply.string()); - value = strpbrk((char *)reply.string(), "="); - if (value != NULL) { - profile->loadOutChannels(value + 1); - } - } - if (((profile->mSamplingRates[0] == 0) && - (profile->mSamplingRates.size() < 2)) || - ((profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) && - (profile->mFormats.size() < 2)) || - ((profile->mChannelMasks[0] == 0) && - (profile->mChannelMasks.size() < 2))) { - ALOGW("checkOutputsForDevice() missing param"); - mpClientInterface->closeOutput(output); - output = AUDIO_IO_HANDLE_NONE; - } else if (profile->mSamplingRates[0] == 0 || profile->mFormats[0] == 0 || - profile->mChannelMasks[0] == 0) { - mpClientInterface->closeOutput(output); - config.sample_rate = profile->pickSamplingRate(); - config.channel_mask = profile->pickChannelMask(); - config.format = profile->pickFormat(); - config.offload_info.sample_rate = config.sample_rate; - config.offload_info.channel_mask = config.channel_mask; - config.offload_info.format = config.format; - status = mpClientInterface->openOutput(profile->mModule->mHandle, - &output, - &config, - &desc->mDevice, - address, - &desc->mLatency, - desc->mFlags); - if (status == NO_ERROR) { - desc->mSamplingRate = config.sample_rate; - desc->mChannelMask = config.channel_mask; - desc->mFormat = config.format; - } else { - output = AUDIO_IO_HANDLE_NONE; - } - } - - if (output != AUDIO_IO_HANDLE_NONE) { - addOutput(output, desc); - if (deviceDistinguishesOnAddress(device) && address != "0") { - ssize_t index = mPolicyMixes.indexOfKey(address); - if (index >= 0) { - mPolicyMixes[index]->mOutput = desc; - desc->mPolicyMix = &mPolicyMixes[index]->mMix; - } else { - ALOGE("checkOutputsForDevice() cannot find policy for address %s", - address.string()); - } - } else if ((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) == 0) { - // no duplicated output for direct outputs and - // outputs used by dynamic policy mixes - audio_io_handle_t duplicatedOutput = AUDIO_IO_HANDLE_NONE; - - // set initial stream volume for device - applyStreamVolumes(output, device, 0, true); - - //TODO: configure audio effect output stage here - - // open a duplicating output thread for the new output and the primary output - duplicatedOutput = mpClientInterface->openDuplicateOutput(output, - mPrimaryOutput); - if (duplicatedOutput != AUDIO_IO_HANDLE_NONE) { - // add duplicated output descriptor - sp dupOutputDesc = - new AudioOutputDescriptor(NULL); - dupOutputDesc->mOutput1 = mOutputs.valueFor(mPrimaryOutput); - dupOutputDesc->mOutput2 = mOutputs.valueFor(output); - dupOutputDesc->mSamplingRate = desc->mSamplingRate; - dupOutputDesc->mFormat = desc->mFormat; - dupOutputDesc->mChannelMask = desc->mChannelMask; - dupOutputDesc->mLatency = desc->mLatency; - addOutput(duplicatedOutput, dupOutputDesc); - applyStreamVolumes(duplicatedOutput, device, 0, true); - } else { - ALOGW("checkOutputsForDevice() could not open dup output for %d and %d", - mPrimaryOutput, output); - mpClientInterface->closeOutput(output); - mOutputs.removeItem(output); - nextAudioPortGeneration(); - output = AUDIO_IO_HANDLE_NONE; - } - } - } - } else { - output = AUDIO_IO_HANDLE_NONE; - } - if (output == AUDIO_IO_HANDLE_NONE) { - ALOGW("checkOutputsForDevice() could not open output for device %x", device); - profiles.removeAt(profile_index); - profile_index--; - } else { - outputs.add(output); - devDesc->importAudioPort(profile); - - if (deviceDistinguishesOnAddress(device)) { - ALOGV("checkOutputsForDevice(): setOutputDevice(dev=0x%x, addr=%s)", - device, address.string()); - setOutputDevice(output, device, true/*force*/, 0/*delay*/, - NULL/*patch handle*/, address.string()); - } - ALOGV("checkOutputsForDevice(): adding output %d", output); - } - } - - if (profiles.isEmpty()) { - ALOGW("checkOutputsForDevice(): No output available for device %04x", device); - return BAD_VALUE; - } - } else { // Disconnect - // check if one opened output is not needed any more after disconnecting one device - for (size_t i = 0; i < mOutputs.size(); i++) { - desc = mOutputs.valueAt(i); - if (!desc->isDuplicated()) { - // exact match on device - if (deviceDistinguishesOnAddress(device) && - (desc->mProfile->mSupportedDevices.types() == device)) { - findIoHandlesByAddress(desc, device, address, outputs); - } else if (!(desc->mProfile->mSupportedDevices.types() - & mAvailableOutputDevices.types())) { - ALOGV("checkOutputsForDevice(): disconnecting adding output %d", - mOutputs.keyAt(i)); - outputs.add(mOutputs.keyAt(i)); - } - } - } - // Clear any profiles associated with the disconnected device. - for (size_t i = 0; i < mHwModules.size(); i++) - { - if (mHwModules[i]->mHandle == 0) { - continue; - } - for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++) - { - sp profile = mHwModules[i]->mOutputProfiles[j]; - if (profile->mSupportedDevices.types() & device) { - ALOGV("checkOutputsForDevice(): " - "clearing direct output profile %zu on module %zu", j, i); - if (profile->mSamplingRates[0] == 0) { - profile->mSamplingRates.clear(); - profile->mSamplingRates.add(0); - } - if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) { - profile->mFormats.clear(); - profile->mFormats.add(AUDIO_FORMAT_DEFAULT); - } - if (profile->mChannelMasks[0] == 0) { - profile->mChannelMasks.clear(); - profile->mChannelMasks.add(0); - } - } - } - } - } - return NO_ERROR; -} - -status_t AudioPolicyManager::checkInputsForDevice(audio_devices_t device, - audio_policy_dev_state_t state, - SortedVector& inputs, - const String8 address) -{ - sp desc; - if (state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) { - // first list already open inputs that can be routed to this device - for (size_t input_index = 0; input_index < mInputs.size(); input_index++) { - desc = mInputs.valueAt(input_index); - if (desc->mProfile->mSupportedDevices.types() & (device & ~AUDIO_DEVICE_BIT_IN)) { - ALOGV("checkInputsForDevice(): adding opened input %d", mInputs.keyAt(input_index)); - inputs.add(mInputs.keyAt(input_index)); - } - } - - // then look for input profiles that can be routed to this device - SortedVector< sp > profiles; - for (size_t module_idx = 0; module_idx < mHwModules.size(); module_idx++) - { - if (mHwModules[module_idx]->mHandle == 0) { - continue; - } - for (size_t profile_index = 0; - profile_index < mHwModules[module_idx]->mInputProfiles.size(); - profile_index++) - { - sp profile = mHwModules[module_idx]->mInputProfiles[profile_index]; - - if (profile->mSupportedDevices.types() & (device & ~AUDIO_DEVICE_BIT_IN)) { - if (!deviceDistinguishesOnAddress(device) || - address == profile->mSupportedDevices[0]->mAddress) { - profiles.add(profile); - ALOGV("checkInputsForDevice(): adding profile %zu from module %zu", - profile_index, module_idx); - } - } - } - } - - if (profiles.isEmpty() && inputs.isEmpty()) { - ALOGW("checkInputsForDevice(): No input available for device 0x%X", device); - return BAD_VALUE; - } - - // open inputs for matching profiles if needed. Direct inputs are also opened to - // query for dynamic parameters and will be closed later by setDeviceConnectionState() - for (ssize_t profile_index = 0; profile_index < (ssize_t)profiles.size(); profile_index++) { - - sp profile = profiles[profile_index]; - // nothing to do if one input is already opened for this profile - size_t input_index; - for (input_index = 0; input_index < mInputs.size(); input_index++) { - desc = mInputs.valueAt(input_index); - if (desc->mProfile == profile) { - break; - } - } - if (input_index != mInputs.size()) { - continue; - } - - ALOGV("opening input for device 0x%X with params %s", device, address.string()); - desc = new AudioInputDescriptor(profile); - desc->mDevice = device; - audio_config_t config = AUDIO_CONFIG_INITIALIZER; - config.sample_rate = desc->mSamplingRate; - config.channel_mask = desc->mChannelMask; - config.format = desc->mFormat; - audio_io_handle_t input = AUDIO_IO_HANDLE_NONE; - status_t status = mpClientInterface->openInput(profile->mModule->mHandle, - &input, - &config, - &desc->mDevice, - address, - AUDIO_SOURCE_MIC, - AUDIO_INPUT_FLAG_NONE /*FIXME*/); - - if (status == NO_ERROR) { - desc->mSamplingRate = config.sample_rate; - desc->mChannelMask = config.channel_mask; - desc->mFormat = config.format; - - if (!address.isEmpty()) { - char *param = audio_device_address_to_parameter(device, address); - mpClientInterface->setParameters(input, String8(param)); - free(param); - } - - // Here is where we step through and resolve any "dynamic" fields - String8 reply; - char *value; - if (profile->mSamplingRates[0] == 0) { - reply = mpClientInterface->getParameters(input, - String8(AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES)); - ALOGV("checkInputsForDevice() direct input sup sampling rates %s", - reply.string()); - value = strpbrk((char *)reply.string(), "="); - if (value != NULL) { - profile->loadSamplingRates(value + 1); - } - } - if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) { - reply = mpClientInterface->getParameters(input, - String8(AUDIO_PARAMETER_STREAM_SUP_FORMATS)); - ALOGV("checkInputsForDevice() direct input sup formats %s", reply.string()); - value = strpbrk((char *)reply.string(), "="); - if (value != NULL) { - profile->loadFormats(value + 1); - } - } - if (profile->mChannelMasks[0] == 0) { - reply = mpClientInterface->getParameters(input, - String8(AUDIO_PARAMETER_STREAM_SUP_CHANNELS)); - ALOGV("checkInputsForDevice() direct input sup channel masks %s", - reply.string()); - value = strpbrk((char *)reply.string(), "="); - if (value != NULL) { - profile->loadInChannels(value + 1); - } - } - if (((profile->mSamplingRates[0] == 0) && (profile->mSamplingRates.size() < 2)) || - ((profile->mFormats[0] == 0) && (profile->mFormats.size() < 2)) || - ((profile->mChannelMasks[0] == 0) && (profile->mChannelMasks.size() < 2))) { - ALOGW("checkInputsForDevice() direct input missing param"); - mpClientInterface->closeInput(input); - input = AUDIO_IO_HANDLE_NONE; - } - - if (input != 0) { - addInput(input, desc); - } - } // endif input != 0 - - if (input == AUDIO_IO_HANDLE_NONE) { - ALOGW("checkInputsForDevice() could not open input for device 0x%X", device); - profiles.removeAt(profile_index); - profile_index--; - } else { - inputs.add(input); - ALOGV("checkInputsForDevice(): adding input %d", input); - } - } // end scan profiles - - if (profiles.isEmpty()) { - ALOGW("checkInputsForDevice(): No input available for device 0x%X", device); - return BAD_VALUE; - } - } else { - // Disconnect - // check if one opened input is not needed any more after disconnecting one device - for (size_t input_index = 0; input_index < mInputs.size(); input_index++) { - desc = mInputs.valueAt(input_index); - if (!(desc->mProfile->mSupportedDevices.types() & mAvailableInputDevices.types() & - ~AUDIO_DEVICE_BIT_IN)) { - ALOGV("checkInputsForDevice(): disconnecting adding input %d", - mInputs.keyAt(input_index)); - inputs.add(mInputs.keyAt(input_index)); - } - } - // Clear any profiles associated with the disconnected device. - for (size_t module_index = 0; module_index < mHwModules.size(); module_index++) { - if (mHwModules[module_index]->mHandle == 0) { - continue; - } - for (size_t profile_index = 0; - profile_index < mHwModules[module_index]->mInputProfiles.size(); - profile_index++) { - sp profile = mHwModules[module_index]->mInputProfiles[profile_index]; - if (profile->mSupportedDevices.types() & device & ~AUDIO_DEVICE_BIT_IN) { - ALOGV("checkInputsForDevice(): clearing direct input profile %zu on module %zu", - profile_index, module_index); - if (profile->mSamplingRates[0] == 0) { - profile->mSamplingRates.clear(); - profile->mSamplingRates.add(0); - } - if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) { - profile->mFormats.clear(); - profile->mFormats.add(AUDIO_FORMAT_DEFAULT); - } - if (profile->mChannelMasks[0] == 0) { - profile->mChannelMasks.clear(); - profile->mChannelMasks.add(0); - } - } - } - } - } // end disconnect - - return NO_ERROR; -} - - -void AudioPolicyManager::closeOutput(audio_io_handle_t output) -{ - ALOGV("closeOutput(%d)", output); - - sp outputDesc = mOutputs.valueFor(output); - if (outputDesc == NULL) { - ALOGW("closeOutput() unknown output %d", output); - return; - } - - for (size_t i = 0; i < mPolicyMixes.size(); i++) { - if (mPolicyMixes[i]->mOutput == outputDesc) { - mPolicyMixes[i]->mOutput.clear(); - } - } - - // look for duplicated outputs connected to the output being removed. - for (size_t i = 0; i < mOutputs.size(); i++) { - sp dupOutputDesc = mOutputs.valueAt(i); - if (dupOutputDesc->isDuplicated() && - (dupOutputDesc->mOutput1 == outputDesc || - dupOutputDesc->mOutput2 == outputDesc)) { - sp outputDesc2; - if (dupOutputDesc->mOutput1 == outputDesc) { - outputDesc2 = dupOutputDesc->mOutput2; - } else { - outputDesc2 = dupOutputDesc->mOutput1; - } - // As all active tracks on duplicated output will be deleted, - // and as they were also referenced on the other output, the reference - // count for their stream type must be adjusted accordingly on - // the other output. - for (int j = 0; j < AUDIO_STREAM_CNT; j++) { - int refCount = dupOutputDesc->mRefCount[j]; - outputDesc2->changeRefCount((audio_stream_type_t)j,-refCount); - } - audio_io_handle_t duplicatedOutput = mOutputs.keyAt(i); - ALOGV("closeOutput() closing also duplicated output %d", duplicatedOutput); - - mpClientInterface->closeOutput(duplicatedOutput); - mOutputs.removeItem(duplicatedOutput); - } - } - - nextAudioPortGeneration(); - - ssize_t index = mAudioPatches.indexOfKey(outputDesc->mPatchHandle); - if (index >= 0) { - sp patchDesc = mAudioPatches.valueAt(index); - status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0); - mAudioPatches.removeItemsAt(index); - mpClientInterface->onAudioPatchListUpdate(); - } - - AudioParameter param; - param.add(String8("closing"), String8("true")); - mpClientInterface->setParameters(output, param.toString()); - - mpClientInterface->closeOutput(output); - mOutputs.removeItem(output); - mPreviousOutputs = mOutputs; -} - -void AudioPolicyManager::closeInput(audio_io_handle_t input) -{ - ALOGV("closeInput(%d)", input); - - sp inputDesc = mInputs.valueFor(input); - if (inputDesc == NULL) { - ALOGW("closeInput() unknown input %d", input); - return; - } - - nextAudioPortGeneration(); - - ssize_t index = mAudioPatches.indexOfKey(inputDesc->mPatchHandle); - if (index >= 0) { - sp patchDesc = mAudioPatches.valueAt(index); - status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0); - mAudioPatches.removeItemsAt(index); - mpClientInterface->onAudioPatchListUpdate(); - } - - mpClientInterface->closeInput(input); - mInputs.removeItem(input); -} - -SortedVector AudioPolicyManager::getOutputsForDevice(audio_devices_t device, - DefaultKeyedVector > openOutputs) -{ - SortedVector outputs; - - ALOGVV("getOutputsForDevice() device %04x", device); - for (size_t i = 0; i < openOutputs.size(); i++) { - ALOGVV("output %d isDuplicated=%d device=%04x", - i, openOutputs.valueAt(i)->isDuplicated(), openOutputs.valueAt(i)->supportedDevices()); - if ((device & openOutputs.valueAt(i)->supportedDevices()) == device) { - ALOGVV("getOutputsForDevice() found output %d", openOutputs.keyAt(i)); - outputs.add(openOutputs.keyAt(i)); - } - } - return outputs; -} - -bool AudioPolicyManager::vectorsEqual(SortedVector& outputs1, - SortedVector& outputs2) -{ - if (outputs1.size() != outputs2.size()) { - return false; - } - for (size_t i = 0; i < outputs1.size(); i++) { - if (outputs1[i] != outputs2[i]) { - return false; - } - } - return true; -} - -void AudioPolicyManager::checkOutputForStrategy(routing_strategy strategy) -{ - audio_devices_t oldDevice = getDeviceForStrategy(strategy, true /*fromCache*/); - audio_devices_t newDevice = getDeviceForStrategy(strategy, false /*fromCache*/); - SortedVector srcOutputs = getOutputsForDevice(oldDevice, mPreviousOutputs); - SortedVector dstOutputs = getOutputsForDevice(newDevice, mOutputs); - - // also take into account external policy-related changes: add all outputs which are - // associated with policies in the "before" and "after" output vectors - ALOGVV("checkOutputForStrategy(): policy related outputs"); - for (size_t i = 0 ; i < mPreviousOutputs.size() ; i++) { - const sp desc = mPreviousOutputs.valueAt(i); - if (desc != 0 && desc->mPolicyMix != NULL) { - srcOutputs.add(desc->mIoHandle); - ALOGVV(" previous outputs: adding %d", desc->mIoHandle); - } - } - for (size_t i = 0 ; i < mOutputs.size() ; i++) { - const sp desc = mOutputs.valueAt(i); - if (desc != 0 && desc->mPolicyMix != NULL) { - dstOutputs.add(desc->mIoHandle); - ALOGVV(" new outputs: adding %d", desc->mIoHandle); - } - } - - if (!vectorsEqual(srcOutputs,dstOutputs)) { - ALOGV("checkOutputForStrategy() strategy %d, moving from output %d to output %d", - strategy, srcOutputs[0], dstOutputs[0]); - // mute strategy while moving tracks from one output to another - for (size_t i = 0; i < srcOutputs.size(); i++) { - sp desc = mOutputs.valueFor(srcOutputs[i]); - if (desc->isStrategyActive(strategy)) { - setStrategyMute(strategy, true, srcOutputs[i]); - setStrategyMute(strategy, false, srcOutputs[i], MUTE_TIME_MS, newDevice); - } - } - - // Move effects associated to this strategy from previous output to new output - if (strategy == STRATEGY_MEDIA) { - audio_io_handle_t fxOutput = selectOutputForEffects(dstOutputs); - SortedVector moved; - for (size_t i = 0; i < mEffects.size(); i++) { - sp effectDesc = mEffects.valueAt(i); - if (effectDesc->mSession == AUDIO_SESSION_OUTPUT_MIX && - effectDesc->mIo != fxOutput) { - if (moved.indexOf(effectDesc->mIo) < 0) { - ALOGV("checkOutputForStrategy() moving effect %d to output %d", - mEffects.keyAt(i), fxOutput); - mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, effectDesc->mIo, - fxOutput); - moved.add(effectDesc->mIo); - } - effectDesc->mIo = fxOutput; - } - } - } - // Move tracks associated to this strategy from previous output to new output - for (int i = 0; i < AUDIO_STREAM_CNT; i++) { - if (i == AUDIO_STREAM_PATCH) { - continue; - } - if (getStrategy((audio_stream_type_t)i) == strategy) { - mpClientInterface->invalidateStream((audio_stream_type_t)i); - } - } - } -} - -void AudioPolicyManager::checkOutputForAllStrategies() -{ - if (mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) - checkOutputForStrategy(STRATEGY_ENFORCED_AUDIBLE); - checkOutputForStrategy(STRATEGY_PHONE); - if (mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] != AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) - checkOutputForStrategy(STRATEGY_ENFORCED_AUDIBLE); - checkOutputForStrategy(STRATEGY_SONIFICATION); - checkOutputForStrategy(STRATEGY_SONIFICATION_RESPECTFUL); - checkOutputForStrategy(STRATEGY_ACCESSIBILITY); - checkOutputForStrategy(STRATEGY_MEDIA); - checkOutputForStrategy(STRATEGY_DTMF); - checkOutputForStrategy(STRATEGY_REROUTING); -} - -audio_io_handle_t AudioPolicyManager::getA2dpOutput() -{ - for (size_t i = 0; i < mOutputs.size(); i++) { - sp outputDesc = mOutputs.valueAt(i); - if (!outputDesc->isDuplicated() && outputDesc->device() & AUDIO_DEVICE_OUT_ALL_A2DP) { - return mOutputs.keyAt(i); - } - } - - return 0; -} - -void AudioPolicyManager::checkA2dpSuspend() -{ - audio_io_handle_t a2dpOutput = getA2dpOutput(); - if (a2dpOutput == 0) { - mA2dpSuspended = false; - return; - } - - bool isScoConnected = - ((mAvailableInputDevices.types() & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET & - ~AUDIO_DEVICE_BIT_IN) != 0) || - ((mAvailableOutputDevices.types() & AUDIO_DEVICE_OUT_ALL_SCO) != 0); - // suspend A2DP output if: - // (NOT already suspended) && - // ((SCO device is connected && - // (forced usage for communication || for record is SCO))) || - // (phone state is ringing || in call) - // - // restore A2DP output if: - // (Already suspended) && - // ((SCO device is NOT connected || - // (forced usage NOT for communication && NOT for record is SCO))) && - // (phone state is NOT ringing && NOT in call) - // - if (mA2dpSuspended) { - if ((!isScoConnected || - ((mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION] != AUDIO_POLICY_FORCE_BT_SCO) && - (mForceUse[AUDIO_POLICY_FORCE_FOR_RECORD] != AUDIO_POLICY_FORCE_BT_SCO))) && - ((mPhoneState != AUDIO_MODE_IN_CALL) && - (mPhoneState != AUDIO_MODE_RINGTONE))) { - - mpClientInterface->restoreOutput(a2dpOutput); - mA2dpSuspended = false; - } - } else { - if ((isScoConnected && - ((mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION] == AUDIO_POLICY_FORCE_BT_SCO) || - (mForceUse[AUDIO_POLICY_FORCE_FOR_RECORD] == AUDIO_POLICY_FORCE_BT_SCO))) || - ((mPhoneState == AUDIO_MODE_IN_CALL) || - (mPhoneState == AUDIO_MODE_RINGTONE))) { - - mpClientInterface->suspendOutput(a2dpOutput); - mA2dpSuspended = true; - } - } -} - -audio_devices_t AudioPolicyManager::getNewOutputDevice(audio_io_handle_t output, bool fromCache) -{ - audio_devices_t device = AUDIO_DEVICE_NONE; - - sp outputDesc = mOutputs.valueFor(output); - - ssize_t index = mAudioPatches.indexOfKey(outputDesc->mPatchHandle); - if (index >= 0) { - sp patchDesc = mAudioPatches.valueAt(index); - if (patchDesc->mUid != mUidCached) { - ALOGV("getNewOutputDevice() device %08x forced by patch %d", - outputDesc->device(), outputDesc->mPatchHandle); - return outputDesc->device(); - } - } - - // check the following by order of priority to request a routing change if necessary: - // 1: the strategy enforced audible is active and enforced on the output: - // use device for strategy enforced audible - // 2: we are in call or the strategy phone is active on the output: - // use device for strategy phone - // 3: the strategy for enforced audible is active but not enforced on the output: - // use the device for strategy enforced audible - // 4: the strategy sonification is active on the output: - // use device for strategy sonification - // 5: the strategy "respectful" sonification is active on the output: - // use device for strategy "respectful" sonification - // 6: the strategy accessibility is active on the output: - // use device for strategy accessibility - // 7: the strategy media is active on the output: - // use device for strategy media - // 8: the strategy DTMF is active on the output: - // use device for strategy DTMF - // 9: the strategy for beacon, a.k.a. "transmitted through speaker" is active on the output: - // use device for strategy t-t-s - if (outputDesc->isStrategyActive(STRATEGY_ENFORCED_AUDIBLE) && - mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) { - device = getDeviceForStrategy(STRATEGY_ENFORCED_AUDIBLE, fromCache); - } else if (isInCall() || - outputDesc->isStrategyActive(STRATEGY_PHONE)) { - device = getDeviceForStrategy(STRATEGY_PHONE, fromCache); - } else if (outputDesc->isStrategyActive(STRATEGY_ENFORCED_AUDIBLE)) { - device = getDeviceForStrategy(STRATEGY_ENFORCED_AUDIBLE, fromCache); - } else if (outputDesc->isStrategyActive(STRATEGY_SONIFICATION)) { - device = getDeviceForStrategy(STRATEGY_SONIFICATION, fromCache); - } else if (outputDesc->isStrategyActive(STRATEGY_SONIFICATION_RESPECTFUL)) { - device = getDeviceForStrategy(STRATEGY_SONIFICATION_RESPECTFUL, fromCache); - } else if (outputDesc->isStrategyActive(STRATEGY_ACCESSIBILITY)) { - device = getDeviceForStrategy(STRATEGY_ACCESSIBILITY, fromCache); - } else if (outputDesc->isStrategyActive(STRATEGY_MEDIA)) { - device = getDeviceForStrategy(STRATEGY_MEDIA, fromCache); - } else if (outputDesc->isStrategyActive(STRATEGY_DTMF)) { - device = getDeviceForStrategy(STRATEGY_DTMF, fromCache); - } else if (outputDesc->isStrategyActive(STRATEGY_TRANSMITTED_THROUGH_SPEAKER)) { - device = getDeviceForStrategy(STRATEGY_TRANSMITTED_THROUGH_SPEAKER, fromCache); - } else if (outputDesc->isStrategyActive(STRATEGY_REROUTING)) { - device = getDeviceForStrategy(STRATEGY_REROUTING, fromCache); - } - - ALOGV("getNewOutputDevice() selected device %x", device); - return device; -} - -audio_devices_t AudioPolicyManager::getNewInputDevice(audio_io_handle_t input) -{ - sp inputDesc = mInputs.valueFor(input); - - ssize_t index = mAudioPatches.indexOfKey(inputDesc->mPatchHandle); - if (index >= 0) { - sp patchDesc = mAudioPatches.valueAt(index); - if (patchDesc->mUid != mUidCached) { - ALOGV("getNewInputDevice() device %08x forced by patch %d", - inputDesc->mDevice, inputDesc->mPatchHandle); - return inputDesc->mDevice; - } - } - - audio_devices_t device = getDeviceAndMixForInputSource(inputDesc->mInputSource); - - ALOGV("getNewInputDevice() selected device %x", device); - return device; -} - -uint32_t AudioPolicyManager::getStrategyForStream(audio_stream_type_t stream) { - return (uint32_t)getStrategy(stream); -} - -audio_devices_t AudioPolicyManager::getDevicesForStream(audio_stream_type_t stream) { - // By checking the range of stream before calling getStrategy, we avoid - // getStrategy's behavior for invalid streams. getStrategy would do a ALOGE - // and then return STRATEGY_MEDIA, but we want to return the empty set. - if (stream < (audio_stream_type_t) 0 || stream >= AUDIO_STREAM_PUBLIC_CNT) { - return AUDIO_DEVICE_NONE; - } - audio_devices_t devices; - AudioPolicyManager::routing_strategy strategy = getStrategy(stream); - devices = getDeviceForStrategy(strategy, true /*fromCache*/); - SortedVector outputs = getOutputsForDevice(devices, mOutputs); - for (size_t i = 0; i < outputs.size(); i++) { - sp outputDesc = mOutputs.valueFor(outputs[i]); - if (outputDesc->isStrategyActive(strategy)) { - devices = outputDesc->device(); - break; - } - } - - /*Filter SPEAKER_SAFE out of results, as AudioService doesn't know about it - and doesn't really need to.*/ - if (devices & AUDIO_DEVICE_OUT_SPEAKER_SAFE) { - devices |= AUDIO_DEVICE_OUT_SPEAKER; - devices &= ~AUDIO_DEVICE_OUT_SPEAKER_SAFE; - } - - return devices; -} - -AudioPolicyManager::routing_strategy AudioPolicyManager::getStrategy( - audio_stream_type_t stream) { - - ALOG_ASSERT(stream != AUDIO_STREAM_PATCH,"getStrategy() called for AUDIO_STREAM_PATCH"); - - // stream to strategy mapping - switch (stream) { - case AUDIO_STREAM_VOICE_CALL: - case AUDIO_STREAM_BLUETOOTH_SCO: - return STRATEGY_PHONE; - case AUDIO_STREAM_RING: - case AUDIO_STREAM_ALARM: - return STRATEGY_SONIFICATION; - case AUDIO_STREAM_NOTIFICATION: - return STRATEGY_SONIFICATION_RESPECTFUL; - case AUDIO_STREAM_DTMF: - return STRATEGY_DTMF; - default: - ALOGE("unknown stream type %d", stream); - case AUDIO_STREAM_SYSTEM: - // NOTE: SYSTEM stream uses MEDIA strategy because muting music and switching outputs - // while key clicks are played produces a poor result - case AUDIO_STREAM_MUSIC: - return STRATEGY_MEDIA; - case AUDIO_STREAM_ENFORCED_AUDIBLE: - return STRATEGY_ENFORCED_AUDIBLE; - case AUDIO_STREAM_TTS: - return STRATEGY_TRANSMITTED_THROUGH_SPEAKER; - case AUDIO_STREAM_ACCESSIBILITY: - return STRATEGY_ACCESSIBILITY; - case AUDIO_STREAM_REROUTING: - return STRATEGY_REROUTING; - } -} - -uint32_t AudioPolicyManager::getStrategyForAttr(const audio_attributes_t *attr) { - // flags to strategy mapping - if ((attr->flags & AUDIO_FLAG_BEACON) == AUDIO_FLAG_BEACON) { - return (uint32_t) STRATEGY_TRANSMITTED_THROUGH_SPEAKER; - } - if ((attr->flags & AUDIO_FLAG_AUDIBILITY_ENFORCED) == AUDIO_FLAG_AUDIBILITY_ENFORCED) { - return (uint32_t) STRATEGY_ENFORCED_AUDIBLE; - } - - // usage to strategy mapping - switch (attr->usage) { - case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY: - if (isStreamActive(AUDIO_STREAM_RING) || isStreamActive(AUDIO_STREAM_ALARM)) { - return (uint32_t) STRATEGY_SONIFICATION; - } - if (isInCall()) { - return (uint32_t) STRATEGY_PHONE; - } - return (uint32_t) STRATEGY_ACCESSIBILITY; - - case AUDIO_USAGE_MEDIA: - case AUDIO_USAGE_GAME: - case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE: - case AUDIO_USAGE_ASSISTANCE_SONIFICATION: - return (uint32_t) STRATEGY_MEDIA; - - case AUDIO_USAGE_VOICE_COMMUNICATION: - return (uint32_t) STRATEGY_PHONE; - - case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING: - return (uint32_t) STRATEGY_DTMF; - - case AUDIO_USAGE_ALARM: - case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE: - return (uint32_t) STRATEGY_SONIFICATION; - - case AUDIO_USAGE_NOTIFICATION: - case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST: - case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT: - case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED: - case AUDIO_USAGE_NOTIFICATION_EVENT: - return (uint32_t) STRATEGY_SONIFICATION_RESPECTFUL; - - case AUDIO_USAGE_UNKNOWN: - default: - return (uint32_t) STRATEGY_MEDIA; - } -} - -void AudioPolicyManager::handleNotificationRoutingForStream(audio_stream_type_t stream) { - switch(stream) { - case AUDIO_STREAM_MUSIC: - checkOutputForStrategy(STRATEGY_SONIFICATION_RESPECTFUL); - updateDevicesAndOutputs(); - break; - default: - break; - } -} - -bool AudioPolicyManager::isAnyOutputActive(audio_stream_type_t streamToIgnore) { - for (size_t s = 0 ; s < AUDIO_STREAM_CNT ; s++) { - if (s == (size_t) streamToIgnore) { - continue; - } - for (size_t i = 0; i < mOutputs.size(); i++) { - const sp outputDesc = mOutputs.valueAt(i); - if (outputDesc->mRefCount[s] != 0) { - return true; - } - } - } - return false; -} - -uint32_t AudioPolicyManager::handleEventForBeacon(int event) { - switch(event) { - case STARTING_OUTPUT: - mBeaconMuteRefCount++; - break; - case STOPPING_OUTPUT: - if (mBeaconMuteRefCount > 0) { - mBeaconMuteRefCount--; - } - break; - case STARTING_BEACON: - mBeaconPlayingRefCount++; - break; - case STOPPING_BEACON: - if (mBeaconPlayingRefCount > 0) { - mBeaconPlayingRefCount--; - } - break; - } - - if (mBeaconMuteRefCount > 0) { - // any playback causes beacon to be muted - return setBeaconMute(true); - } else { - // no other playback: unmute when beacon starts playing, mute when it stops - return setBeaconMute(mBeaconPlayingRefCount == 0); - } -} - -uint32_t AudioPolicyManager::setBeaconMute(bool mute) { - ALOGV("setBeaconMute(%d) mBeaconMuteRefCount=%d mBeaconPlayingRefCount=%d", - mute, mBeaconMuteRefCount, mBeaconPlayingRefCount); - // keep track of muted state to avoid repeating mute/unmute operations - if (mBeaconMuted != mute) { - // mute/unmute AUDIO_STREAM_TTS on all outputs - ALOGV("\t muting %d", mute); - uint32_t maxLatency = 0; - for (size_t i = 0; i < mOutputs.size(); i++) { - sp desc = mOutputs.valueAt(i); - setStreamMute(AUDIO_STREAM_TTS, mute/*on*/, - desc->mIoHandle, - 0 /*delay*/, AUDIO_DEVICE_NONE); - const uint32_t latency = desc->latency() * 2; - if (latency > maxLatency) { - maxLatency = latency; - } - } - mBeaconMuted = mute; - return maxLatency; - } - return 0; -} - -audio_devices_t AudioPolicyManager::getDeviceForStrategy(routing_strategy strategy, - bool fromCache) -{ - uint32_t device = AUDIO_DEVICE_NONE; - - if (fromCache) { - ALOGVV("getDeviceForStrategy() from cache strategy %d, device %x", - strategy, mDeviceForStrategy[strategy]); - return mDeviceForStrategy[strategy]; - } - audio_devices_t availableOutputDeviceTypes = mAvailableOutputDevices.types(); - switch (strategy) { - - case STRATEGY_TRANSMITTED_THROUGH_SPEAKER: - device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER; - if (!device) { - ALOGE("getDeviceForStrategy() no device found for "\ - "STRATEGY_TRANSMITTED_THROUGH_SPEAKER"); - } - break; - - case STRATEGY_SONIFICATION_RESPECTFUL: - if (isInCall()) { - device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/); - } else if (isStreamActiveRemotely(AUDIO_STREAM_MUSIC, - SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY)) { - // while media is playing on a remote device, use the the sonification behavior. - // Note that we test this usecase before testing if media is playing because - // the isStreamActive() method only informs about the activity of a stream, not - // if it's for local playback. Note also that we use the same delay between both tests - device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/); - //user "safe" speaker if available instead of normal speaker to avoid triggering - //other acoustic safety mechanisms for notification - if (device == AUDIO_DEVICE_OUT_SPEAKER && (availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER_SAFE)) - device = AUDIO_DEVICE_OUT_SPEAKER_SAFE; - } else if (isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY)) { - // while media is playing (or has recently played), use the same device - device = getDeviceForStrategy(STRATEGY_MEDIA, false /*fromCache*/); - } else { - // when media is not playing anymore, fall back on the sonification behavior - device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/); - //user "safe" speaker if available instead of normal speaker to avoid triggering - //other acoustic safety mechanisms for notification - if (device == AUDIO_DEVICE_OUT_SPEAKER && (availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER_SAFE)) - device = AUDIO_DEVICE_OUT_SPEAKER_SAFE; - } - - break; - - case STRATEGY_DTMF: - if (!isInCall()) { - // when off call, DTMF strategy follows the same rules as MEDIA strategy - device = getDeviceForStrategy(STRATEGY_MEDIA, false /*fromCache*/); - break; - } - // when in call, DTMF and PHONE strategies follow the same rules - // FALL THROUGH - - case STRATEGY_PHONE: - // Force use of only devices on primary output if: - // - in call AND - // - cannot route from voice call RX OR - // - audio HAL version is < 3.0 and TX device is on the primary HW module - if (mPhoneState == AUDIO_MODE_IN_CALL) { - audio_devices_t txDevice = - getDeviceAndMixForInputSource(AUDIO_SOURCE_VOICE_COMMUNICATION); - sp hwOutputDesc = mOutputs.valueFor(mPrimaryOutput); - if (((mAvailableInputDevices.types() & - AUDIO_DEVICE_IN_TELEPHONY_RX & ~AUDIO_DEVICE_BIT_IN) == 0) || - (((txDevice & availablePrimaryInputDevices() & ~AUDIO_DEVICE_BIT_IN) != 0) && - (hwOutputDesc->getAudioPort()->mModule->mHalVersion < - AUDIO_DEVICE_API_VERSION_3_0))) { - availableOutputDeviceTypes = availablePrimaryOutputDevices(); - } - } - // for phone strategy, we first consider the forced use and then the available devices by order - // of priority - switch (mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION]) { - case AUDIO_POLICY_FORCE_BT_SCO: - if (!isInCall() || strategy != STRATEGY_DTMF) { - device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT; - if (device) break; - } - device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET; - if (device) break; - device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_SCO; - if (device) break; - // if SCO device is requested but no SCO device is available, fall back to default case - // FALL THROUGH - - default: // FORCE_NONE - // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to A2DP - if (!isInCall() && - (mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] != AUDIO_POLICY_FORCE_NO_BT_A2DP) && - (getA2dpOutput() != 0)) { - device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP; - if (device) break; - device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES; - if (device) break; - } - device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_WIRED_HEADPHONE; - if (device) break; - device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_WIRED_HEADSET; - if (device) break; - device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_DEVICE; - if (device) break; - if (mPhoneState != AUDIO_MODE_IN_CALL) { - device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_ACCESSORY; - if (device) break; - device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET; - if (device) break; - device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_AUX_DIGITAL; - if (device) break; - device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET; - if (device) break; - } - device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_EARPIECE; - if (device) break; - device = mDefaultOutputDevice->mDeviceType; - if (device == AUDIO_DEVICE_NONE) { - ALOGE("getDeviceForStrategy() no device found for STRATEGY_PHONE"); - } - break; - - case AUDIO_POLICY_FORCE_SPEAKER: - // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to - // A2DP speaker when forcing to speaker output - if (!isInCall() && - (mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] != AUDIO_POLICY_FORCE_NO_BT_A2DP) && - (getA2dpOutput() != 0)) { - device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER; - if (device) break; - } - if (!isInCall()) { - device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_ACCESSORY; - if (device) break; - device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_DEVICE; - if (device) break; - device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET; - if (device) break; - device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_AUX_DIGITAL; - if (device) break; - device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET; - if (device) break; - } - device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_LINE; - if (device) break; - device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER; - if (device) break; - device = mDefaultOutputDevice->mDeviceType; - if (device == AUDIO_DEVICE_NONE) { - ALOGE("getDeviceForStrategy() no device found for STRATEGY_PHONE, FORCE_SPEAKER"); - } - break; - } - break; - - case STRATEGY_SONIFICATION: - - // If incall, just select the STRATEGY_PHONE device: The rest of the behavior is handled by - // handleIncallSonification(). - if (isInCall()) { - device = getDeviceForStrategy(STRATEGY_PHONE, false /*fromCache*/); - break; - } - // FALL THROUGH - - case STRATEGY_ENFORCED_AUDIBLE: - // strategy STRATEGY_ENFORCED_AUDIBLE uses same routing policy as STRATEGY_SONIFICATION - // except: - // - when in call where it doesn't default to STRATEGY_PHONE behavior - // - in countries where not enforced in which case it follows STRATEGY_MEDIA - - if ((strategy == STRATEGY_SONIFICATION) || - (mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED)) { - device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER; - if (device == AUDIO_DEVICE_NONE) { - ALOGE("getDeviceForStrategy() speaker device not found for STRATEGY_SONIFICATION"); - } - } - // The second device used for sonification is the same as the device used by media strategy - // FALL THROUGH - - // FIXME: STRATEGY_ACCESSIBILITY and STRATEGY_REROUTING follow STRATEGY_MEDIA for now - case STRATEGY_ACCESSIBILITY: - if (strategy == STRATEGY_ACCESSIBILITY) { - // do not route accessibility prompts to a digital output currently configured with a - // compressed format as they would likely not be mixed and dropped. - for (size_t i = 0; i < mOutputs.size(); i++) { - sp desc = mOutputs.valueAt(i); - audio_devices_t devices = desc->device() & - (AUDIO_DEVICE_OUT_HDMI | AUDIO_DEVICE_OUT_SPDIF | AUDIO_DEVICE_OUT_HDMI_ARC); - if (desc->isActive() && !audio_is_linear_pcm(desc->mFormat) && - devices != AUDIO_DEVICE_NONE) { - availableOutputDeviceTypes = availableOutputDeviceTypes & ~devices; - } - } - } - // FALL THROUGH - - case STRATEGY_REROUTING: - case STRATEGY_MEDIA: { - uint32_t device2 = AUDIO_DEVICE_NONE; - if (strategy != STRATEGY_SONIFICATION) { - // no sonification on remote submix (e.g. WFD) - if (mAvailableOutputDevices.getDevice(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, String8("0")) != 0) { - device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_REMOTE_SUBMIX; - } - } - if ((device2 == AUDIO_DEVICE_NONE) && - (mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] != AUDIO_POLICY_FORCE_NO_BT_A2DP) && - (getA2dpOutput() != 0)) { - device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP; - if (device2 == AUDIO_DEVICE_NONE) { - device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES; - } - if (device2 == AUDIO_DEVICE_NONE) { - device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER; - } - } - if ((device2 == AUDIO_DEVICE_NONE) && - (mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] == AUDIO_POLICY_FORCE_SPEAKER)) { - device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER; - } - if (device2 == AUDIO_DEVICE_NONE) { - device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_WIRED_HEADPHONE; - } - if ((device2 == AUDIO_DEVICE_NONE)) { - device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_LINE; - } - if (device2 == AUDIO_DEVICE_NONE) { - device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_WIRED_HEADSET; - } - if (device2 == AUDIO_DEVICE_NONE) { - device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_ACCESSORY; - } - if (device2 == AUDIO_DEVICE_NONE) { - device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_DEVICE; - } - if (device2 == AUDIO_DEVICE_NONE) { - device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET; - } - if ((device2 == AUDIO_DEVICE_NONE) && (strategy != STRATEGY_SONIFICATION)) { - // no sonification on aux digital (e.g. HDMI) - device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_AUX_DIGITAL; - } - if ((device2 == AUDIO_DEVICE_NONE) && - (mForceUse[AUDIO_POLICY_FORCE_FOR_DOCK] == AUDIO_POLICY_FORCE_ANALOG_DOCK)) { - device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET; - } - if (device2 == AUDIO_DEVICE_NONE) { - device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER; - } - int device3 = AUDIO_DEVICE_NONE; - if (strategy == STRATEGY_MEDIA) { - // ARC, SPDIF and AUX_LINE can co-exist with others. - device3 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_HDMI_ARC; - device3 |= (availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPDIF); - device3 |= (availableOutputDeviceTypes & AUDIO_DEVICE_OUT_AUX_LINE); - } - - device2 |= device3; - // device is DEVICE_OUT_SPEAKER if we come from case STRATEGY_SONIFICATION or - // STRATEGY_ENFORCED_AUDIBLE, AUDIO_DEVICE_NONE otherwise - device |= device2; - - // If hdmi system audio mode is on, remove speaker out of output list. - if ((strategy == STRATEGY_MEDIA) && - (mForceUse[AUDIO_POLICY_FORCE_FOR_HDMI_SYSTEM_AUDIO] == - AUDIO_POLICY_FORCE_HDMI_SYSTEM_AUDIO_ENFORCED)) { - device &= ~AUDIO_DEVICE_OUT_SPEAKER; - } - - if (device) break; - device = mDefaultOutputDevice->mDeviceType; - if (device == AUDIO_DEVICE_NONE) { - ALOGE("getDeviceForStrategy() no device found for STRATEGY_MEDIA"); - } - } break; - - default: - ALOGW("getDeviceForStrategy() unknown strategy: %d", strategy); - break; - } - - ALOGVV("getDeviceForStrategy() strategy %d, device %x", strategy, device); - return device; -} - -void AudioPolicyManager::updateDevicesAndOutputs() -{ - for (int i = 0; i < NUM_STRATEGIES; i++) { - mDeviceForStrategy[i] = getDeviceForStrategy((routing_strategy)i, false /*fromCache*/); - } - mPreviousOutputs = mOutputs; -} - -uint32_t AudioPolicyManager::checkDeviceMuteStrategies(sp outputDesc, - audio_devices_t prevDevice, - uint32_t delayMs) -{ - // mute/unmute strategies using an incompatible device combination - // if muting, wait for the audio in pcm buffer to be drained before proceeding - // if unmuting, unmute only after the specified delay - if (outputDesc->isDuplicated()) { - return 0; - } - - uint32_t muteWaitMs = 0; - audio_devices_t device = outputDesc->device(); - bool shouldMute = outputDesc->isActive() && (popcount(device) >= 2); - - for (size_t i = 0; i < NUM_STRATEGIES; i++) { - audio_devices_t curDevice = getDeviceForStrategy((routing_strategy)i, false /*fromCache*/); - curDevice = curDevice & outputDesc->mProfile->mSupportedDevices.types(); - bool mute = shouldMute && (curDevice & device) && (curDevice != device); - bool doMute = false; - - if (mute && !outputDesc->mStrategyMutedByDevice[i]) { - doMute = true; - outputDesc->mStrategyMutedByDevice[i] = true; - } else if (!mute && outputDesc->mStrategyMutedByDevice[i]){ - doMute = true; - outputDesc->mStrategyMutedByDevice[i] = false; - } - if (doMute) { - for (size_t j = 0; j < mOutputs.size(); j++) { - sp desc = mOutputs.valueAt(j); - // skip output if it does not share any device with current output - if ((desc->supportedDevices() & outputDesc->supportedDevices()) - == AUDIO_DEVICE_NONE) { - continue; - } - audio_io_handle_t curOutput = mOutputs.keyAt(j); - ALOGVV("checkDeviceMuteStrategies() %s strategy %d (curDevice %04x) on output %d", - mute ? "muting" : "unmuting", i, curDevice, curOutput); - setStrategyMute((routing_strategy)i, mute, curOutput, mute ? 0 : delayMs); - if (desc->isStrategyActive((routing_strategy)i)) { - if (mute) { - // FIXME: should not need to double latency if volume could be applied - // immediately by the audioflinger mixer. We must account for the delay - // between now and the next time the audioflinger thread for this output - // will process a buffer (which corresponds to one buffer size, - // usually 1/2 or 1/4 of the latency). - if (muteWaitMs < desc->latency() * 2) { - muteWaitMs = desc->latency() * 2; - } - } - } - } - } - } - - // temporary mute output if device selection changes to avoid volume bursts due to - // different per device volumes - if (outputDesc->isActive() && (device != prevDevice)) { - if (muteWaitMs < outputDesc->latency() * 2) { - muteWaitMs = outputDesc->latency() * 2; - } - for (size_t i = 0; i < NUM_STRATEGIES; i++) { - if (outputDesc->isStrategyActive((routing_strategy)i)) { - setStrategyMute((routing_strategy)i, true, outputDesc->mIoHandle); - // do tempMute unmute after twice the mute wait time - setStrategyMute((routing_strategy)i, false, outputDesc->mIoHandle, - muteWaitMs *2, device); - } - } - } - - // wait for the PCM output buffers to empty before proceeding with the rest of the command - if (muteWaitMs > delayMs) { - muteWaitMs -= delayMs; - usleep(muteWaitMs * 1000); - return muteWaitMs; - } - return 0; -} - -uint32_t AudioPolicyManager::setOutputDevice(audio_io_handle_t output, - audio_devices_t device, - bool force, - int delayMs, - audio_patch_handle_t *patchHandle, - const char* address) -{ - ALOGV("setOutputDevice() output %d device %04x delayMs %d", output, device, delayMs); - sp outputDesc = mOutputs.valueFor(output); - AudioParameter param; - uint32_t muteWaitMs; - - if (outputDesc->isDuplicated()) { - muteWaitMs = setOutputDevice(outputDesc->mOutput1->mIoHandle, device, force, delayMs); - muteWaitMs += setOutputDevice(outputDesc->mOutput2->mIoHandle, device, force, delayMs); - return muteWaitMs; - } - // no need to proceed if new device is not AUDIO_DEVICE_NONE and not supported by current - // output profile - if ((device != AUDIO_DEVICE_NONE) && - ((device & outputDesc->mProfile->mSupportedDevices.types()) == 0)) { - return 0; - } - - // filter devices according to output selected - device = (audio_devices_t)(device & outputDesc->mProfile->mSupportedDevices.types()); - - audio_devices_t prevDevice = outputDesc->mDevice; - - ALOGV("setOutputDevice() prevDevice %04x", prevDevice); - - if (device != AUDIO_DEVICE_NONE) { - outputDesc->mDevice = device; - } - muteWaitMs = checkDeviceMuteStrategies(outputDesc, prevDevice, delayMs); - - // Do not change the routing if: - // the requested device is AUDIO_DEVICE_NONE - // OR the requested device is the same as current device - // AND force is not specified - // AND the output is connected by a valid audio patch. - // Doing this check here allows the caller to call setOutputDevice() without conditions - if ((device == AUDIO_DEVICE_NONE || device == prevDevice) && !force && - outputDesc->mPatchHandle != 0) { - ALOGV("setOutputDevice() setting same device %04x or null device for output %d", - device, output); - return muteWaitMs; - } - - ALOGV("setOutputDevice() changing device"); - - // do the routing - if (device == AUDIO_DEVICE_NONE) { - resetOutputDevice(output, delayMs, NULL); - } else { - DeviceVector deviceList = (address == NULL) ? - mAvailableOutputDevices.getDevicesFromType(device) - : mAvailableOutputDevices.getDevicesFromTypeAddr(device, String8(address)); - if (!deviceList.isEmpty()) { - struct audio_patch patch; - outputDesc->toAudioPortConfig(&patch.sources[0]); - patch.num_sources = 1; - patch.num_sinks = 0; - for (size_t i = 0; i < deviceList.size() && i < AUDIO_PATCH_PORTS_MAX; i++) { - deviceList.itemAt(i)->toAudioPortConfig(&patch.sinks[i]); - patch.num_sinks++; - } - ssize_t index; - if (patchHandle && *patchHandle != AUDIO_PATCH_HANDLE_NONE) { - index = mAudioPatches.indexOfKey(*patchHandle); - } else { - index = mAudioPatches.indexOfKey(outputDesc->mPatchHandle); - } - sp< AudioPatch> patchDesc; - audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE; - if (index >= 0) { - patchDesc = mAudioPatches.valueAt(index); - afPatchHandle = patchDesc->mAfPatchHandle; - } - - status_t status = mpClientInterface->createAudioPatch(&patch, - &afPatchHandle, - delayMs); - ALOGV("setOutputDevice() createAudioPatch returned %d patchHandle %d" - "num_sources %d num_sinks %d", - status, afPatchHandle, patch.num_sources, patch.num_sinks); - if (status == NO_ERROR) { - if (index < 0) { - patchDesc = new AudioPatch((audio_patch_handle_t)nextUniqueId(), - &patch, mUidCached); - addAudioPatch(patchDesc->mHandle, patchDesc); - } else { - patchDesc->mPatch = patch; - } - patchDesc->mAfPatchHandle = afPatchHandle; - patchDesc->mUid = mUidCached; - if (patchHandle) { - *patchHandle = patchDesc->mHandle; - } - outputDesc->mPatchHandle = patchDesc->mHandle; - nextAudioPortGeneration(); - mpClientInterface->onAudioPatchListUpdate(); - } - } - - // inform all input as well - for (size_t i = 0; i < mInputs.size(); i++) { - const sp inputDescriptor = mInputs.valueAt(i); - if (!isVirtualInputDevice(inputDescriptor->mDevice)) { - AudioParameter inputCmd = AudioParameter(); - ALOGV("%s: inform input %d of device:%d", __func__, - inputDescriptor->mIoHandle, device); - inputCmd.addInt(String8(AudioParameter::keyRouting),device); - mpClientInterface->setParameters(inputDescriptor->mIoHandle, - inputCmd.toString(), - delayMs); - } - } - } - - // update stream volumes according to new device - applyStreamVolumes(output, device, delayMs); - - return muteWaitMs; -} - -status_t AudioPolicyManager::resetOutputDevice(audio_io_handle_t output, - int delayMs, - audio_patch_handle_t *patchHandle) -{ - sp outputDesc = mOutputs.valueFor(output); - ssize_t index; - if (patchHandle) { - index = mAudioPatches.indexOfKey(*patchHandle); - } else { - index = mAudioPatches.indexOfKey(outputDesc->mPatchHandle); - } - if (index < 0) { - return INVALID_OPERATION; - } - sp< AudioPatch> patchDesc = mAudioPatches.valueAt(index); - status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, delayMs); - ALOGV("resetOutputDevice() releaseAudioPatch returned %d", status); - outputDesc->mPatchHandle = 0; - removeAudioPatch(patchDesc->mHandle); - nextAudioPortGeneration(); - mpClientInterface->onAudioPatchListUpdate(); - return status; -} - -status_t AudioPolicyManager::setInputDevice(audio_io_handle_t input, - audio_devices_t device, - bool force, - audio_patch_handle_t *patchHandle) -{ - status_t status = NO_ERROR; - - sp inputDesc = mInputs.valueFor(input); - if ((device != AUDIO_DEVICE_NONE) && ((device != inputDesc->mDevice) || force)) { - inputDesc->mDevice = device; - - DeviceVector deviceList = mAvailableInputDevices.getDevicesFromType(device); - if (!deviceList.isEmpty()) { - struct audio_patch patch; - inputDesc->toAudioPortConfig(&patch.sinks[0]); - // AUDIO_SOURCE_HOTWORD is for internal use only: - // handled as AUDIO_SOURCE_VOICE_RECOGNITION by the audio HAL - if (patch.sinks[0].ext.mix.usecase.source == AUDIO_SOURCE_HOTWORD && - !inputDesc->mIsSoundTrigger) { - patch.sinks[0].ext.mix.usecase.source = AUDIO_SOURCE_VOICE_RECOGNITION; - } - patch.num_sinks = 1; - //only one input device for now - deviceList.itemAt(0)->toAudioPortConfig(&patch.sources[0]); - patch.num_sources = 1; - ssize_t index; - if (patchHandle && *patchHandle != AUDIO_PATCH_HANDLE_NONE) { - index = mAudioPatches.indexOfKey(*patchHandle); - } else { - index = mAudioPatches.indexOfKey(inputDesc->mPatchHandle); - } - sp< AudioPatch> patchDesc; - audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE; - if (index >= 0) { - patchDesc = mAudioPatches.valueAt(index); - afPatchHandle = patchDesc->mAfPatchHandle; - } - - status_t status = mpClientInterface->createAudioPatch(&patch, - &afPatchHandle, - 0); - ALOGV("setInputDevice() createAudioPatch returned %d patchHandle %d", - status, afPatchHandle); - if (status == NO_ERROR) { - if (index < 0) { - patchDesc = new AudioPatch((audio_patch_handle_t)nextUniqueId(), - &patch, mUidCached); - addAudioPatch(patchDesc->mHandle, patchDesc); - } else { - patchDesc->mPatch = patch; - } - patchDesc->mAfPatchHandle = afPatchHandle; - patchDesc->mUid = mUidCached; - if (patchHandle) { - *patchHandle = patchDesc->mHandle; - } - inputDesc->mPatchHandle = patchDesc->mHandle; - nextAudioPortGeneration(); - mpClientInterface->onAudioPatchListUpdate(); - } - } - } - return status; -} - -status_t AudioPolicyManager::resetInputDevice(audio_io_handle_t input, - audio_patch_handle_t *patchHandle) -{ - sp inputDesc = mInputs.valueFor(input); - ssize_t index; - if (patchHandle) { - index = mAudioPatches.indexOfKey(*patchHandle); - } else { - index = mAudioPatches.indexOfKey(inputDesc->mPatchHandle); - } - if (index < 0) { - return INVALID_OPERATION; - } - sp< AudioPatch> patchDesc = mAudioPatches.valueAt(index); - status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0); - ALOGV("resetInputDevice() releaseAudioPatch returned %d", status); - inputDesc->mPatchHandle = 0; - removeAudioPatch(patchDesc->mHandle); - nextAudioPortGeneration(); - mpClientInterface->onAudioPatchListUpdate(); - return status; -} - -sp AudioPolicyManager::getInputProfile(audio_devices_t device, - String8 address, - uint32_t& samplingRate, - audio_format_t format, - audio_channel_mask_t channelMask, - audio_input_flags_t flags) -{ - // Choose an input profile based on the requested capture parameters: select the first available - // profile supporting all requested parameters. - - for (size_t i = 0; i < mHwModules.size(); i++) - { - if (mHwModules[i]->mHandle == 0) { - continue; - } - for (size_t j = 0; j < mHwModules[i]->mInputProfiles.size(); j++) - { - sp profile = mHwModules[i]->mInputProfiles[j]; - // profile->log(); - if (profile->isCompatibleProfile(device, address, samplingRate, - &samplingRate /*updatedSamplingRate*/, - format, channelMask, (audio_output_flags_t) flags)) { - - return profile; - } - } - } - return NULL; -} - - -audio_devices_t AudioPolicyManager::getDeviceAndMixForInputSource(audio_source_t inputSource, - AudioMix **policyMix) -{ - audio_devices_t availableDeviceTypes = mAvailableInputDevices.types() & - ~AUDIO_DEVICE_BIT_IN; - - for (size_t i = 0; i < mPolicyMixes.size(); i++) { - if (mPolicyMixes[i]->mMix.mMixType != MIX_TYPE_RECORDERS) { - continue; - } - for (size_t j = 0; j < mPolicyMixes[i]->mMix.mCriteria.size(); j++) { - if ((RULE_MATCH_ATTRIBUTE_CAPTURE_PRESET == mPolicyMixes[i]->mMix.mCriteria[j].mRule && - mPolicyMixes[i]->mMix.mCriteria[j].mAttr.mSource == inputSource) || - (RULE_EXCLUDE_ATTRIBUTE_CAPTURE_PRESET == mPolicyMixes[i]->mMix.mCriteria[j].mRule && - mPolicyMixes[i]->mMix.mCriteria[j].mAttr.mSource != inputSource)) { - if (availableDeviceTypes & AUDIO_DEVICE_IN_REMOTE_SUBMIX) { - if (policyMix != NULL) { - *policyMix = &mPolicyMixes[i]->mMix; - } - return AUDIO_DEVICE_IN_REMOTE_SUBMIX; - } - break; - } - } - } - - return getDeviceForInputSource(inputSource); -} - -audio_devices_t AudioPolicyManager::getDeviceForInputSource(audio_source_t inputSource) -{ - uint32_t device = AUDIO_DEVICE_NONE; - audio_devices_t availableDeviceTypes = mAvailableInputDevices.types() & - ~AUDIO_DEVICE_BIT_IN; - - switch (inputSource) { - case AUDIO_SOURCE_VOICE_UPLINK: - if (availableDeviceTypes & AUDIO_DEVICE_IN_VOICE_CALL) { - device = AUDIO_DEVICE_IN_VOICE_CALL; - break; - } - break; - - case AUDIO_SOURCE_DEFAULT: - case AUDIO_SOURCE_MIC: - if (availableDeviceTypes & AUDIO_DEVICE_IN_BLUETOOTH_A2DP) { - device = AUDIO_DEVICE_IN_BLUETOOTH_A2DP; - } else if ((mForceUse[AUDIO_POLICY_FORCE_FOR_RECORD] == AUDIO_POLICY_FORCE_BT_SCO) && - (availableDeviceTypes & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET)) { - device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET; - } else if (availableDeviceTypes & AUDIO_DEVICE_IN_WIRED_HEADSET) { - device = AUDIO_DEVICE_IN_WIRED_HEADSET; - } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_DEVICE) { - device = AUDIO_DEVICE_IN_USB_DEVICE; - } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) { - device = AUDIO_DEVICE_IN_BUILTIN_MIC; - } - break; - - case AUDIO_SOURCE_VOICE_COMMUNICATION: - // Allow only use of devices on primary input if in call and HAL does not support routing - // to voice call path. - if ((mPhoneState == AUDIO_MODE_IN_CALL) && - (mAvailableOutputDevices.types() & AUDIO_DEVICE_OUT_TELEPHONY_TX) == 0) { - availableDeviceTypes = availablePrimaryInputDevices() & ~AUDIO_DEVICE_BIT_IN; - } - - switch (mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION]) { - case AUDIO_POLICY_FORCE_BT_SCO: - // if SCO device is requested but no SCO device is available, fall back to default case - if (availableDeviceTypes & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) { - device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET; - break; - } - // FALL THROUGH - - default: // FORCE_NONE - if (availableDeviceTypes & AUDIO_DEVICE_IN_WIRED_HEADSET) { - device = AUDIO_DEVICE_IN_WIRED_HEADSET; - } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_DEVICE) { - device = AUDIO_DEVICE_IN_USB_DEVICE; - } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) { - device = AUDIO_DEVICE_IN_BUILTIN_MIC; - } - break; - - case AUDIO_POLICY_FORCE_SPEAKER: - if (availableDeviceTypes & AUDIO_DEVICE_IN_BACK_MIC) { - device = AUDIO_DEVICE_IN_BACK_MIC; - } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) { - device = AUDIO_DEVICE_IN_BUILTIN_MIC; - } - break; - } - break; - - case AUDIO_SOURCE_VOICE_RECOGNITION: - case AUDIO_SOURCE_HOTWORD: - if (mForceUse[AUDIO_POLICY_FORCE_FOR_RECORD] == AUDIO_POLICY_FORCE_BT_SCO && - availableDeviceTypes & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) { - device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET; - } else if (availableDeviceTypes & AUDIO_DEVICE_IN_WIRED_HEADSET) { - device = AUDIO_DEVICE_IN_WIRED_HEADSET; - } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_DEVICE) { - device = AUDIO_DEVICE_IN_USB_DEVICE; - } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) { - device = AUDIO_DEVICE_IN_BUILTIN_MIC; - } - break; - case AUDIO_SOURCE_CAMCORDER: - if (availableDeviceTypes & AUDIO_DEVICE_IN_BACK_MIC) { - device = AUDIO_DEVICE_IN_BACK_MIC; - } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) { - device = AUDIO_DEVICE_IN_BUILTIN_MIC; - } - break; - case AUDIO_SOURCE_VOICE_DOWNLINK: - case AUDIO_SOURCE_VOICE_CALL: - if (availableDeviceTypes & AUDIO_DEVICE_IN_VOICE_CALL) { - device = AUDIO_DEVICE_IN_VOICE_CALL; - } - break; - case AUDIO_SOURCE_REMOTE_SUBMIX: - if (availableDeviceTypes & AUDIO_DEVICE_IN_REMOTE_SUBMIX) { - device = AUDIO_DEVICE_IN_REMOTE_SUBMIX; - } - break; - case AUDIO_SOURCE_FM_TUNER: - if (availableDeviceTypes & AUDIO_DEVICE_IN_FM_TUNER) { - device = AUDIO_DEVICE_IN_FM_TUNER; - } - break; - default: - ALOGW("getDeviceForInputSource() invalid input source %d", inputSource); - break; - } - ALOGV("getDeviceForInputSource()input source %d, device %08x", inputSource, device); - return device; -} - -bool AudioPolicyManager::isVirtualInputDevice(audio_devices_t device) -{ - if ((device & AUDIO_DEVICE_BIT_IN) != 0) { - device &= ~AUDIO_DEVICE_BIT_IN; - if ((popcount(device) == 1) && ((device & ~APM_AUDIO_IN_DEVICE_VIRTUAL_ALL) == 0)) - return true; - } - return false; -} - -bool AudioPolicyManager::deviceDistinguishesOnAddress(audio_devices_t device) { - return ((device & APM_AUDIO_DEVICE_MATCH_ADDRESS_ALL & ~AUDIO_DEVICE_BIT_IN) != 0); -} - -audio_io_handle_t AudioPolicyManager::getActiveInput(bool ignoreVirtualInputs) -{ - for (size_t i = 0; i < mInputs.size(); i++) { - const sp input_descriptor = mInputs.valueAt(i); - if ((input_descriptor->mRefCount > 0) - && (!ignoreVirtualInputs || !isVirtualInputDevice(input_descriptor->mDevice))) { - return mInputs.keyAt(i); - } - } - return 0; -} - -uint32_t AudioPolicyManager::activeInputsCount() const -{ - uint32_t count = 0; - for (size_t i = 0; i < mInputs.size(); i++) { - const sp desc = mInputs.valueAt(i); - if (desc->mRefCount > 0) { - count++; - } - } - return count; -} - - -audio_devices_t AudioPolicyManager::getDeviceForVolume(audio_devices_t device) -{ - if (device == AUDIO_DEVICE_NONE) { - // this happens when forcing a route update and no track is active on an output. - // In this case the returned category is not important. - device = AUDIO_DEVICE_OUT_SPEAKER; - } else if (popcount(device) > 1) { - // Multiple device selection is either: - // - speaker + one other device: give priority to speaker in this case. - // - one A2DP device + another device: happens with duplicated output. In this case - // retain the device on the A2DP output as the other must not correspond to an active - // selection if not the speaker. - // - HDMI-CEC system audio mode only output: give priority to available item in order. - if (device & AUDIO_DEVICE_OUT_SPEAKER) { - device = AUDIO_DEVICE_OUT_SPEAKER; - } else if (device & AUDIO_DEVICE_OUT_HDMI_ARC) { - device = AUDIO_DEVICE_OUT_HDMI_ARC; - } else if (device & AUDIO_DEVICE_OUT_AUX_LINE) { - device = AUDIO_DEVICE_OUT_AUX_LINE; - } else if (device & AUDIO_DEVICE_OUT_SPDIF) { - device = AUDIO_DEVICE_OUT_SPDIF; - } else { - device = (audio_devices_t)(device & AUDIO_DEVICE_OUT_ALL_A2DP); - } - } - - /*SPEAKER_SAFE is an alias of SPEAKER for purposes of volume control*/ - if (device == AUDIO_DEVICE_OUT_SPEAKER_SAFE) - device = AUDIO_DEVICE_OUT_SPEAKER; - - ALOGW_IF(popcount(device) != 1, - "getDeviceForVolume() invalid device combination: %08x", - device); - - return device; -} - -AudioPolicyManager::device_category AudioPolicyManager::getDeviceCategory(audio_devices_t device) -{ - switch(getDeviceForVolume(device)) { - case AUDIO_DEVICE_OUT_EARPIECE: - return DEVICE_CATEGORY_EARPIECE; - case AUDIO_DEVICE_OUT_WIRED_HEADSET: - case AUDIO_DEVICE_OUT_WIRED_HEADPHONE: - case AUDIO_DEVICE_OUT_BLUETOOTH_SCO: - case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET: - case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP: - case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES: - return DEVICE_CATEGORY_HEADSET; - case AUDIO_DEVICE_OUT_LINE: - case AUDIO_DEVICE_OUT_AUX_DIGITAL: - /*USB? Remote submix?*/ - return DEVICE_CATEGORY_EXT_MEDIA; - case AUDIO_DEVICE_OUT_SPEAKER: - case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT: - case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER: - case AUDIO_DEVICE_OUT_USB_ACCESSORY: - case AUDIO_DEVICE_OUT_USB_DEVICE: - case AUDIO_DEVICE_OUT_REMOTE_SUBMIX: - default: - return DEVICE_CATEGORY_SPEAKER; - } -} - -/* static */ -float AudioPolicyManager::volIndexToAmpl(audio_devices_t device, const StreamDescriptor& streamDesc, - int indexInUi) -{ - device_category deviceCategory = getDeviceCategory(device); - const VolumeCurvePoint *curve = streamDesc.mVolumeCurve[deviceCategory]; - - // the volume index in the UI is relative to the min and max volume indices for this stream type - int nbSteps = 1 + curve[VOLMAX].mIndex - - curve[VOLMIN].mIndex; - int volIdx = (nbSteps * (indexInUi - streamDesc.mIndexMin)) / - (streamDesc.mIndexMax - streamDesc.mIndexMin); - - // find what part of the curve this index volume belongs to, or if it's out of bounds - int segment = 0; - if (volIdx < curve[VOLMIN].mIndex) { // out of bounds - return 0.0f; - } else if (volIdx < curve[VOLKNEE1].mIndex) { - segment = 0; - } else if (volIdx < curve[VOLKNEE2].mIndex) { - segment = 1; - } else if (volIdx <= curve[VOLMAX].mIndex) { - segment = 2; - } else { // out of bounds - return 1.0f; - } - - // linear interpolation in the attenuation table in dB - float decibels = curve[segment].mDBAttenuation + - ((float)(volIdx - curve[segment].mIndex)) * - ( (curve[segment+1].mDBAttenuation - - curve[segment].mDBAttenuation) / - ((float)(curve[segment+1].mIndex - - curve[segment].mIndex)) ); - - float amplification = exp( decibels * 0.115129f); // exp( dB * ln(10) / 20 ) - - ALOGVV("VOLUME vol index=[%d %d %d], dB=[%.1f %.1f %.1f] ampl=%.5f", - curve[segment].mIndex, volIdx, - curve[segment+1].mIndex, - curve[segment].mDBAttenuation, - decibels, - curve[segment+1].mDBAttenuation, - amplification); - - return amplification; -} - -const AudioPolicyManager::VolumeCurvePoint - AudioPolicyManager::sDefaultVolumeCurve[AudioPolicyManager::VOLCNT] = { - {1, -49.5f}, {33, -33.5f}, {66, -17.0f}, {100, 0.0f} -}; - -const AudioPolicyManager::VolumeCurvePoint - AudioPolicyManager::sDefaultMediaVolumeCurve[AudioPolicyManager::VOLCNT] = { - {1, -58.0f}, {20, -40.0f}, {60, -17.0f}, {100, 0.0f} -}; - -const AudioPolicyManager::VolumeCurvePoint - AudioPolicyManager::sExtMediaSystemVolumeCurve[AudioPolicyManager::VOLCNT] = { - {1, -58.0f}, {20, -40.0f}, {60, -21.0f}, {100, -10.0f} -}; - -const AudioPolicyManager::VolumeCurvePoint - AudioPolicyManager::sSpeakerMediaVolumeCurve[AudioPolicyManager::VOLCNT] = { - {1, -56.0f}, {20, -34.0f}, {60, -11.0f}, {100, 0.0f} -}; - -const AudioPolicyManager::VolumeCurvePoint - AudioPolicyManager::sSpeakerMediaVolumeCurveDrc[AudioPolicyManager::VOLCNT] = { - {1, -55.0f}, {20, -43.0f}, {86, -12.0f}, {100, 0.0f} -}; - -const AudioPolicyManager::VolumeCurvePoint - AudioPolicyManager::sSpeakerSonificationVolumeCurve[AudioPolicyManager::VOLCNT] = { - {1, -29.7f}, {33, -20.1f}, {66, -10.2f}, {100, 0.0f} -}; - -const AudioPolicyManager::VolumeCurvePoint - AudioPolicyManager::sSpeakerSonificationVolumeCurveDrc[AudioPolicyManager::VOLCNT] = { - {1, -35.7f}, {33, -26.1f}, {66, -13.2f}, {100, 0.0f} -}; - -// AUDIO_STREAM_SYSTEM, AUDIO_STREAM_ENFORCED_AUDIBLE and AUDIO_STREAM_DTMF volume tracks -// AUDIO_STREAM_RING on phones and AUDIO_STREAM_MUSIC on tablets. -// AUDIO_STREAM_DTMF tracks AUDIO_STREAM_VOICE_CALL while in call (See AudioService.java). -// The range is constrained between -24dB and -6dB over speaker and -30dB and -18dB over headset. - -const AudioPolicyManager::VolumeCurvePoint - AudioPolicyManager::sDefaultSystemVolumeCurve[AudioPolicyManager::VOLCNT] = { - {1, -24.0f}, {33, -18.0f}, {66, -12.0f}, {100, -6.0f} -}; - -const AudioPolicyManager::VolumeCurvePoint - AudioPolicyManager::sDefaultSystemVolumeCurveDrc[AudioPolicyManager::VOLCNT] = { - {1, -34.0f}, {33, -24.0f}, {66, -15.0f}, {100, -6.0f} -}; - -const AudioPolicyManager::VolumeCurvePoint - AudioPolicyManager::sHeadsetSystemVolumeCurve[AudioPolicyManager::VOLCNT] = { - {1, -30.0f}, {33, -26.0f}, {66, -22.0f}, {100, -18.0f} -}; - -const AudioPolicyManager::VolumeCurvePoint - AudioPolicyManager::sDefaultVoiceVolumeCurve[AudioPolicyManager::VOLCNT] = { - {0, -42.0f}, {33, -28.0f}, {66, -14.0f}, {100, 0.0f} -}; - -const AudioPolicyManager::VolumeCurvePoint - AudioPolicyManager::sSpeakerVoiceVolumeCurve[AudioPolicyManager::VOLCNT] = { - {0, -24.0f}, {33, -16.0f}, {66, -8.0f}, {100, 0.0f} -}; - -const AudioPolicyManager::VolumeCurvePoint - AudioPolicyManager::sLinearVolumeCurve[AudioPolicyManager::VOLCNT] = { - {0, -96.0f}, {33, -68.0f}, {66, -34.0f}, {100, 0.0f} -}; - -const AudioPolicyManager::VolumeCurvePoint - AudioPolicyManager::sSilentVolumeCurve[AudioPolicyManager::VOLCNT] = { - {0, -96.0f}, {1, -96.0f}, {2, -96.0f}, {100, -96.0f} -}; - -const AudioPolicyManager::VolumeCurvePoint - AudioPolicyManager::sFullScaleVolumeCurve[AudioPolicyManager::VOLCNT] = { - {0, 0.0f}, {1, 0.0f}, {2, 0.0f}, {100, 0.0f} -}; - -const AudioPolicyManager::VolumeCurvePoint - *AudioPolicyManager::sVolumeProfiles[AUDIO_STREAM_CNT] - [AudioPolicyManager::DEVICE_CATEGORY_CNT] = { - { // AUDIO_STREAM_VOICE_CALL - sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_HEADSET - sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_SPEAKER - sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_EARPIECE - sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA - }, - { // AUDIO_STREAM_SYSTEM - sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET - sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER - sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_EARPIECE - sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA - }, - { // AUDIO_STREAM_RING - sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET - sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER - sDefaultVolumeCurve, // DEVICE_CATEGORY_EARPIECE - sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA - }, - { // AUDIO_STREAM_MUSIC - sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_HEADSET - sSpeakerMediaVolumeCurve, // DEVICE_CATEGORY_SPEAKER - sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_EARPIECE - sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA - }, - { // AUDIO_STREAM_ALARM - sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET - sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER - sDefaultVolumeCurve, // DEVICE_CATEGORY_EARPIECE - sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA - }, - { // AUDIO_STREAM_NOTIFICATION - sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET - sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER - sDefaultVolumeCurve, // DEVICE_CATEGORY_EARPIECE - sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA - }, - { // AUDIO_STREAM_BLUETOOTH_SCO - sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_HEADSET - sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_SPEAKER - sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_EARPIECE - sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA - }, - { // AUDIO_STREAM_ENFORCED_AUDIBLE - sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET - sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER - sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_EARPIECE - sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA - }, - { // AUDIO_STREAM_DTMF - sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET - sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER - sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_EARPIECE - sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA - }, - { // AUDIO_STREAM_TTS - // "Transmitted Through Speaker": always silent except on DEVICE_CATEGORY_SPEAKER - sSilentVolumeCurve, // DEVICE_CATEGORY_HEADSET - sLinearVolumeCurve, // DEVICE_CATEGORY_SPEAKER - sSilentVolumeCurve, // DEVICE_CATEGORY_EARPIECE - sSilentVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA - }, - { // AUDIO_STREAM_ACCESSIBILITY - sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_HEADSET - sSpeakerMediaVolumeCurve, // DEVICE_CATEGORY_SPEAKER - sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_EARPIECE - sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA - }, - { // AUDIO_STREAM_REROUTING - sFullScaleVolumeCurve, // DEVICE_CATEGORY_HEADSET - sFullScaleVolumeCurve, // DEVICE_CATEGORY_SPEAKER - sFullScaleVolumeCurve, // DEVICE_CATEGORY_EARPIECE - sFullScaleVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA - }, - { // AUDIO_STREAM_PATCH - sFullScaleVolumeCurve, // DEVICE_CATEGORY_HEADSET - sFullScaleVolumeCurve, // DEVICE_CATEGORY_SPEAKER - sFullScaleVolumeCurve, // DEVICE_CATEGORY_EARPIECE - sFullScaleVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA - }, -}; - -void AudioPolicyManager::initializeVolumeCurves() -{ - for (int i = 0; i < AUDIO_STREAM_CNT; i++) { - for (int j = 0; j < DEVICE_CATEGORY_CNT; j++) { - mStreams[i].mVolumeCurve[j] = - sVolumeProfiles[i][j]; - } - } - - // Check availability of DRC on speaker path: if available, override some of the speaker curves - if (mSpeakerDrcEnabled) { - mStreams[AUDIO_STREAM_SYSTEM].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] = - sDefaultSystemVolumeCurveDrc; - mStreams[AUDIO_STREAM_RING].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] = - sSpeakerSonificationVolumeCurveDrc; - mStreams[AUDIO_STREAM_ALARM].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] = - sSpeakerSonificationVolumeCurveDrc; - mStreams[AUDIO_STREAM_NOTIFICATION].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] = - sSpeakerSonificationVolumeCurveDrc; - mStreams[AUDIO_STREAM_MUSIC].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] = - sSpeakerMediaVolumeCurveDrc; - mStreams[AUDIO_STREAM_ACCESSIBILITY].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] = - sSpeakerMediaVolumeCurveDrc; - } -} - -float AudioPolicyManager::computeVolume(audio_stream_type_t stream, - int index, - audio_io_handle_t output, - audio_devices_t device) -{ - float volume = 1.0; - sp outputDesc = mOutputs.valueFor(output); - StreamDescriptor &streamDesc = mStreams[stream]; - - if (device == AUDIO_DEVICE_NONE) { - device = outputDesc->device(); - } - - volume = volIndexToAmpl(device, streamDesc, index); - - // if a headset is connected, apply the following rules to ring tones and notifications - // to avoid sound level bursts in user's ears: - // - always attenuate ring tones and notifications volume by 6dB - // - if music is playing, always limit the volume to current music volume, - // with a minimum threshold at -36dB so that notification is always perceived. - const routing_strategy stream_strategy = getStrategy(stream); - if ((device & (AUDIO_DEVICE_OUT_BLUETOOTH_A2DP | - AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES | - AUDIO_DEVICE_OUT_WIRED_HEADSET | - AUDIO_DEVICE_OUT_WIRED_HEADPHONE)) && - ((stream_strategy == STRATEGY_SONIFICATION) - || (stream_strategy == STRATEGY_SONIFICATION_RESPECTFUL) - || (stream == AUDIO_STREAM_SYSTEM) - || ((stream_strategy == STRATEGY_ENFORCED_AUDIBLE) && - (mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] == AUDIO_POLICY_FORCE_NONE))) && - streamDesc.mCanBeMuted) { - volume *= SONIFICATION_HEADSET_VOLUME_FACTOR; - // when the phone is ringing we must consider that music could have been paused just before - // by the music application and behave as if music was active if the last music track was - // just stopped - if (isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY) || - mLimitRingtoneVolume) { - audio_devices_t musicDevice = getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/); - float musicVol = computeVolume(AUDIO_STREAM_MUSIC, - mStreams[AUDIO_STREAM_MUSIC].getVolumeIndex(musicDevice), - output, - musicDevice); - float minVol = (musicVol > SONIFICATION_HEADSET_VOLUME_MIN) ? - musicVol : SONIFICATION_HEADSET_VOLUME_MIN; - if (volume > minVol) { - volume = minVol; - ALOGV("computeVolume limiting volume to %f musicVol %f", minVol, musicVol); - } - } - } - - return volume; -} - -status_t AudioPolicyManager::checkAndSetVolume(audio_stream_type_t stream, - int index, - audio_io_handle_t output, - audio_devices_t device, - int delayMs, - bool force) -{ - - // do not change actual stream volume if the stream is muted - if (mOutputs.valueFor(output)->mMuteCount[stream] != 0) { - ALOGVV("checkAndSetVolume() stream %d muted count %d", - stream, mOutputs.valueFor(output)->mMuteCount[stream]); - return NO_ERROR; - } - - // do not change in call volume if bluetooth is connected and vice versa - if ((stream == AUDIO_STREAM_VOICE_CALL && - mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION] == AUDIO_POLICY_FORCE_BT_SCO) || - (stream == AUDIO_STREAM_BLUETOOTH_SCO && - mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION] != AUDIO_POLICY_FORCE_BT_SCO)) { - ALOGV("checkAndSetVolume() cannot set stream %d volume with force use = %d for comm", - stream, mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION]); - return INVALID_OPERATION; - } - - float volume = computeVolume(stream, index, output, device); - // unit gain if rerouting to external policy - if (device == AUDIO_DEVICE_OUT_REMOTE_SUBMIX) { - ssize_t index = mOutputs.indexOfKey(output); - if (index >= 0) { - sp outputDesc = mOutputs.valueAt(index); - if (outputDesc->mPolicyMix != NULL) { - ALOGV("max gain when rerouting for output=%d", output); - volume = 1.0f; - } - } - - } - // We actually change the volume if: - // - the float value returned by computeVolume() changed - // - the force flag is set - if (volume != mOutputs.valueFor(output)->mCurVolume[stream] || - force) { - mOutputs.valueFor(output)->mCurVolume[stream] = volume; - ALOGVV("checkAndSetVolume() for output %d stream %d, volume %f, delay %d", output, stream, volume, delayMs); - // Force VOICE_CALL to track BLUETOOTH_SCO stream volume when bluetooth audio is - // enabled - if (stream == AUDIO_STREAM_BLUETOOTH_SCO) { - mpClientInterface->setStreamVolume(AUDIO_STREAM_VOICE_CALL, volume, output, delayMs); - } - mpClientInterface->setStreamVolume(stream, volume, output, delayMs); - } - - if (stream == AUDIO_STREAM_VOICE_CALL || - stream == AUDIO_STREAM_BLUETOOTH_SCO) { - float voiceVolume; - // Force voice volume to max for bluetooth SCO as volume is managed by the headset - if (stream == AUDIO_STREAM_VOICE_CALL) { - voiceVolume = (float)index/(float)mStreams[stream].mIndexMax; - } else { - voiceVolume = 1.0; - } - - if (voiceVolume != mLastVoiceVolume && output == mPrimaryOutput) { - mpClientInterface->setVoiceVolume(voiceVolume, delayMs); - mLastVoiceVolume = voiceVolume; - } - } - - return NO_ERROR; -} - -void AudioPolicyManager::applyStreamVolumes(audio_io_handle_t output, - audio_devices_t device, - int delayMs, - bool force) -{ - ALOGVV("applyStreamVolumes() for output %d and device %x", output, device); - - for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) { - if (stream == AUDIO_STREAM_PATCH) { - continue; - } - checkAndSetVolume((audio_stream_type_t)stream, - mStreams[stream].getVolumeIndex(device), - output, - device, - delayMs, - force); - } -} - -void AudioPolicyManager::setStrategyMute(routing_strategy strategy, - bool on, - audio_io_handle_t output, - int delayMs, - audio_devices_t device) -{ - ALOGVV("setStrategyMute() strategy %d, mute %d, output %d", strategy, on, output); - for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) { - if (stream == AUDIO_STREAM_PATCH) { - continue; - } - if (getStrategy((audio_stream_type_t)stream) == strategy) { - setStreamMute((audio_stream_type_t)stream, on, output, delayMs, device); - } - } -} - -void AudioPolicyManager::setStreamMute(audio_stream_type_t stream, - bool on, - audio_io_handle_t output, - int delayMs, - audio_devices_t device) -{ - StreamDescriptor &streamDesc = mStreams[stream]; - sp outputDesc = mOutputs.valueFor(output); - if (device == AUDIO_DEVICE_NONE) { - device = outputDesc->device(); - } - - ALOGVV("setStreamMute() stream %d, mute %d, output %d, mMuteCount %d device %04x", - stream, on, output, outputDesc->mMuteCount[stream], device); - - if (on) { - if (outputDesc->mMuteCount[stream] == 0) { - if (streamDesc.mCanBeMuted && - ((stream != AUDIO_STREAM_ENFORCED_AUDIBLE) || - (mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] == AUDIO_POLICY_FORCE_NONE))) { - checkAndSetVolume(stream, 0, output, device, delayMs); - } - } - // increment mMuteCount after calling checkAndSetVolume() so that volume change is not ignored - outputDesc->mMuteCount[stream]++; - } else { - if (outputDesc->mMuteCount[stream] == 0) { - ALOGV("setStreamMute() unmuting non muted stream!"); - return; - } - if (--outputDesc->mMuteCount[stream] == 0) { - checkAndSetVolume(stream, - streamDesc.getVolumeIndex(device), - output, - device, - delayMs); - } - } -} - -void AudioPolicyManager::handleIncallSonification(audio_stream_type_t stream, - bool starting, bool stateChange) -{ - // if the stream pertains to sonification strategy and we are in call we must - // mute the stream if it is low visibility. If it is high visibility, we must play a tone - // in the device used for phone strategy and play the tone if the selected device does not - // interfere with the device used for phone strategy - // if stateChange is true, we are called from setPhoneState() and we must mute or unmute as - // many times as there are active tracks on the output - const routing_strategy stream_strategy = getStrategy(stream); - if ((stream_strategy == STRATEGY_SONIFICATION) || - ((stream_strategy == STRATEGY_SONIFICATION_RESPECTFUL))) { - sp outputDesc = mOutputs.valueFor(mPrimaryOutput); - ALOGV("handleIncallSonification() stream %d starting %d device %x stateChange %d", - stream, starting, outputDesc->mDevice, stateChange); - if (outputDesc->mRefCount[stream]) { - int muteCount = 1; - if (stateChange) { - muteCount = outputDesc->mRefCount[stream]; - } - if (audio_is_low_visibility(stream)) { - ALOGV("handleIncallSonification() low visibility, muteCount %d", muteCount); - for (int i = 0; i < muteCount; i++) { - setStreamMute(stream, starting, mPrimaryOutput); - } - } else { - ALOGV("handleIncallSonification() high visibility"); - if (outputDesc->device() & - getDeviceForStrategy(STRATEGY_PHONE, true /*fromCache*/)) { - ALOGV("handleIncallSonification() high visibility muted, muteCount %d", muteCount); - for (int i = 0; i < muteCount; i++) { - setStreamMute(stream, starting, mPrimaryOutput); - } - } - if (starting) { - mpClientInterface->startTone(AUDIO_POLICY_TONE_IN_CALL_NOTIFICATION, - AUDIO_STREAM_VOICE_CALL); - } else { - mpClientInterface->stopTone(); - } - } - } - } -} - -bool AudioPolicyManager::isInCall() -{ - return isStateInCall(mPhoneState); -} - -bool AudioPolicyManager::isStateInCall(int state) { - return ((state == AUDIO_MODE_IN_CALL) || - (state == AUDIO_MODE_IN_COMMUNICATION)); -} - -uint32_t AudioPolicyManager::getMaxEffectsCpuLoad() -{ - return MAX_EFFECTS_CPU_LOAD; -} - -uint32_t AudioPolicyManager::getMaxEffectsMemory() -{ - return MAX_EFFECTS_MEMORY; -} - - -// --- AudioOutputDescriptor class implementation - -AudioPolicyManager::AudioOutputDescriptor::AudioOutputDescriptor( - const sp& profile) - : mId(0), mIoHandle(0), mLatency(0), - mFlags((audio_output_flags_t)0), mDevice(AUDIO_DEVICE_NONE), mPolicyMix(NULL), - mPatchHandle(0), - mOutput1(0), mOutput2(0), mProfile(profile), mDirectOpenCount(0) -{ - // clear usage count for all stream types - for (int i = 0; i < AUDIO_STREAM_CNT; i++) { - mRefCount[i] = 0; - mCurVolume[i] = -1.0; - mMuteCount[i] = 0; - mStopTime[i] = 0; - } - for (int i = 0; i < NUM_STRATEGIES; i++) { - mStrategyMutedByDevice[i] = false; - } - if (profile != NULL) { - mFlags = (audio_output_flags_t)profile->mFlags; - mSamplingRate = profile->pickSamplingRate(); - mFormat = profile->pickFormat(); - mChannelMask = profile->pickChannelMask(); - if (profile->mGains.size() > 0) { - profile->mGains[0]->getDefaultConfig(&mGain); - } - } -} - -audio_devices_t AudioPolicyManager::AudioOutputDescriptor::device() const -{ - if (isDuplicated()) { - return (audio_devices_t)(mOutput1->mDevice | mOutput2->mDevice); - } else { - return mDevice; - } -} - -uint32_t AudioPolicyManager::AudioOutputDescriptor::latency() -{ - if (isDuplicated()) { - return (mOutput1->mLatency > mOutput2->mLatency) ? mOutput1->mLatency : mOutput2->mLatency; - } else { - return mLatency; - } -} - -bool AudioPolicyManager::AudioOutputDescriptor::sharesHwModuleWith( - const sp outputDesc) -{ - if (isDuplicated()) { - return mOutput1->sharesHwModuleWith(outputDesc) || mOutput2->sharesHwModuleWith(outputDesc); - } else if (outputDesc->isDuplicated()){ - return sharesHwModuleWith(outputDesc->mOutput1) || sharesHwModuleWith(outputDesc->mOutput2); - } else { - return (mProfile->mModule == outputDesc->mProfile->mModule); - } -} - -void AudioPolicyManager::AudioOutputDescriptor::changeRefCount(audio_stream_type_t stream, - int delta) -{ - // forward usage count change to attached outputs - if (isDuplicated()) { - mOutput1->changeRefCount(stream, delta); - mOutput2->changeRefCount(stream, delta); - } - if ((delta + (int)mRefCount[stream]) < 0) { - ALOGW("changeRefCount() invalid delta %d for stream %d, refCount %d", - delta, stream, mRefCount[stream]); - mRefCount[stream] = 0; - return; - } - mRefCount[stream] += delta; - ALOGV("changeRefCount() stream %d, count %d", stream, mRefCount[stream]); -} - -audio_devices_t AudioPolicyManager::AudioOutputDescriptor::supportedDevices() -{ - if (isDuplicated()) { - return (audio_devices_t)(mOutput1->supportedDevices() | mOutput2->supportedDevices()); - } else { - return mProfile->mSupportedDevices.types() ; - } -} - -bool AudioPolicyManager::AudioOutputDescriptor::isActive(uint32_t inPastMs) const -{ - return isStrategyActive(NUM_STRATEGIES, inPastMs); -} - -bool AudioPolicyManager::AudioOutputDescriptor::isStrategyActive(routing_strategy strategy, - uint32_t inPastMs, - nsecs_t sysTime) const -{ - if ((sysTime == 0) && (inPastMs != 0)) { - sysTime = systemTime(); - } - for (int i = 0; i < (int)AUDIO_STREAM_CNT; i++) { - if (i == AUDIO_STREAM_PATCH) { - continue; - } - if (((getStrategy((audio_stream_type_t)i) == strategy) || - (NUM_STRATEGIES == strategy)) && - isStreamActive((audio_stream_type_t)i, inPastMs, sysTime)) { - return true; - } - } - return false; -} - -bool AudioPolicyManager::AudioOutputDescriptor::isStreamActive(audio_stream_type_t stream, - uint32_t inPastMs, - nsecs_t sysTime) const -{ - if (mRefCount[stream] != 0) { - return true; - } - if (inPastMs == 0) { - return false; - } - if (sysTime == 0) { - sysTime = systemTime(); - } - if (ns2ms(sysTime - mStopTime[stream]) < inPastMs) { - return true; - } - return false; -} - -void AudioPolicyManager::AudioOutputDescriptor::toAudioPortConfig( - struct audio_port_config *dstConfig, - const struct audio_port_config *srcConfig) const -{ - ALOG_ASSERT(!isDuplicated(), "toAudioPortConfig() called on duplicated output %d", mIoHandle); - - dstConfig->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK| - AUDIO_PORT_CONFIG_FORMAT|AUDIO_PORT_CONFIG_GAIN; - if (srcConfig != NULL) { - dstConfig->config_mask |= srcConfig->config_mask; - } - AudioPortConfig::toAudioPortConfig(dstConfig, srcConfig); - - dstConfig->id = mId; - dstConfig->role = AUDIO_PORT_ROLE_SOURCE; - dstConfig->type = AUDIO_PORT_TYPE_MIX; - dstConfig->ext.mix.hw_module = mProfile->mModule->mHandle; - dstConfig->ext.mix.handle = mIoHandle; - dstConfig->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; -} - -void AudioPolicyManager::AudioOutputDescriptor::toAudioPort( - struct audio_port *port) const -{ - ALOG_ASSERT(!isDuplicated(), "toAudioPort() called on duplicated output %d", mIoHandle); - mProfile->toAudioPort(port); - port->id = mId; - toAudioPortConfig(&port->active_config); - port->ext.mix.hw_module = mProfile->mModule->mHandle; - port->ext.mix.handle = mIoHandle; - port->ext.mix.latency_class = - mFlags & AUDIO_OUTPUT_FLAG_FAST ? AUDIO_LATENCY_LOW : AUDIO_LATENCY_NORMAL; -} - -status_t AudioPolicyManager::AudioOutputDescriptor::dump(int fd) -{ - const size_t SIZE = 256; - char buffer[SIZE]; - String8 result; - - snprintf(buffer, SIZE, " ID: %d\n", mId); - result.append(buffer); - snprintf(buffer, SIZE, " Sampling rate: %d\n", mSamplingRate); - result.append(buffer); - snprintf(buffer, SIZE, " Format: %08x\n", mFormat); - result.append(buffer); - snprintf(buffer, SIZE, " Channels: %08x\n", mChannelMask); - result.append(buffer); - snprintf(buffer, SIZE, " Latency: %d\n", mLatency); - result.append(buffer); - snprintf(buffer, SIZE, " Flags %08x\n", mFlags); - result.append(buffer); - snprintf(buffer, SIZE, " Devices %08x\n", device()); - result.append(buffer); - snprintf(buffer, SIZE, " Stream volume refCount muteCount\n"); - result.append(buffer); - for (int i = 0; i < (int)AUDIO_STREAM_CNT; i++) { - snprintf(buffer, SIZE, " %02d %.03f %02d %02d\n", - i, mCurVolume[i], mRefCount[i], mMuteCount[i]); - result.append(buffer); - } - write(fd, result.string(), result.size()); - - return NO_ERROR; -} - -// --- AudioInputDescriptor class implementation - -AudioPolicyManager::AudioInputDescriptor::AudioInputDescriptor(const sp& profile) - : mId(0), mIoHandle(0), - mDevice(AUDIO_DEVICE_NONE), mPolicyMix(NULL), mPatchHandle(0), mRefCount(0), - mInputSource(AUDIO_SOURCE_DEFAULT), mProfile(profile), mIsSoundTrigger(false) -{ - if (profile != NULL) { - mSamplingRate = profile->pickSamplingRate(); - mFormat = profile->pickFormat(); - mChannelMask = profile->pickChannelMask(); - if (profile->mGains.size() > 0) { - profile->mGains[0]->getDefaultConfig(&mGain); - } - } -} - -void AudioPolicyManager::AudioInputDescriptor::toAudioPortConfig( - struct audio_port_config *dstConfig, - const struct audio_port_config *srcConfig) const -{ - ALOG_ASSERT(mProfile != 0, - "toAudioPortConfig() called on input with null profile %d", mIoHandle); - dstConfig->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK| - AUDIO_PORT_CONFIG_FORMAT|AUDIO_PORT_CONFIG_GAIN; - if (srcConfig != NULL) { - dstConfig->config_mask |= srcConfig->config_mask; - } - - AudioPortConfig::toAudioPortConfig(dstConfig, srcConfig); - - dstConfig->id = mId; - dstConfig->role = AUDIO_PORT_ROLE_SINK; - dstConfig->type = AUDIO_PORT_TYPE_MIX; - dstConfig->ext.mix.hw_module = mProfile->mModule->mHandle; - dstConfig->ext.mix.handle = mIoHandle; - dstConfig->ext.mix.usecase.source = mInputSource; -} - -void AudioPolicyManager::AudioInputDescriptor::toAudioPort( - struct audio_port *port) const -{ - ALOG_ASSERT(mProfile != 0, "toAudioPort() called on input with null profile %d", mIoHandle); - - mProfile->toAudioPort(port); - port->id = mId; - toAudioPortConfig(&port->active_config); - port->ext.mix.hw_module = mProfile->mModule->mHandle; - port->ext.mix.handle = mIoHandle; - port->ext.mix.latency_class = AUDIO_LATENCY_NORMAL; -} - -status_t AudioPolicyManager::AudioInputDescriptor::dump(int fd) -{ - const size_t SIZE = 256; - char buffer[SIZE]; - String8 result; - - snprintf(buffer, SIZE, " ID: %d\n", mId); - result.append(buffer); - snprintf(buffer, SIZE, " Sampling rate: %d\n", mSamplingRate); - result.append(buffer); - snprintf(buffer, SIZE, " Format: %d\n", mFormat); - result.append(buffer); - snprintf(buffer, SIZE, " Channels: %08x\n", mChannelMask); - result.append(buffer); - snprintf(buffer, SIZE, " Devices %08x\n", mDevice); - result.append(buffer); - snprintf(buffer, SIZE, " Ref Count %d\n", mRefCount); - result.append(buffer); - snprintf(buffer, SIZE, " Open Ref Count %d\n", mOpenRefCount); - result.append(buffer); - - write(fd, result.string(), result.size()); - - return NO_ERROR; -} - -// --- StreamDescriptor class implementation - -AudioPolicyManager::StreamDescriptor::StreamDescriptor() - : mIndexMin(0), mIndexMax(1), mCanBeMuted(true) -{ - mIndexCur.add(AUDIO_DEVICE_OUT_DEFAULT, 0); -} - -int AudioPolicyManager::StreamDescriptor::getVolumeIndex(audio_devices_t device) -{ - device = AudioPolicyManager::getDeviceForVolume(device); - // there is always a valid entry for AUDIO_DEVICE_OUT_DEFAULT - if (mIndexCur.indexOfKey(device) < 0) { - device = AUDIO_DEVICE_OUT_DEFAULT; - } - return mIndexCur.valueFor(device); -} - -void AudioPolicyManager::StreamDescriptor::dump(int fd) -{ - const size_t SIZE = 256; - char buffer[SIZE]; - String8 result; - - snprintf(buffer, SIZE, "%s %02d %02d ", - mCanBeMuted ? "true " : "false", mIndexMin, mIndexMax); - result.append(buffer); - for (size_t i = 0; i < mIndexCur.size(); i++) { - snprintf(buffer, SIZE, "%04x : %02d, ", - mIndexCur.keyAt(i), - mIndexCur.valueAt(i)); - result.append(buffer); - } - result.append("\n"); - - write(fd, result.string(), result.size()); -} - -// --- EffectDescriptor class implementation - -status_t AudioPolicyManager::EffectDescriptor::dump(int fd) -{ - const size_t SIZE = 256; - char buffer[SIZE]; - String8 result; - - snprintf(buffer, SIZE, " I/O: %d\n", mIo); - result.append(buffer); - snprintf(buffer, SIZE, " Strategy: %d\n", mStrategy); - result.append(buffer); - snprintf(buffer, SIZE, " Session: %d\n", mSession); - result.append(buffer); - snprintf(buffer, SIZE, " Name: %s\n", mDesc.name); - result.append(buffer); - snprintf(buffer, SIZE, " %s\n", mEnabled ? "Enabled" : "Disabled"); - result.append(buffer); - write(fd, result.string(), result.size()); - - return NO_ERROR; -} - -// --- HwModule class implementation - -AudioPolicyManager::HwModule::HwModule(const char *name) - : mName(strndup(name, AUDIO_HARDWARE_MODULE_ID_MAX_LEN)), - mHalVersion(AUDIO_DEVICE_API_VERSION_MIN), mHandle(0) -{ -} - -AudioPolicyManager::HwModule::~HwModule() -{ - for (size_t i = 0; i < mOutputProfiles.size(); i++) { - mOutputProfiles[i]->mSupportedDevices.clear(); - } - for (size_t i = 0; i < mInputProfiles.size(); i++) { - mInputProfiles[i]->mSupportedDevices.clear(); - } - free((void *)mName); -} - -status_t AudioPolicyManager::HwModule::loadInput(cnode *root) -{ - cnode *node = root->first_child; - - sp profile = new IOProfile(String8(root->name), AUDIO_PORT_ROLE_SINK, this); - - while (node) { - if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) { - profile->loadSamplingRates((char *)node->value); - } else if (strcmp(node->name, FORMATS_TAG) == 0) { - profile->loadFormats((char *)node->value); - } else if (strcmp(node->name, CHANNELS_TAG) == 0) { - profile->loadInChannels((char *)node->value); - } else if (strcmp(node->name, DEVICES_TAG) == 0) { - profile->mSupportedDevices.loadDevicesFromName((char *)node->value, - mDeclaredDevices); - } else if (strcmp(node->name, FLAGS_TAG) == 0) { - profile->mFlags = parseInputFlagNames((char *)node->value); - } else if (strcmp(node->name, GAINS_TAG) == 0) { - profile->loadGains(node); - } - node = node->next; - } - ALOGW_IF(profile->mSupportedDevices.isEmpty(), - "loadInput() invalid supported devices"); - ALOGW_IF(profile->mChannelMasks.size() == 0, - "loadInput() invalid supported channel masks"); - ALOGW_IF(profile->mSamplingRates.size() == 0, - "loadInput() invalid supported sampling rates"); - ALOGW_IF(profile->mFormats.size() == 0, - "loadInput() invalid supported formats"); - if (!profile->mSupportedDevices.isEmpty() && - (profile->mChannelMasks.size() != 0) && - (profile->mSamplingRates.size() != 0) && - (profile->mFormats.size() != 0)) { - - ALOGV("loadInput() adding input Supported Devices %04x", - profile->mSupportedDevices.types()); - - mInputProfiles.add(profile); - return NO_ERROR; - } else { - return BAD_VALUE; - } -} - -status_t AudioPolicyManager::HwModule::loadOutput(cnode *root) -{ - cnode *node = root->first_child; - - sp profile = new IOProfile(String8(root->name), AUDIO_PORT_ROLE_SOURCE, this); - - while (node) { - if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) { - profile->loadSamplingRates((char *)node->value); - } else if (strcmp(node->name, FORMATS_TAG) == 0) { - profile->loadFormats((char *)node->value); - } else if (strcmp(node->name, CHANNELS_TAG) == 0) { - profile->loadOutChannels((char *)node->value); - } else if (strcmp(node->name, DEVICES_TAG) == 0) { - profile->mSupportedDevices.loadDevicesFromName((char *)node->value, - mDeclaredDevices); - } else if (strcmp(node->name, FLAGS_TAG) == 0) { - profile->mFlags = parseOutputFlagNames((char *)node->value); - } else if (strcmp(node->name, GAINS_TAG) == 0) { - profile->loadGains(node); - } - node = node->next; - } - ALOGW_IF(profile->mSupportedDevices.isEmpty(), - "loadOutput() invalid supported devices"); - ALOGW_IF(profile->mChannelMasks.size() == 0, - "loadOutput() invalid supported channel masks"); - ALOGW_IF(profile->mSamplingRates.size() == 0, - "loadOutput() invalid supported sampling rates"); - ALOGW_IF(profile->mFormats.size() == 0, - "loadOutput() invalid supported formats"); - if (!profile->mSupportedDevices.isEmpty() && - (profile->mChannelMasks.size() != 0) && - (profile->mSamplingRates.size() != 0) && - (profile->mFormats.size() != 0)) { - - ALOGV("loadOutput() adding output Supported Devices %04x, mFlags %04x", - profile->mSupportedDevices.types(), profile->mFlags); - - mOutputProfiles.add(profile); - return NO_ERROR; - } else { - return BAD_VALUE; - } -} - -status_t AudioPolicyManager::HwModule::loadDevice(cnode *root) -{ - cnode *node = root->first_child; - - audio_devices_t type = AUDIO_DEVICE_NONE; - while (node) { - if (strcmp(node->name, DEVICE_TYPE) == 0) { - type = parseDeviceNames((char *)node->value); - break; - } - node = node->next; - } - if (type == AUDIO_DEVICE_NONE || - (!audio_is_input_device(type) && !audio_is_output_device(type))) { - ALOGW("loadDevice() bad type %08x", type); - return BAD_VALUE; - } - sp deviceDesc = new DeviceDescriptor(String8(root->name), type); - deviceDesc->mModule = this; - - node = root->first_child; - while (node) { - if (strcmp(node->name, DEVICE_ADDRESS) == 0) { - deviceDesc->mAddress = String8((char *)node->value); - } else if (strcmp(node->name, CHANNELS_TAG) == 0) { - if (audio_is_input_device(type)) { - deviceDesc->loadInChannels((char *)node->value); - } else { - deviceDesc->loadOutChannels((char *)node->value); - } - } else if (strcmp(node->name, GAINS_TAG) == 0) { - deviceDesc->loadGains(node); - } - node = node->next; - } - - ALOGV("loadDevice() adding device name %s type %08x address %s", - deviceDesc->mName.string(), type, deviceDesc->mAddress.string()); - - mDeclaredDevices.add(deviceDesc); - - return NO_ERROR; -} - -status_t AudioPolicyManager::HwModule::addOutputProfile(String8 name, const audio_config_t *config, - audio_devices_t device, String8 address) -{ - sp profile = new IOProfile(name, AUDIO_PORT_ROLE_SOURCE, this); - - profile->mSamplingRates.add(config->sample_rate); - profile->mChannelMasks.add(config->channel_mask); - profile->mFormats.add(config->format); - - sp devDesc = new DeviceDescriptor(name, device); - devDesc->mAddress = address; - profile->mSupportedDevices.add(devDesc); - - mOutputProfiles.add(profile); - - return NO_ERROR; -} - -status_t AudioPolicyManager::HwModule::removeOutputProfile(String8 name) -{ - for (size_t i = 0; i < mOutputProfiles.size(); i++) { - if (mOutputProfiles[i]->mName == name) { - mOutputProfiles.removeAt(i); - break; - } - } - - return NO_ERROR; -} - -status_t AudioPolicyManager::HwModule::addInputProfile(String8 name, const audio_config_t *config, - audio_devices_t device, String8 address) -{ - sp profile = new IOProfile(name, AUDIO_PORT_ROLE_SINK, this); - - profile->mSamplingRates.add(config->sample_rate); - profile->mChannelMasks.add(config->channel_mask); - profile->mFormats.add(config->format); - - sp devDesc = new DeviceDescriptor(name, device); - devDesc->mAddress = address; - profile->mSupportedDevices.add(devDesc); - - ALOGV("addInputProfile() name %s rate %d mask 0x08", name.string(), config->sample_rate, config->channel_mask); - - mInputProfiles.add(profile); - - return NO_ERROR; -} - -status_t AudioPolicyManager::HwModule::removeInputProfile(String8 name) -{ - for (size_t i = 0; i < mInputProfiles.size(); i++) { - if (mInputProfiles[i]->mName == name) { - mInputProfiles.removeAt(i); - break; - } - } - - return NO_ERROR; -} - - -void AudioPolicyManager::HwModule::dump(int fd) -{ - const size_t SIZE = 256; - char buffer[SIZE]; - String8 result; - - snprintf(buffer, SIZE, " - name: %s\n", mName); - result.append(buffer); - snprintf(buffer, SIZE, " - handle: %d\n", mHandle); - result.append(buffer); - snprintf(buffer, SIZE, " - version: %u.%u\n", mHalVersion >> 8, mHalVersion & 0xFF); - result.append(buffer); - write(fd, result.string(), result.size()); - if (mOutputProfiles.size()) { - write(fd, " - outputs:\n", strlen(" - outputs:\n")); - for (size_t i = 0; i < mOutputProfiles.size(); i++) { - snprintf(buffer, SIZE, " output %zu:\n", i); - write(fd, buffer, strlen(buffer)); - mOutputProfiles[i]->dump(fd); - } - } - if (mInputProfiles.size()) { - write(fd, " - inputs:\n", strlen(" - inputs:\n")); - for (size_t i = 0; i < mInputProfiles.size(); i++) { - snprintf(buffer, SIZE, " input %zu:\n", i); - write(fd, buffer, strlen(buffer)); - mInputProfiles[i]->dump(fd); - } - } - if (mDeclaredDevices.size()) { - write(fd, " - devices:\n", strlen(" - devices:\n")); - for (size_t i = 0; i < mDeclaredDevices.size(); i++) { - mDeclaredDevices[i]->dump(fd, 4, i); - } - } -} - -// --- AudioPort class implementation - - -AudioPolicyManager::AudioPort::AudioPort(const String8& name, audio_port_type_t type, - audio_port_role_t role, const sp& module) : - mName(name), mType(type), mRole(role), mModule(module), mFlags(0), mId(0) -{ - mUseInChannelMask = ((type == AUDIO_PORT_TYPE_DEVICE) && (role == AUDIO_PORT_ROLE_SOURCE)) || - ((type == AUDIO_PORT_TYPE_MIX) && (role == AUDIO_PORT_ROLE_SINK)); -} - -void AudioPolicyManager::AudioPort::attach(const sp& module) { - mId = AudioPolicyManager::nextUniqueId(); - mModule = module; -} - -void AudioPolicyManager::AudioPort::toAudioPort(struct audio_port *port) const -{ - port->role = mRole; - port->type = mType; - strlcpy(port->name, mName, AUDIO_PORT_MAX_NAME_LEN); - unsigned int i; - for (i = 0; i < mSamplingRates.size() && i < AUDIO_PORT_MAX_SAMPLING_RATES; i++) { - if (mSamplingRates[i] != 0) { - port->sample_rates[i] = mSamplingRates[i]; - } - } - port->num_sample_rates = i; - for (i = 0; i < mChannelMasks.size() && i < AUDIO_PORT_MAX_CHANNEL_MASKS; i++) { - if (mChannelMasks[i] != 0) { - port->channel_masks[i] = mChannelMasks[i]; - } - } - port->num_channel_masks = i; - for (i = 0; i < mFormats.size() && i < AUDIO_PORT_MAX_FORMATS; i++) { - if (mFormats[i] != 0) { - port->formats[i] = mFormats[i]; - } - } - port->num_formats = i; - - ALOGV("AudioPort::toAudioPort() num gains %zu", mGains.size()); - - for (i = 0; i < mGains.size() && i < AUDIO_PORT_MAX_GAINS; i++) { - port->gains[i] = mGains[i]->mGain; - } - port->num_gains = i; -} - -void AudioPolicyManager::AudioPort::importAudioPort(const sp port) { - for (size_t k = 0 ; k < port->mSamplingRates.size() ; k++) { - const uint32_t rate = port->mSamplingRates.itemAt(k); - if (rate != 0) { // skip "dynamic" rates - bool hasRate = false; - for (size_t l = 0 ; l < mSamplingRates.size() ; l++) { - if (rate == mSamplingRates.itemAt(l)) { - hasRate = true; - break; - } - } - if (!hasRate) { // never import a sampling rate twice - mSamplingRates.add(rate); - } - } - } - for (size_t k = 0 ; k < port->mChannelMasks.size() ; k++) { - const audio_channel_mask_t mask = port->mChannelMasks.itemAt(k); - if (mask != 0) { // skip "dynamic" masks - bool hasMask = false; - for (size_t l = 0 ; l < mChannelMasks.size() ; l++) { - if (mask == mChannelMasks.itemAt(l)) { - hasMask = true; - break; - } - } - if (!hasMask) { // never import a channel mask twice - mChannelMasks.add(mask); - } - } - } - for (size_t k = 0 ; k < port->mFormats.size() ; k++) { - const audio_format_t format = port->mFormats.itemAt(k); - if (format != 0) { // skip "dynamic" formats - bool hasFormat = false; - for (size_t l = 0 ; l < mFormats.size() ; l++) { - if (format == mFormats.itemAt(l)) { - hasFormat = true; - break; - } - } - if (!hasFormat) { // never import a channel mask twice - mFormats.add(format); - } - } - } - for (size_t k = 0 ; k < port->mGains.size() ; k++) { - sp gain = port->mGains.itemAt(k); - if (gain != 0) { - bool hasGain = false; - for (size_t l = 0 ; l < mGains.size() ; l++) { - if (gain == mGains.itemAt(l)) { - hasGain = true; - break; - } - } - if (!hasGain) { // never import a gain twice - mGains.add(gain); - } - } - } -} - -void AudioPolicyManager::AudioPort::clearCapabilities() { - mChannelMasks.clear(); - mFormats.clear(); - mSamplingRates.clear(); - mGains.clear(); -} - -void AudioPolicyManager::AudioPort::loadSamplingRates(char *name) -{ - char *str = strtok(name, "|"); - - // by convention, "0' in the first entry in mSamplingRates indicates the supported sampling - // rates should be read from the output stream after it is opened for the first time - if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) { - mSamplingRates.add(0); - return; - } - - while (str != NULL) { - uint32_t rate = atoi(str); - if (rate != 0) { - ALOGV("loadSamplingRates() adding rate %d", rate); - mSamplingRates.add(rate); - } - str = strtok(NULL, "|"); - } -} - -void AudioPolicyManager::AudioPort::loadFormats(char *name) -{ - char *str = strtok(name, "|"); - - // by convention, "0' in the first entry in mFormats indicates the supported formats - // should be read from the output stream after it is opened for the first time - if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) { - mFormats.add(AUDIO_FORMAT_DEFAULT); - return; - } - - while (str != NULL) { - audio_format_t format = (audio_format_t)stringToEnum(sFormatNameToEnumTable, - ARRAY_SIZE(sFormatNameToEnumTable), - str); - if (format != AUDIO_FORMAT_DEFAULT) { - mFormats.add(format); - } - str = strtok(NULL, "|"); - } -} - -void AudioPolicyManager::AudioPort::loadInChannels(char *name) -{ - const char *str = strtok(name, "|"); - - ALOGV("loadInChannels() %s", name); - - if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) { - mChannelMasks.add(0); - return; - } - - while (str != NULL) { - audio_channel_mask_t channelMask = - (audio_channel_mask_t)stringToEnum(sInChannelsNameToEnumTable, - ARRAY_SIZE(sInChannelsNameToEnumTable), - str); - if (channelMask != 0) { - ALOGV("loadInChannels() adding channelMask %04x", channelMask); - mChannelMasks.add(channelMask); - } - str = strtok(NULL, "|"); - } -} - -void AudioPolicyManager::AudioPort::loadOutChannels(char *name) -{ - const char *str = strtok(name, "|"); - - ALOGV("loadOutChannels() %s", name); - - // by convention, "0' in the first entry in mChannelMasks indicates the supported channel - // masks should be read from the output stream after it is opened for the first time - if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) { - mChannelMasks.add(0); - return; - } - - while (str != NULL) { - audio_channel_mask_t channelMask = - (audio_channel_mask_t)stringToEnum(sOutChannelsNameToEnumTable, - ARRAY_SIZE(sOutChannelsNameToEnumTable), - str); - if (channelMask != 0) { - mChannelMasks.add(channelMask); - } - str = strtok(NULL, "|"); - } - return; -} - -audio_gain_mode_t AudioPolicyManager::AudioPort::loadGainMode(char *name) -{ - const char *str = strtok(name, "|"); - - ALOGV("loadGainMode() %s", name); - audio_gain_mode_t mode = 0; - while (str != NULL) { - mode |= (audio_gain_mode_t)stringToEnum(sGainModeNameToEnumTable, - ARRAY_SIZE(sGainModeNameToEnumTable), - str); - str = strtok(NULL, "|"); - } - return mode; -} - -void AudioPolicyManager::AudioPort::loadGain(cnode *root, int index) -{ - cnode *node = root->first_child; - - sp gain = new AudioGain(index, mUseInChannelMask); - - while (node) { - if (strcmp(node->name, GAIN_MODE) == 0) { - gain->mGain.mode = loadGainMode((char *)node->value); - } else if (strcmp(node->name, GAIN_CHANNELS) == 0) { - if (mUseInChannelMask) { - gain->mGain.channel_mask = - (audio_channel_mask_t)stringToEnum(sInChannelsNameToEnumTable, - ARRAY_SIZE(sInChannelsNameToEnumTable), - (char *)node->value); - } else { - gain->mGain.channel_mask = - (audio_channel_mask_t)stringToEnum(sOutChannelsNameToEnumTable, - ARRAY_SIZE(sOutChannelsNameToEnumTable), - (char *)node->value); - } - } else if (strcmp(node->name, GAIN_MIN_VALUE) == 0) { - gain->mGain.min_value = atoi((char *)node->value); - } else if (strcmp(node->name, GAIN_MAX_VALUE) == 0) { - gain->mGain.max_value = atoi((char *)node->value); - } else if (strcmp(node->name, GAIN_DEFAULT_VALUE) == 0) { - gain->mGain.default_value = atoi((char *)node->value); - } else if (strcmp(node->name, GAIN_STEP_VALUE) == 0) { - gain->mGain.step_value = atoi((char *)node->value); - } else if (strcmp(node->name, GAIN_MIN_RAMP_MS) == 0) { - gain->mGain.min_ramp_ms = atoi((char *)node->value); - } else if (strcmp(node->name, GAIN_MAX_RAMP_MS) == 0) { - gain->mGain.max_ramp_ms = atoi((char *)node->value); - } - node = node->next; - } - - ALOGV("loadGain() adding new gain mode %08x channel mask %08x min mB %d max mB %d", - gain->mGain.mode, gain->mGain.channel_mask, gain->mGain.min_value, gain->mGain.max_value); - - if (gain->mGain.mode == 0) { - return; - } - mGains.add(gain); -} - -void AudioPolicyManager::AudioPort::loadGains(cnode *root) -{ - cnode *node = root->first_child; - int index = 0; - while (node) { - ALOGV("loadGains() loading gain %s", node->name); - loadGain(node, index++); - node = node->next; - } -} - -status_t AudioPolicyManager::AudioPort::checkExactSamplingRate(uint32_t samplingRate) const -{ - if (mSamplingRates.isEmpty()) { - return NO_ERROR; - } - - for (size_t i = 0; i < mSamplingRates.size(); i ++) { - if (mSamplingRates[i] == samplingRate) { - return NO_ERROR; - } - } - return BAD_VALUE; -} - -status_t AudioPolicyManager::AudioPort::checkCompatibleSamplingRate(uint32_t samplingRate, - uint32_t *updatedSamplingRate) const -{ - if (mSamplingRates.isEmpty()) { - return NO_ERROR; - } - - // Search for the closest supported sampling rate that is above (preferred) - // or below (acceptable) the desired sampling rate, within a permitted ratio. - // The sampling rates do not need to be sorted in ascending order. - ssize_t maxBelow = -1; - ssize_t minAbove = -1; - uint32_t candidate; - for (size_t i = 0; i < mSamplingRates.size(); i++) { - candidate = mSamplingRates[i]; - if (candidate == samplingRate) { - if (updatedSamplingRate != NULL) { - *updatedSamplingRate = candidate; - } - return NO_ERROR; - } - // candidate < desired - if (candidate < samplingRate) { - if (maxBelow < 0 || candidate > mSamplingRates[maxBelow]) { - maxBelow = i; - } - // candidate > desired - } else { - if (minAbove < 0 || candidate < mSamplingRates[minAbove]) { - minAbove = i; - } - } - } - // This uses hard-coded knowledge about AudioFlinger resampling ratios. - // TODO Move these assumptions out. - static const uint32_t kMaxDownSampleRatio = 6; // beyond this aliasing occurs - static const uint32_t kMaxUpSampleRatio = 256; // beyond this sample rate inaccuracies occur - // due to approximation by an int32_t of the - // phase increments - // Prefer to down-sample from a higher sampling rate, as we get the desired frequency spectrum. - if (minAbove >= 0) { - candidate = mSamplingRates[minAbove]; - if (candidate / kMaxDownSampleRatio <= samplingRate) { - if (updatedSamplingRate != NULL) { - *updatedSamplingRate = candidate; - } - return NO_ERROR; - } - } - // But if we have to up-sample from a lower sampling rate, that's OK. - if (maxBelow >= 0) { - candidate = mSamplingRates[maxBelow]; - if (candidate * kMaxUpSampleRatio >= samplingRate) { - if (updatedSamplingRate != NULL) { - *updatedSamplingRate = candidate; - } - return NO_ERROR; - } - } - // leave updatedSamplingRate unmodified - return BAD_VALUE; -} - -status_t AudioPolicyManager::AudioPort::checkExactChannelMask(audio_channel_mask_t channelMask) const -{ - if (mChannelMasks.isEmpty()) { - return NO_ERROR; - } - - for (size_t i = 0; i < mChannelMasks.size(); i++) { - if (mChannelMasks[i] == channelMask) { - return NO_ERROR; - } - } - return BAD_VALUE; -} - -status_t AudioPolicyManager::AudioPort::checkCompatibleChannelMask(audio_channel_mask_t channelMask) - const -{ - if (mChannelMasks.isEmpty()) { - return NO_ERROR; - } - - const bool isRecordThread = mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SINK; - for (size_t i = 0; i < mChannelMasks.size(); i ++) { - // FIXME Does not handle multi-channel automatic conversions yet - audio_channel_mask_t supported = mChannelMasks[i]; - if (supported == channelMask) { - return NO_ERROR; - } - if (isRecordThread) { - // This uses hard-coded knowledge that AudioFlinger can silently down-mix and up-mix. - // FIXME Abstract this out to a table. - if (((supported == AUDIO_CHANNEL_IN_FRONT_BACK || supported == AUDIO_CHANNEL_IN_STEREO) - && channelMask == AUDIO_CHANNEL_IN_MONO) || - (supported == AUDIO_CHANNEL_IN_MONO && (channelMask == AUDIO_CHANNEL_IN_FRONT_BACK - || channelMask == AUDIO_CHANNEL_IN_STEREO))) { - return NO_ERROR; - } - } - } - return BAD_VALUE; -} - -status_t AudioPolicyManager::AudioPort::checkFormat(audio_format_t format) const -{ - if (mFormats.isEmpty()) { - return NO_ERROR; - } - - for (size_t i = 0; i < mFormats.size(); i ++) { - if (mFormats[i] == format) { - return NO_ERROR; - } - } - return BAD_VALUE; -} - - -uint32_t AudioPolicyManager::AudioPort::pickSamplingRate() const -{ - // special case for uninitialized dynamic profile - if (mSamplingRates.size() == 1 && mSamplingRates[0] == 0) { - return 0; - } - - // For direct outputs, pick minimum sampling rate: this helps ensuring that the - // channel count / sampling rate combination chosen will be supported by the connected - // sink - if ((mType == AUDIO_PORT_TYPE_MIX) && (mRole == AUDIO_PORT_ROLE_SOURCE) && - (mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD))) { - uint32_t samplingRate = UINT_MAX; - for (size_t i = 0; i < mSamplingRates.size(); i ++) { - if ((mSamplingRates[i] < samplingRate) && (mSamplingRates[i] > 0)) { - samplingRate = mSamplingRates[i]; - } - } - return (samplingRate == UINT_MAX) ? 0 : samplingRate; - } - - uint32_t samplingRate = 0; - uint32_t maxRate = MAX_MIXER_SAMPLING_RATE; - - // For mixed output and inputs, use max mixer sampling rates. Do not - // limit sampling rate otherwise - if (mType != AUDIO_PORT_TYPE_MIX) { - maxRate = UINT_MAX; - } - for (size_t i = 0; i < mSamplingRates.size(); i ++) { - if ((mSamplingRates[i] > samplingRate) && (mSamplingRates[i] <= maxRate)) { - samplingRate = mSamplingRates[i]; - } - } - return samplingRate; -} - -audio_channel_mask_t AudioPolicyManager::AudioPort::pickChannelMask() const -{ - // special case for uninitialized dynamic profile - if (mChannelMasks.size() == 1 && mChannelMasks[0] == 0) { - return AUDIO_CHANNEL_NONE; - } - audio_channel_mask_t channelMask = AUDIO_CHANNEL_NONE; - - // For direct outputs, pick minimum channel count: this helps ensuring that the - // channel count / sampling rate combination chosen will be supported by the connected - // sink - if ((mType == AUDIO_PORT_TYPE_MIX) && (mRole == AUDIO_PORT_ROLE_SOURCE) && - (mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD))) { - uint32_t channelCount = UINT_MAX; - for (size_t i = 0; i < mChannelMasks.size(); i ++) { - uint32_t cnlCount; - if (mUseInChannelMask) { - cnlCount = audio_channel_count_from_in_mask(mChannelMasks[i]); - } else { - cnlCount = audio_channel_count_from_out_mask(mChannelMasks[i]); - } - if ((cnlCount < channelCount) && (cnlCount > 0)) { - channelMask = mChannelMasks[i]; - channelCount = cnlCount; - } - } - return channelMask; - } - - uint32_t channelCount = 0; - uint32_t maxCount = MAX_MIXER_CHANNEL_COUNT; - - // For mixed output and inputs, use max mixer channel count. Do not - // limit channel count otherwise - if (mType != AUDIO_PORT_TYPE_MIX) { - maxCount = UINT_MAX; - } - for (size_t i = 0; i < mChannelMasks.size(); i ++) { - uint32_t cnlCount; - if (mUseInChannelMask) { - cnlCount = audio_channel_count_from_in_mask(mChannelMasks[i]); - } else { - cnlCount = audio_channel_count_from_out_mask(mChannelMasks[i]); - } - if ((cnlCount > channelCount) && (cnlCount <= maxCount)) { - channelMask = mChannelMasks[i]; - channelCount = cnlCount; - } - } - return channelMask; -} - -/* format in order of increasing preference */ -const audio_format_t AudioPolicyManager::AudioPort::sPcmFormatCompareTable[] = { - AUDIO_FORMAT_DEFAULT, - AUDIO_FORMAT_PCM_16_BIT, - AUDIO_FORMAT_PCM_8_24_BIT, - AUDIO_FORMAT_PCM_24_BIT_PACKED, - AUDIO_FORMAT_PCM_32_BIT, - AUDIO_FORMAT_PCM_FLOAT, -}; - -int AudioPolicyManager::AudioPort::compareFormats(audio_format_t format1, - audio_format_t format2) -{ - // NOTE: AUDIO_FORMAT_INVALID is also considered not PCM and will be compared equal to any - // compressed format and better than any PCM format. This is by design of pickFormat() - if (!audio_is_linear_pcm(format1)) { - if (!audio_is_linear_pcm(format2)) { - return 0; - } - return 1; - } - if (!audio_is_linear_pcm(format2)) { - return -1; - } - - int index1 = -1, index2 = -1; - for (size_t i = 0; - (i < ARRAY_SIZE(sPcmFormatCompareTable)) && ((index1 == -1) || (index2 == -1)); - i ++) { - if (sPcmFormatCompareTable[i] == format1) { - index1 = i; - } - if (sPcmFormatCompareTable[i] == format2) { - index2 = i; - } - } - // format1 not found => index1 < 0 => format2 > format1 - // format2 not found => index2 < 0 => format2 < format1 - return index1 - index2; -} - -audio_format_t AudioPolicyManager::AudioPort::pickFormat() const -{ - // special case for uninitialized dynamic profile - if (mFormats.size() == 1 && mFormats[0] == 0) { - return AUDIO_FORMAT_DEFAULT; - } - - audio_format_t format = AUDIO_FORMAT_DEFAULT; - audio_format_t bestFormat = - AudioPolicyManager::AudioPort::sPcmFormatCompareTable[ - ARRAY_SIZE(AudioPolicyManager::AudioPort::sPcmFormatCompareTable) - 1]; - // For mixed output and inputs, use best mixer output format. Do not - // limit format otherwise - if ((mType != AUDIO_PORT_TYPE_MIX) || - ((mRole == AUDIO_PORT_ROLE_SOURCE) && - (((mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)) != 0)))) { - bestFormat = AUDIO_FORMAT_INVALID; - } - - for (size_t i = 0; i < mFormats.size(); i ++) { - if ((compareFormats(mFormats[i], format) > 0) && - (compareFormats(mFormats[i], bestFormat) <= 0)) { - format = mFormats[i]; - } - } - return format; -} - -status_t AudioPolicyManager::AudioPort::checkGain(const struct audio_gain_config *gainConfig, - int index) const -{ - if (index < 0 || (size_t)index >= mGains.size()) { - return BAD_VALUE; - } - return mGains[index]->checkConfig(gainConfig); -} - -void AudioPolicyManager::AudioPort::dump(int fd, int spaces) const -{ - const size_t SIZE = 256; - char buffer[SIZE]; - String8 result; - - if (mName.size() != 0) { - snprintf(buffer, SIZE, "%*s- name: %s\n", spaces, "", mName.string()); - result.append(buffer); - } - - if (mSamplingRates.size() != 0) { - snprintf(buffer, SIZE, "%*s- sampling rates: ", spaces, ""); - result.append(buffer); - for (size_t i = 0; i < mSamplingRates.size(); i++) { - if (i == 0 && mSamplingRates[i] == 0) { - snprintf(buffer, SIZE, "Dynamic"); - } else { - snprintf(buffer, SIZE, "%d", mSamplingRates[i]); - } - result.append(buffer); - result.append(i == (mSamplingRates.size() - 1) ? "" : ", "); - } - result.append("\n"); - } - - if (mChannelMasks.size() != 0) { - snprintf(buffer, SIZE, "%*s- channel masks: ", spaces, ""); - result.append(buffer); - for (size_t i = 0; i < mChannelMasks.size(); i++) { - ALOGV("AudioPort::dump mChannelMasks %zu %08x", i, mChannelMasks[i]); - - if (i == 0 && mChannelMasks[i] == 0) { - snprintf(buffer, SIZE, "Dynamic"); - } else { - snprintf(buffer, SIZE, "0x%04x", mChannelMasks[i]); - } - result.append(buffer); - result.append(i == (mChannelMasks.size() - 1) ? "" : ", "); - } - result.append("\n"); - } - - if (mFormats.size() != 0) { - snprintf(buffer, SIZE, "%*s- formats: ", spaces, ""); - result.append(buffer); - for (size_t i = 0; i < mFormats.size(); i++) { - const char *formatStr = enumToString(sFormatNameToEnumTable, - ARRAY_SIZE(sFormatNameToEnumTable), - mFormats[i]); - if (i == 0 && strcmp(formatStr, "") == 0) { - snprintf(buffer, SIZE, "Dynamic"); - } else { - snprintf(buffer, SIZE, "%s", formatStr); - } - result.append(buffer); - result.append(i == (mFormats.size() - 1) ? "" : ", "); - } - result.append("\n"); - } - write(fd, result.string(), result.size()); - if (mGains.size() != 0) { - snprintf(buffer, SIZE, "%*s- gains:\n", spaces, ""); - write(fd, buffer, strlen(buffer) + 1); - result.append(buffer); - for (size_t i = 0; i < mGains.size(); i++) { - mGains[i]->dump(fd, spaces + 2, i); - } - } -} - -// --- AudioGain class implementation - -AudioPolicyManager::AudioGain::AudioGain(int index, bool useInChannelMask) -{ - mIndex = index; - mUseInChannelMask = useInChannelMask; - memset(&mGain, 0, sizeof(struct audio_gain)); -} - -void AudioPolicyManager::AudioGain::getDefaultConfig(struct audio_gain_config *config) -{ - config->index = mIndex; - config->mode = mGain.mode; - config->channel_mask = mGain.channel_mask; - if ((mGain.mode & AUDIO_GAIN_MODE_JOINT) == AUDIO_GAIN_MODE_JOINT) { - config->values[0] = mGain.default_value; - } else { - uint32_t numValues; - if (mUseInChannelMask) { - numValues = audio_channel_count_from_in_mask(mGain.channel_mask); - } else { - numValues = audio_channel_count_from_out_mask(mGain.channel_mask); - } - for (size_t i = 0; i < numValues; i++) { - config->values[i] = mGain.default_value; - } - } - if ((mGain.mode & AUDIO_GAIN_MODE_RAMP) == AUDIO_GAIN_MODE_RAMP) { - config->ramp_duration_ms = mGain.min_ramp_ms; - } -} - -status_t AudioPolicyManager::AudioGain::checkConfig(const struct audio_gain_config *config) -{ - if ((config->mode & ~mGain.mode) != 0) { - return BAD_VALUE; - } - if ((config->mode & AUDIO_GAIN_MODE_JOINT) == AUDIO_GAIN_MODE_JOINT) { - if ((config->values[0] < mGain.min_value) || - (config->values[0] > mGain.max_value)) { - return BAD_VALUE; - } - } else { - if ((config->channel_mask & ~mGain.channel_mask) != 0) { - return BAD_VALUE; - } - uint32_t numValues; - if (mUseInChannelMask) { - numValues = audio_channel_count_from_in_mask(config->channel_mask); - } else { - numValues = audio_channel_count_from_out_mask(config->channel_mask); - } - for (size_t i = 0; i < numValues; i++) { - if ((config->values[i] < mGain.min_value) || - (config->values[i] > mGain.max_value)) { - return BAD_VALUE; - } - } - } - if ((config->mode & AUDIO_GAIN_MODE_RAMP) == AUDIO_GAIN_MODE_RAMP) { - if ((config->ramp_duration_ms < mGain.min_ramp_ms) || - (config->ramp_duration_ms > mGain.max_ramp_ms)) { - return BAD_VALUE; - } - } - return NO_ERROR; -} - -void AudioPolicyManager::AudioGain::dump(int fd, int spaces, int index) const -{ - const size_t SIZE = 256; - char buffer[SIZE]; - String8 result; - - snprintf(buffer, SIZE, "%*sGain %d:\n", spaces, "", index+1); - result.append(buffer); - snprintf(buffer, SIZE, "%*s- mode: %08x\n", spaces, "", mGain.mode); - result.append(buffer); - snprintf(buffer, SIZE, "%*s- channel_mask: %08x\n", spaces, "", mGain.channel_mask); - result.append(buffer); - snprintf(buffer, SIZE, "%*s- min_value: %d mB\n", spaces, "", mGain.min_value); - result.append(buffer); - snprintf(buffer, SIZE, "%*s- max_value: %d mB\n", spaces, "", mGain.max_value); - result.append(buffer); - snprintf(buffer, SIZE, "%*s- default_value: %d mB\n", spaces, "", mGain.default_value); - result.append(buffer); - snprintf(buffer, SIZE, "%*s- step_value: %d mB\n", spaces, "", mGain.step_value); - result.append(buffer); - snprintf(buffer, SIZE, "%*s- min_ramp_ms: %d ms\n", spaces, "", mGain.min_ramp_ms); - result.append(buffer); - snprintf(buffer, SIZE, "%*s- max_ramp_ms: %d ms\n", spaces, "", mGain.max_ramp_ms); - result.append(buffer); - - write(fd, result.string(), result.size()); -} - -// --- AudioPortConfig class implementation - -AudioPolicyManager::AudioPortConfig::AudioPortConfig() -{ - mSamplingRate = 0; - mChannelMask = AUDIO_CHANNEL_NONE; - mFormat = AUDIO_FORMAT_INVALID; - mGain.index = -1; -} - -status_t AudioPolicyManager::AudioPortConfig::applyAudioPortConfig( - const struct audio_port_config *config, - struct audio_port_config *backupConfig) -{ - struct audio_port_config localBackupConfig; - status_t status = NO_ERROR; - - localBackupConfig.config_mask = config->config_mask; - toAudioPortConfig(&localBackupConfig); - - sp audioport = getAudioPort(); - if (audioport == 0) { - status = NO_INIT; - goto exit; - } - if (config->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) { - status = audioport->checkExactSamplingRate(config->sample_rate); - if (status != NO_ERROR) { - goto exit; - } - mSamplingRate = config->sample_rate; - } - if (config->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) { - status = audioport->checkExactChannelMask(config->channel_mask); - if (status != NO_ERROR) { - goto exit; - } - mChannelMask = config->channel_mask; - } - if (config->config_mask & AUDIO_PORT_CONFIG_FORMAT) { - status = audioport->checkFormat(config->format); - if (status != NO_ERROR) { - goto exit; - } - mFormat = config->format; - } - if (config->config_mask & AUDIO_PORT_CONFIG_GAIN) { - status = audioport->checkGain(&config->gain, config->gain.index); - if (status != NO_ERROR) { - goto exit; - } - mGain = config->gain; - } - -exit: - if (status != NO_ERROR) { - applyAudioPortConfig(&localBackupConfig); - } - if (backupConfig != NULL) { - *backupConfig = localBackupConfig; - } - return status; -} - -void AudioPolicyManager::AudioPortConfig::toAudioPortConfig( - struct audio_port_config *dstConfig, - const struct audio_port_config *srcConfig) const -{ - if (dstConfig->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) { - dstConfig->sample_rate = mSamplingRate; - if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE)) { - dstConfig->sample_rate = srcConfig->sample_rate; - } - } else { - dstConfig->sample_rate = 0; - } - if (dstConfig->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) { - dstConfig->channel_mask = mChannelMask; - if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK)) { - dstConfig->channel_mask = srcConfig->channel_mask; - } - } else { - dstConfig->channel_mask = AUDIO_CHANNEL_NONE; - } - if (dstConfig->config_mask & AUDIO_PORT_CONFIG_FORMAT) { - dstConfig->format = mFormat; - if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_FORMAT)) { - dstConfig->format = srcConfig->format; - } - } else { - dstConfig->format = AUDIO_FORMAT_INVALID; - } - if (dstConfig->config_mask & AUDIO_PORT_CONFIG_GAIN) { - dstConfig->gain = mGain; - if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_GAIN)) { - dstConfig->gain = srcConfig->gain; - } - } else { - dstConfig->gain.index = -1; - } - if (dstConfig->gain.index != -1) { - dstConfig->config_mask |= AUDIO_PORT_CONFIG_GAIN; - } else { - dstConfig->config_mask &= ~AUDIO_PORT_CONFIG_GAIN; - } -} - -// --- IOProfile class implementation - -AudioPolicyManager::IOProfile::IOProfile(const String8& name, audio_port_role_t role, - const sp& module) - : AudioPort(name, AUDIO_PORT_TYPE_MIX, role, module) -{ -} - -AudioPolicyManager::IOProfile::~IOProfile() -{ -} - -// checks if the IO profile is compatible with specified parameters. -// Sampling rate, format and channel mask must be specified in order to -// get a valid a match -bool AudioPolicyManager::IOProfile::isCompatibleProfile(audio_devices_t device, - String8 address, - uint32_t samplingRate, - uint32_t *updatedSamplingRate, - audio_format_t format, - audio_channel_mask_t channelMask, - uint32_t flags) const -{ - const bool isPlaybackThread = mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SOURCE; - const bool isRecordThread = mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SINK; - ALOG_ASSERT(isPlaybackThread != isRecordThread); - - if (device != AUDIO_DEVICE_NONE && mSupportedDevices.getDevice(device, address) == 0) { - return false; - } - - if (samplingRate == 0) { - return false; - } - uint32_t myUpdatedSamplingRate = samplingRate; - if (isPlaybackThread && checkExactSamplingRate(samplingRate) != NO_ERROR) { - return false; - } - if (isRecordThread && checkCompatibleSamplingRate(samplingRate, &myUpdatedSamplingRate) != - NO_ERROR) { - return false; - } - - if (!audio_is_valid_format(format) || checkFormat(format) != NO_ERROR) { - return false; - } - - if (isPlaybackThread && (!audio_is_output_channel(channelMask) || - checkExactChannelMask(channelMask) != NO_ERROR)) { - return false; - } - if (isRecordThread && (!audio_is_input_channel(channelMask) || - checkCompatibleChannelMask(channelMask) != NO_ERROR)) { - return false; - } - - if (isPlaybackThread && (mFlags & flags) != flags) { - return false; - } - // The only input flag that is allowed to be different is the fast flag. - // An existing fast stream is compatible with a normal track request. - // An existing normal stream is compatible with a fast track request, - // but the fast request will be denied by AudioFlinger and converted to normal track. - if (isRecordThread && ((mFlags ^ flags) & - ~AUDIO_INPUT_FLAG_FAST)) { - return false; - } - - if (updatedSamplingRate != NULL) { - *updatedSamplingRate = myUpdatedSamplingRate; - } - return true; -} - -void AudioPolicyManager::IOProfile::dump(int fd) -{ - const size_t SIZE = 256; - char buffer[SIZE]; - String8 result; - - AudioPort::dump(fd, 4); - - snprintf(buffer, SIZE, " - flags: 0x%04x\n", mFlags); - result.append(buffer); - snprintf(buffer, SIZE, " - devices:\n"); - result.append(buffer); - write(fd, result.string(), result.size()); - for (size_t i = 0; i < mSupportedDevices.size(); i++) { - mSupportedDevices[i]->dump(fd, 6, i); - } -} - -void AudioPolicyManager::IOProfile::log() -{ - const size_t SIZE = 256; - char buffer[SIZE]; - String8 result; - - ALOGV(" - sampling rates: "); - for (size_t i = 0; i < mSamplingRates.size(); i++) { - ALOGV(" %d", mSamplingRates[i]); - } - - ALOGV(" - channel masks: "); - for (size_t i = 0; i < mChannelMasks.size(); i++) { - ALOGV(" 0x%04x", mChannelMasks[i]); - } - - ALOGV(" - formats: "); - for (size_t i = 0; i < mFormats.size(); i++) { - ALOGV(" 0x%08x", mFormats[i]); - } - - ALOGV(" - devices: 0x%04x\n", mSupportedDevices.types()); - ALOGV(" - flags: 0x%04x\n", mFlags); -} - - -// --- DeviceDescriptor implementation - -String8 AudioPolicyManager::DeviceDescriptor::emptyNameStr = String8(""); - -AudioPolicyManager::DeviceDescriptor::DeviceDescriptor(const String8& name, audio_devices_t type) : - AudioPort(name, AUDIO_PORT_TYPE_DEVICE, - audio_is_output_device(type) ? AUDIO_PORT_ROLE_SINK : - AUDIO_PORT_ROLE_SOURCE, - NULL), - mDeviceType(type), mAddress("") -{ -} - -bool AudioPolicyManager::DeviceDescriptor::equals(const sp& other) const -{ - // Devices are considered equal if they: - // - are of the same type (a device type cannot be AUDIO_DEVICE_NONE) - // - have the same address or one device does not specify the address - // - have the same channel mask or one device does not specify the channel mask - return (mDeviceType == other->mDeviceType) && - (mAddress == "" || other->mAddress == "" || mAddress == other->mAddress) && - (mChannelMask == 0 || other->mChannelMask == 0 || - mChannelMask == other->mChannelMask); -} - -void AudioPolicyManager::DeviceDescriptor::loadGains(cnode *root) -{ - AudioPort::loadGains(root); - if (mGains.size() > 0) { - mGains[0]->getDefaultConfig(&mGain); - } -} - - -void AudioPolicyManager::DeviceVector::refreshTypes() -{ - mDeviceTypes = AUDIO_DEVICE_NONE; - for(size_t i = 0; i < size(); i++) { - mDeviceTypes |= itemAt(i)->mDeviceType; - } - ALOGV("DeviceVector::refreshTypes() mDeviceTypes %08x", mDeviceTypes); -} - -ssize_t AudioPolicyManager::DeviceVector::indexOf(const sp& item) const -{ - for(size_t i = 0; i < size(); i++) { - if (item->equals(itemAt(i))) { - return i; - } - } - return -1; -} - -ssize_t AudioPolicyManager::DeviceVector::add(const sp& item) -{ - ssize_t ret = indexOf(item); - - if (ret < 0) { - ret = SortedVector::add(item); - if (ret >= 0) { - refreshTypes(); - } - } else { - ALOGW("DeviceVector::add device %08x already in", item->mDeviceType); - ret = -1; - } - return ret; -} - -ssize_t AudioPolicyManager::DeviceVector::remove(const sp& item) -{ - size_t i; - ssize_t ret = indexOf(item); - - if (ret < 0) { - ALOGW("DeviceVector::remove device %08x not in", item->mDeviceType); - } else { - ret = SortedVector::removeAt(ret); - if (ret >= 0) { - refreshTypes(); - } - } - return ret; -} - -void AudioPolicyManager::DeviceVector::loadDevicesFromType(audio_devices_t types) -{ - DeviceVector deviceList; - - uint32_t role_bit = AUDIO_DEVICE_BIT_IN & types; - types &= ~role_bit; - - while (types) { - uint32_t i = 31 - __builtin_clz(types); - uint32_t type = 1 << i; - types &= ~type; - add(new DeviceDescriptor(String8("device_type"), type | role_bit)); - } -} - -void AudioPolicyManager::DeviceVector::loadDevicesFromName(char *name, - const DeviceVector& declaredDevices) -{ - char *devName = strtok(name, "|"); - while (devName != NULL) { - if (strlen(devName) != 0) { - audio_devices_t type = stringToEnum(sDeviceNameToEnumTable, - ARRAY_SIZE(sDeviceNameToEnumTable), - devName); - if (type != AUDIO_DEVICE_NONE) { - sp dev = new DeviceDescriptor(String8(name), type); - if (type == AUDIO_DEVICE_IN_REMOTE_SUBMIX || - type == AUDIO_DEVICE_OUT_REMOTE_SUBMIX ) { - dev->mAddress = String8("0"); - } - add(dev); - } else { - sp deviceDesc = - declaredDevices.getDeviceFromName(String8(devName)); - if (deviceDesc != 0) { - add(deviceDesc); - } - } - } - devName = strtok(NULL, "|"); - } -} - -sp AudioPolicyManager::DeviceVector::getDevice( - audio_devices_t type, String8 address) const -{ - sp device; - for (size_t i = 0; i < size(); i++) { - if (itemAt(i)->mDeviceType == type) { - if (address == "" || itemAt(i)->mAddress == address) { - device = itemAt(i); - if (itemAt(i)->mAddress == address) { - break; - } - } - } - } - ALOGV("DeviceVector::getDevice() for type %08x address %s found %p", - type, address.string(), device.get()); - return device; -} - -sp AudioPolicyManager::DeviceVector::getDeviceFromId( - audio_port_handle_t id) const -{ - sp device; - for (size_t i = 0; i < size(); i++) { - if (itemAt(i)->getHandle() == id) { - device = itemAt(i); - break; - } - } - return device; -} - -AudioPolicyManager::DeviceVector AudioPolicyManager::DeviceVector::getDevicesFromType( - audio_devices_t type) const -{ - DeviceVector devices; - for (size_t i = 0; (i < size()) && (type != AUDIO_DEVICE_NONE); i++) { - if (itemAt(i)->mDeviceType & type & ~AUDIO_DEVICE_BIT_IN) { - devices.add(itemAt(i)); - type &= ~itemAt(i)->mDeviceType; - ALOGV("DeviceVector::getDevicesFromType() for type %x found %p", - itemAt(i)->mDeviceType, itemAt(i).get()); - } - } - return devices; -} - -AudioPolicyManager::DeviceVector AudioPolicyManager::DeviceVector::getDevicesFromTypeAddr( - audio_devices_t type, String8 address) const -{ - DeviceVector devices; - for (size_t i = 0; i < size(); i++) { - if (itemAt(i)->mDeviceType == type) { - if (itemAt(i)->mAddress == address) { - devices.add(itemAt(i)); - } - } - } - return devices; -} - -sp AudioPolicyManager::DeviceVector::getDeviceFromName( - const String8& name) const -{ - sp device; - for (size_t i = 0; i < size(); i++) { - if (itemAt(i)->mName == name) { - device = itemAt(i); - break; - } - } - return device; -} - -void AudioPolicyManager::DeviceDescriptor::toAudioPortConfig( - struct audio_port_config *dstConfig, - const struct audio_port_config *srcConfig) const -{ - dstConfig->config_mask = AUDIO_PORT_CONFIG_CHANNEL_MASK|AUDIO_PORT_CONFIG_GAIN; - if (srcConfig != NULL) { - dstConfig->config_mask |= srcConfig->config_mask; - } - - AudioPortConfig::toAudioPortConfig(dstConfig, srcConfig); - dstConfig->id = mId; - dstConfig->role = audio_is_output_device(mDeviceType) ? - AUDIO_PORT_ROLE_SINK : AUDIO_PORT_ROLE_SOURCE; - dstConfig->type = AUDIO_PORT_TYPE_DEVICE; - dstConfig->ext.device.type = mDeviceType; - - //TODO Understand why this test is necessary. i.e. why at boot time does it crash - // without the test? - // This has been demonstrated to NOT be true (at start up) - // ALOG_ASSERT(mModule != NULL); - dstConfig->ext.device.hw_module = mModule != NULL ? mModule->mHandle : NULL; - strncpy(dstConfig->ext.device.address, mAddress.string(), AUDIO_DEVICE_MAX_ADDRESS_LEN); -} - -void AudioPolicyManager::DeviceDescriptor::toAudioPort(struct audio_port *port) const -{ - ALOGV("DeviceDescriptor::toAudioPort() handle %d type %x", mId, mDeviceType); - AudioPort::toAudioPort(port); - port->id = mId; - toAudioPortConfig(&port->active_config); - port->ext.device.type = mDeviceType; - port->ext.device.hw_module = mModule->mHandle; - strncpy(port->ext.device.address, mAddress.string(), AUDIO_DEVICE_MAX_ADDRESS_LEN); -} - -status_t AudioPolicyManager::DeviceDescriptor::dump(int fd, int spaces, int index) const -{ - const size_t SIZE = 256; - char buffer[SIZE]; - String8 result; - - snprintf(buffer, SIZE, "%*sDevice %d:\n", spaces, "", index+1); - result.append(buffer); - if (mId != 0) { - snprintf(buffer, SIZE, "%*s- id: %2d\n", spaces, "", mId); - result.append(buffer); - } - snprintf(buffer, SIZE, "%*s- type: %-48s\n", spaces, "", - enumToString(sDeviceNameToEnumTable, - ARRAY_SIZE(sDeviceNameToEnumTable), - mDeviceType)); - result.append(buffer); - if (mAddress.size() != 0) { - snprintf(buffer, SIZE, "%*s- address: %-32s\n", spaces, "", mAddress.string()); - result.append(buffer); - } - write(fd, result.string(), result.size()); - AudioPort::dump(fd, spaces); - - return NO_ERROR; -} - -status_t AudioPolicyManager::AudioPatch::dump(int fd, int spaces, int index) const -{ - const size_t SIZE = 256; - char buffer[SIZE]; - String8 result; - - - snprintf(buffer, SIZE, "%*sAudio patch %d:\n", spaces, "", index+1); - result.append(buffer); - snprintf(buffer, SIZE, "%*s- handle: %2d\n", spaces, "", mHandle); - result.append(buffer); - snprintf(buffer, SIZE, "%*s- audio flinger handle: %2d\n", spaces, "", mAfPatchHandle); - result.append(buffer); - snprintf(buffer, SIZE, "%*s- owner uid: %2d\n", spaces, "", mUid); - result.append(buffer); - snprintf(buffer, SIZE, "%*s- %d sources:\n", spaces, "", mPatch.num_sources); - result.append(buffer); - for (size_t i = 0; i < mPatch.num_sources; i++) { - if (mPatch.sources[i].type == AUDIO_PORT_TYPE_DEVICE) { - snprintf(buffer, SIZE, "%*s- Device ID %d %s\n", spaces + 2, "", - mPatch.sources[i].id, enumToString(sDeviceNameToEnumTable, - ARRAY_SIZE(sDeviceNameToEnumTable), - mPatch.sources[i].ext.device.type)); - } else { - snprintf(buffer, SIZE, "%*s- Mix ID %d I/O handle %d\n", spaces + 2, "", - mPatch.sources[i].id, mPatch.sources[i].ext.mix.handle); - } - result.append(buffer); - } - snprintf(buffer, SIZE, "%*s- %d sinks:\n", spaces, "", mPatch.num_sinks); - result.append(buffer); - for (size_t i = 0; i < mPatch.num_sinks; i++) { - if (mPatch.sinks[i].type == AUDIO_PORT_TYPE_DEVICE) { - snprintf(buffer, SIZE, "%*s- Device ID %d %s\n", spaces + 2, "", - mPatch.sinks[i].id, enumToString(sDeviceNameToEnumTable, - ARRAY_SIZE(sDeviceNameToEnumTable), - mPatch.sinks[i].ext.device.type)); - } else { - snprintf(buffer, SIZE, "%*s- Mix ID %d I/O handle %d\n", spaces + 2, "", - mPatch.sinks[i].id, mPatch.sinks[i].ext.mix.handle); - } - result.append(buffer); - } - - write(fd, result.string(), result.size()); - return NO_ERROR; -} - -// --- audio_policy.conf file parsing - -uint32_t AudioPolicyManager::parseOutputFlagNames(char *name) -{ - uint32_t flag = 0; - - // it is OK to cast name to non const here as we are not going to use it after - // strtok() modifies it - char *flagName = strtok(name, "|"); - while (flagName != NULL) { - if (strlen(flagName) != 0) { - flag |= stringToEnum(sOutputFlagNameToEnumTable, - ARRAY_SIZE(sOutputFlagNameToEnumTable), - flagName); - } - flagName = strtok(NULL, "|"); - } - //force direct flag if offload flag is set: offloading implies a direct output stream - // and all common behaviors are driven by checking only the direct flag - // this should normally be set appropriately in the policy configuration file - if ((flag & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) { - flag |= AUDIO_OUTPUT_FLAG_DIRECT; - } - - return flag; -} - -uint32_t AudioPolicyManager::parseInputFlagNames(char *name) -{ - uint32_t flag = 0; - - // it is OK to cast name to non const here as we are not going to use it after - // strtok() modifies it - char *flagName = strtok(name, "|"); - while (flagName != NULL) { - if (strlen(flagName) != 0) { - flag |= stringToEnum(sInputFlagNameToEnumTable, - ARRAY_SIZE(sInputFlagNameToEnumTable), - flagName); - } - flagName = strtok(NULL, "|"); - } - return flag; -} - -audio_devices_t AudioPolicyManager::parseDeviceNames(char *name) -{ - uint32_t device = 0; - - char *devName = strtok(name, "|"); - while (devName != NULL) { - if (strlen(devName) != 0) { - device |= stringToEnum(sDeviceNameToEnumTable, - ARRAY_SIZE(sDeviceNameToEnumTable), - devName); - } - devName = strtok(NULL, "|"); - } - return device; -} - -void AudioPolicyManager::loadHwModule(cnode *root) -{ - status_t status = NAME_NOT_FOUND; - cnode *node; - sp module = new HwModule(root->name); - - node = config_find(root, DEVICES_TAG); - if (node != NULL) { - node = node->first_child; - while (node) { - ALOGV("loadHwModule() loading device %s", node->name); - status_t tmpStatus = module->loadDevice(node); - if (status == NAME_NOT_FOUND || status == NO_ERROR) { - status = tmpStatus; - } - node = node->next; - } - } - node = config_find(root, OUTPUTS_TAG); - if (node != NULL) { - node = node->first_child; - while (node) { - ALOGV("loadHwModule() loading output %s", node->name); - status_t tmpStatus = module->loadOutput(node); - if (status == NAME_NOT_FOUND || status == NO_ERROR) { - status = tmpStatus; - } - node = node->next; - } - } - node = config_find(root, INPUTS_TAG); - if (node != NULL) { - node = node->first_child; - while (node) { - ALOGV("loadHwModule() loading input %s", node->name); - status_t tmpStatus = module->loadInput(node); - if (status == NAME_NOT_FOUND || status == NO_ERROR) { - status = tmpStatus; - } - node = node->next; - } - } - loadGlobalConfig(root, module); - - if (status == NO_ERROR) { - mHwModules.add(module); - } -} - -void AudioPolicyManager::loadHwModules(cnode *root) -{ - cnode *node = config_find(root, AUDIO_HW_MODULE_TAG); - if (node == NULL) { - return; - } - - node = node->first_child; - while (node) { - ALOGV("loadHwModules() loading module %s", node->name); - loadHwModule(node); - node = node->next; - } -} - -void AudioPolicyManager::loadGlobalConfig(cnode *root, const sp& module) -{ - cnode *node = config_find(root, GLOBAL_CONFIG_TAG); - - if (node == NULL) { - return; - } - DeviceVector declaredDevices; - if (module != NULL) { - declaredDevices = module->mDeclaredDevices; - } - - node = node->first_child; - while (node) { - if (strcmp(ATTACHED_OUTPUT_DEVICES_TAG, node->name) == 0) { - mAvailableOutputDevices.loadDevicesFromName((char *)node->value, - declaredDevices); - ALOGV("loadGlobalConfig() Attached Output Devices %08x", - mAvailableOutputDevices.types()); - } else if (strcmp(DEFAULT_OUTPUT_DEVICE_TAG, node->name) == 0) { - audio_devices_t device = (audio_devices_t)stringToEnum(sDeviceNameToEnumTable, - ARRAY_SIZE(sDeviceNameToEnumTable), - (char *)node->value); - if (device != AUDIO_DEVICE_NONE) { - mDefaultOutputDevice = new DeviceDescriptor(String8("default-output"), device); - } else { - ALOGW("loadGlobalConfig() default device not specified"); - } - ALOGV("loadGlobalConfig() mDefaultOutputDevice %08x", mDefaultOutputDevice->mDeviceType); - } else if (strcmp(ATTACHED_INPUT_DEVICES_TAG, node->name) == 0) { - mAvailableInputDevices.loadDevicesFromName((char *)node->value, - declaredDevices); - ALOGV("loadGlobalConfig() Available InputDevices %08x", mAvailableInputDevices.types()); - } else if (strcmp(SPEAKER_DRC_ENABLED_TAG, node->name) == 0) { - mSpeakerDrcEnabled = stringToBool((char *)node->value); - ALOGV("loadGlobalConfig() mSpeakerDrcEnabled = %d", mSpeakerDrcEnabled); - } else if (strcmp(AUDIO_HAL_VERSION_TAG, node->name) == 0) { - uint32_t major, minor; - sscanf((char *)node->value, "%u.%u", &major, &minor); - module->mHalVersion = HARDWARE_DEVICE_API_VERSION(major, minor); - ALOGV("loadGlobalConfig() mHalVersion = %04x major %u minor %u", - module->mHalVersion, major, minor); - } - node = node->next; - } -} - -status_t AudioPolicyManager::loadAudioPolicyConfig(const char *path) -{ - cnode *root; - char *data; - - data = (char *)load_file(path, NULL); - if (data == NULL) { - return -ENODEV; - } - root = config_node("", ""); - config_load(root, data); - - loadHwModules(root); - // legacy audio_policy.conf files have one global_configuration section - loadGlobalConfig(root, getModuleFromName(AUDIO_HARDWARE_MODULE_ID_PRIMARY)); - config_free(root); - free(root); - free(data); - - ALOGI("loadAudioPolicyConfig() loaded %s\n", path); - - return NO_ERROR; -} - -void AudioPolicyManager::defaultAudioPolicyConfig(void) -{ - sp module; - sp profile; - sp defaultInputDevice = - new DeviceDescriptor(String8("builtin-mic"), AUDIO_DEVICE_IN_BUILTIN_MIC); - mAvailableOutputDevices.add(mDefaultOutputDevice); - mAvailableInputDevices.add(defaultInputDevice); - - module = new HwModule("primary"); - - profile = new IOProfile(String8("primary"), AUDIO_PORT_ROLE_SOURCE, module); - profile->mSamplingRates.add(44100); - profile->mFormats.add(AUDIO_FORMAT_PCM_16_BIT); - profile->mChannelMasks.add(AUDIO_CHANNEL_OUT_STEREO); - profile->mSupportedDevices.add(mDefaultOutputDevice); - profile->mFlags = AUDIO_OUTPUT_FLAG_PRIMARY; - module->mOutputProfiles.add(profile); - - profile = new IOProfile(String8("primary"), AUDIO_PORT_ROLE_SINK, module); - profile->mSamplingRates.add(8000); - profile->mFormats.add(AUDIO_FORMAT_PCM_16_BIT); - profile->mChannelMasks.add(AUDIO_CHANNEL_IN_MONO); - profile->mSupportedDevices.add(defaultInputDevice); - module->mInputProfiles.add(profile); - - mHwModules.add(module); -} - -audio_stream_type_t AudioPolicyManager::streamTypefromAttributesInt(const audio_attributes_t *attr) -{ - // flags to stream type mapping - if ((attr->flags & AUDIO_FLAG_AUDIBILITY_ENFORCED) == AUDIO_FLAG_AUDIBILITY_ENFORCED) { - return AUDIO_STREAM_ENFORCED_AUDIBLE; - } - if ((attr->flags & AUDIO_FLAG_SCO) == AUDIO_FLAG_SCO) { - return AUDIO_STREAM_BLUETOOTH_SCO; - } - if ((attr->flags & AUDIO_FLAG_BEACON) == AUDIO_FLAG_BEACON) { - return AUDIO_STREAM_TTS; - } - - // usage to stream type mapping - switch (attr->usage) { - case AUDIO_USAGE_MEDIA: - case AUDIO_USAGE_GAME: - case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE: - return AUDIO_STREAM_MUSIC; - case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY: - if (isStreamActive(AUDIO_STREAM_ALARM)) { - return AUDIO_STREAM_ALARM; - } - if (isStreamActive(AUDIO_STREAM_RING)) { - return AUDIO_STREAM_RING; - } - if (isInCall()) { - return AUDIO_STREAM_VOICE_CALL; - } - return AUDIO_STREAM_ACCESSIBILITY; - case AUDIO_USAGE_ASSISTANCE_SONIFICATION: - return AUDIO_STREAM_SYSTEM; - case AUDIO_USAGE_VOICE_COMMUNICATION: - return AUDIO_STREAM_VOICE_CALL; - - case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING: - return AUDIO_STREAM_DTMF; - - case AUDIO_USAGE_ALARM: - return AUDIO_STREAM_ALARM; - case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE: - return AUDIO_STREAM_RING; - - case AUDIO_USAGE_NOTIFICATION: - case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST: - case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT: - case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED: - case AUDIO_USAGE_NOTIFICATION_EVENT: - return AUDIO_STREAM_NOTIFICATION; - - case AUDIO_USAGE_UNKNOWN: - default: - return AUDIO_STREAM_MUSIC; - } -} - -bool AudioPolicyManager::isValidAttributes(const audio_attributes_t *paa) { - // has flags that map to a strategy? - if ((paa->flags & (AUDIO_FLAG_AUDIBILITY_ENFORCED | AUDIO_FLAG_SCO | AUDIO_FLAG_BEACON)) != 0) { - return true; - } - - // has known usage? - switch (paa->usage) { - case AUDIO_USAGE_UNKNOWN: - case AUDIO_USAGE_MEDIA: - case AUDIO_USAGE_VOICE_COMMUNICATION: - case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING: - case AUDIO_USAGE_ALARM: - case AUDIO_USAGE_NOTIFICATION: - case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE: - case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST: - case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT: - case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED: - case AUDIO_USAGE_NOTIFICATION_EVENT: - case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY: - case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE: - case AUDIO_USAGE_ASSISTANCE_SONIFICATION: - case AUDIO_USAGE_GAME: - case AUDIO_USAGE_VIRTUAL_SOURCE: - break; - default: - return false; - } - return true; -} - -}; // namespace android diff --git a/services/audiopolicy/AudioPolicyManager.h b/services/audiopolicy/AudioPolicyManager.h deleted file mode 100644 index 81d4f14..0000000 --- a/services/audiopolicy/AudioPolicyManager.h +++ /dev/null @@ -1,951 +0,0 @@ -/* - * Copyright (C) 2009 The Android Open Source Project - * - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - */ - - -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include "AudioPolicyInterface.h" - - -namespace android { - -// ---------------------------------------------------------------------------- - -// Attenuation applied to STRATEGY_SONIFICATION streams when a headset is connected: 6dB -#define SONIFICATION_HEADSET_VOLUME_FACTOR 0.5 -// Min volume for STRATEGY_SONIFICATION streams when limited by music volume: -36dB -#define SONIFICATION_HEADSET_VOLUME_MIN 0.016 -// Time in milliseconds during which we consider that music is still active after a music -// track was stopped - see computeVolume() -#define SONIFICATION_HEADSET_MUSIC_DELAY 5000 -// Time in milliseconds after media stopped playing during which we consider that the -// sonification should be as unobtrusive as during the time media was playing. -#define SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY 5000 -// Time in milliseconds during witch some streams are muted while the audio path -// is switched -#define MUTE_TIME_MS 2000 - -#define NUM_TEST_OUTPUTS 5 - -#define NUM_VOL_CURVE_KNEES 2 - -// Default minimum length allowed for offloading a compressed track -// Can be overridden by the audio.offload.min.duration.secs property -#define OFFLOAD_DEFAULT_MIN_DURATION_SECS 60 - -#define MAX_MIXER_SAMPLING_RATE 48000 -#define MAX_MIXER_CHANNEL_COUNT 8 - -// ---------------------------------------------------------------------------- -// AudioPolicyManager implements audio policy manager behavior common to all platforms. -// ---------------------------------------------------------------------------- - -class AudioPolicyManager: public AudioPolicyInterface -#ifdef AUDIO_POLICY_TEST - , public Thread -#endif //AUDIO_POLICY_TEST -{ - -public: - AudioPolicyManager(AudioPolicyClientInterface *clientInterface); - virtual ~AudioPolicyManager(); - - // AudioPolicyInterface - virtual status_t setDeviceConnectionState(audio_devices_t device, - audio_policy_dev_state_t state, - const char *device_address, - const char *device_name); - virtual audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device, - const char *device_address); - virtual void setPhoneState(audio_mode_t state); - virtual void setForceUse(audio_policy_force_use_t usage, - audio_policy_forced_cfg_t config); - virtual audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage); - virtual void setSystemProperty(const char* property, const char* value); - virtual status_t initCheck(); - virtual audio_io_handle_t getOutput(audio_stream_type_t stream, - uint32_t samplingRate, - audio_format_t format, - audio_channel_mask_t channelMask, - audio_output_flags_t flags, - const audio_offload_info_t *offloadInfo); - virtual status_t getOutputForAttr(const audio_attributes_t *attr, - audio_io_handle_t *output, - audio_session_t session, - audio_stream_type_t *stream, - uint32_t samplingRate, - audio_format_t format, - audio_channel_mask_t channelMask, - audio_output_flags_t flags, - const audio_offload_info_t *offloadInfo); - virtual status_t startOutput(audio_io_handle_t output, - audio_stream_type_t stream, - audio_session_t session); - virtual status_t stopOutput(audio_io_handle_t output, - audio_stream_type_t stream, - audio_session_t session); - virtual void releaseOutput(audio_io_handle_t output, - audio_stream_type_t stream, - audio_session_t session); - virtual status_t getInputForAttr(const audio_attributes_t *attr, - audio_io_handle_t *input, - audio_session_t session, - uint32_t samplingRate, - audio_format_t format, - audio_channel_mask_t channelMask, - audio_input_flags_t flags, - input_type_t *inputType); - - // indicates to the audio policy manager that the input starts being used. - virtual status_t startInput(audio_io_handle_t input, - audio_session_t session); - - // indicates to the audio policy manager that the input stops being used. - virtual status_t stopInput(audio_io_handle_t input, - audio_session_t session); - virtual void releaseInput(audio_io_handle_t input, - audio_session_t session); - virtual void closeAllInputs(); - virtual void initStreamVolume(audio_stream_type_t stream, - int indexMin, - int indexMax); - virtual status_t setStreamVolumeIndex(audio_stream_type_t stream, - int index, - audio_devices_t device); - virtual status_t getStreamVolumeIndex(audio_stream_type_t stream, - int *index, - audio_devices_t device); - - // return the strategy corresponding to a given stream type - virtual uint32_t getStrategyForStream(audio_stream_type_t stream); - // return the strategy corresponding to the given audio attributes - virtual uint32_t getStrategyForAttr(const audio_attributes_t *attr); - - // return the enabled output devices for the given stream type - virtual audio_devices_t getDevicesForStream(audio_stream_type_t stream); - - virtual audio_io_handle_t getOutputForEffect(const effect_descriptor_t *desc = NULL); - virtual status_t registerEffect(const effect_descriptor_t *desc, - audio_io_handle_t io, - uint32_t strategy, - int session, - int id); - virtual status_t unregisterEffect(int id); - virtual status_t setEffectEnabled(int id, bool enabled); - - virtual bool isStreamActive(audio_stream_type_t stream, uint32_t inPastMs = 0) const; - // return whether a stream is playing remotely, override to change the definition of - // local/remote playback, used for instance by notification manager to not make - // media players lose audio focus when not playing locally - // For the base implementation, "remotely" means playing during screen mirroring which - // uses an output for playback with a non-empty, non "0" address. - virtual bool isStreamActiveRemotely(audio_stream_type_t stream, uint32_t inPastMs = 0) const; - virtual bool isSourceActive(audio_source_t source) const; - - virtual status_t dump(int fd); - - virtual bool isOffloadSupported(const audio_offload_info_t& offloadInfo); - - virtual status_t listAudioPorts(audio_port_role_t role, - audio_port_type_t type, - unsigned int *num_ports, - struct audio_port *ports, - unsigned int *generation); - virtual status_t getAudioPort(struct audio_port *port); - virtual status_t createAudioPatch(const struct audio_patch *patch, - audio_patch_handle_t *handle, - uid_t uid); - virtual status_t releaseAudioPatch(audio_patch_handle_t handle, - uid_t uid); - virtual status_t listAudioPatches(unsigned int *num_patches, - struct audio_patch *patches, - unsigned int *generation); - virtual status_t setAudioPortConfig(const struct audio_port_config *config); - virtual void clearAudioPatches(uid_t uid); - - virtual status_t acquireSoundTriggerSession(audio_session_t *session, - audio_io_handle_t *ioHandle, - audio_devices_t *device); - - virtual status_t releaseSoundTriggerSession(audio_session_t session); - - virtual status_t registerPolicyMixes(Vector mixes); - virtual status_t unregisterPolicyMixes(Vector mixes); - -protected: - - enum routing_strategy { - STRATEGY_MEDIA, - STRATEGY_PHONE, - STRATEGY_SONIFICATION, - STRATEGY_SONIFICATION_RESPECTFUL, - STRATEGY_DTMF, - STRATEGY_ENFORCED_AUDIBLE, - STRATEGY_TRANSMITTED_THROUGH_SPEAKER, - STRATEGY_ACCESSIBILITY, - STRATEGY_REROUTING, - NUM_STRATEGIES - }; - - // 4 points to define the volume attenuation curve, each characterized by the volume - // index (from 0 to 100) at which they apply, and the attenuation in dB at that index. - // we use 100 steps to avoid rounding errors when computing the volume in volIndexToAmpl() - - enum { VOLMIN = 0, VOLKNEE1 = 1, VOLKNEE2 = 2, VOLMAX = 3, VOLCNT = 4}; - - class VolumeCurvePoint - { - public: - int mIndex; - float mDBAttenuation; - }; - - // device categories used for volume curve management. - enum device_category { - DEVICE_CATEGORY_HEADSET, - DEVICE_CATEGORY_SPEAKER, - DEVICE_CATEGORY_EARPIECE, - DEVICE_CATEGORY_EXT_MEDIA, - DEVICE_CATEGORY_CNT - }; - - class HwModule; - - class AudioGain: public RefBase - { - public: - AudioGain(int index, bool useInChannelMask); - virtual ~AudioGain() {} - - void dump(int fd, int spaces, int index) const; - - void getDefaultConfig(struct audio_gain_config *config); - status_t checkConfig(const struct audio_gain_config *config); - int mIndex; - struct audio_gain mGain; - bool mUseInChannelMask; - }; - - class AudioPort: public virtual RefBase - { - public: - AudioPort(const String8& name, audio_port_type_t type, - audio_port_role_t role, const sp& module); - virtual ~AudioPort() {} - - audio_port_handle_t getHandle() { return mId; } - - void attach(const sp& module); - bool isAttached() { return mId != 0; } - - virtual void toAudioPort(struct audio_port *port) const; - - void importAudioPort(const sp port); - void clearCapabilities(); - - void loadSamplingRates(char *name); - void loadFormats(char *name); - void loadOutChannels(char *name); - void loadInChannels(char *name); - - audio_gain_mode_t loadGainMode(char *name); - void loadGain(cnode *root, int index); - virtual void loadGains(cnode *root); - - // searches for an exact match - status_t checkExactSamplingRate(uint32_t samplingRate) const; - // searches for a compatible match, and returns the best match via updatedSamplingRate - status_t checkCompatibleSamplingRate(uint32_t samplingRate, - uint32_t *updatedSamplingRate) const; - // searches for an exact match - status_t checkExactChannelMask(audio_channel_mask_t channelMask) const; - // searches for a compatible match, currently implemented for input channel masks only - status_t checkCompatibleChannelMask(audio_channel_mask_t channelMask) const; - status_t checkFormat(audio_format_t format) const; - status_t checkGain(const struct audio_gain_config *gainConfig, int index) const; - - uint32_t pickSamplingRate() const; - audio_channel_mask_t pickChannelMask() const; - audio_format_t pickFormat() const; - - static const audio_format_t sPcmFormatCompareTable[]; - static int compareFormats(audio_format_t format1, audio_format_t format2); - - void dump(int fd, int spaces) const; - - String8 mName; - audio_port_type_t mType; - audio_port_role_t mRole; - bool mUseInChannelMask; - // by convention, "0' in the first entry in mSamplingRates, mChannelMasks or mFormats - // indicates the supported parameters should be read from the output stream - // after it is opened for the first time - Vector mSamplingRates; // supported sampling rates - Vector mChannelMasks; // supported channel masks - Vector mFormats; // supported audio formats - Vector < sp > mGains; // gain controllers - sp mModule; // audio HW module exposing this I/O stream - uint32_t mFlags; // attribute flags (e.g primary output, - // direct output...). - - protected: - //TODO - clarify the role of mId in this case, both an "attached" indicator - // and a unique ID for identifying a port to the (upcoming) selection API, - // and its relationship to the mId in AudioOutputDescriptor and AudioInputDescriptor. - audio_port_handle_t mId; - }; - - class AudioPortConfig: public virtual RefBase - { - public: - AudioPortConfig(); - virtual ~AudioPortConfig() {} - - status_t applyAudioPortConfig(const struct audio_port_config *config, - struct audio_port_config *backupConfig = NULL); - virtual void toAudioPortConfig(struct audio_port_config *dstConfig, - const struct audio_port_config *srcConfig = NULL) const = 0; - virtual sp getAudioPort() const = 0; - uint32_t mSamplingRate; - audio_format_t mFormat; - audio_channel_mask_t mChannelMask; - struct audio_gain_config mGain; - }; - - - class AudioPatch: public RefBase - { - public: - AudioPatch(audio_patch_handle_t handle, - const struct audio_patch *patch, uid_t uid) : - mHandle(handle), mPatch(*patch), mUid(uid), mAfPatchHandle(0) {} - - status_t dump(int fd, int spaces, int index) const; - - audio_patch_handle_t mHandle; - struct audio_patch mPatch; - uid_t mUid; - audio_patch_handle_t mAfPatchHandle; - }; - - class DeviceDescriptor: public AudioPort, public AudioPortConfig - { - public: - DeviceDescriptor(const String8& name, audio_devices_t type); - - virtual ~DeviceDescriptor() {} - - bool equals(const sp& other) const; - - // AudioPortConfig - virtual sp getAudioPort() const { return (AudioPort*) this; } - virtual void toAudioPortConfig(struct audio_port_config *dstConfig, - const struct audio_port_config *srcConfig = NULL) const; - - // AudioPort - virtual void loadGains(cnode *root); - virtual void toAudioPort(struct audio_port *port) const; - - status_t dump(int fd, int spaces, int index) const; - - audio_devices_t mDeviceType; - String8 mAddress; - - static String8 emptyNameStr; - }; - - class DeviceVector : public SortedVector< sp > - { - public: - DeviceVector() : SortedVector(), mDeviceTypes(AUDIO_DEVICE_NONE) {} - - ssize_t add(const sp& item); - ssize_t remove(const sp& item); - ssize_t indexOf(const sp& item) const; - - audio_devices_t types() const { return mDeviceTypes; } - - void loadDevicesFromType(audio_devices_t types); - void loadDevicesFromName(char *name, const DeviceVector& declaredDevices); - - sp getDevice(audio_devices_t type, String8 address) const; - DeviceVector getDevicesFromType(audio_devices_t types) const; - sp getDeviceFromId(audio_port_handle_t id) const; - sp getDeviceFromName(const String8& name) const; - DeviceVector getDevicesFromTypeAddr(audio_devices_t type, String8 address) - const; - - private: - void refreshTypes(); - audio_devices_t mDeviceTypes; - }; - - // the IOProfile class describes the capabilities of an output or input stream. - // It is currently assumed that all combination of listed parameters are supported. - // It is used by the policy manager to determine if an output or input is suitable for - // a given use case, open/close it accordingly and connect/disconnect audio tracks - // to/from it. - class IOProfile : public AudioPort - { - public: - IOProfile(const String8& name, audio_port_role_t role, const sp& module); - virtual ~IOProfile(); - - // This method is used for both output and input. - // If parameter updatedSamplingRate is non-NULL, it is assigned the actual sample rate. - // For input, flags is interpreted as audio_input_flags_t. - // TODO: merge audio_output_flags_t and audio_input_flags_t. - bool isCompatibleProfile(audio_devices_t device, - String8 address, - uint32_t samplingRate, - uint32_t *updatedSamplingRate, - audio_format_t format, - audio_channel_mask_t channelMask, - uint32_t flags) const; - - void dump(int fd); - void log(); - - DeviceVector mSupportedDevices; // supported devices - // (devices this output can be routed to) - }; - - class HwModule : public RefBase - { - public: - HwModule(const char *name); - ~HwModule(); - - status_t loadOutput(cnode *root); - status_t loadInput(cnode *root); - status_t loadDevice(cnode *root); - - status_t addOutputProfile(String8 name, const audio_config_t *config, - audio_devices_t device, String8 address); - status_t removeOutputProfile(String8 name); - status_t addInputProfile(String8 name, const audio_config_t *config, - audio_devices_t device, String8 address); - status_t removeInputProfile(String8 name); - - void dump(int fd); - - const char *const mName; // base name of the audio HW module (primary, a2dp ...) - uint32_t mHalVersion; // audio HAL API version - audio_module_handle_t mHandle; - Vector < sp > mOutputProfiles; // output profiles exposed by this module - Vector < sp > mInputProfiles; // input profiles exposed by this module - DeviceVector mDeclaredDevices; // devices declared in audio_policy.conf - - }; - - // default volume curve - static const VolumeCurvePoint sDefaultVolumeCurve[AudioPolicyManager::VOLCNT]; - // default volume curve for media strategy - static const VolumeCurvePoint sDefaultMediaVolumeCurve[AudioPolicyManager::VOLCNT]; - // volume curve for non-media audio on ext media outputs (HDMI, Line, etc) - static const VolumeCurvePoint sExtMediaSystemVolumeCurve[AudioPolicyManager::VOLCNT]; - // volume curve for media strategy on speakers - static const VolumeCurvePoint sSpeakerMediaVolumeCurve[AudioPolicyManager::VOLCNT]; - static const VolumeCurvePoint sSpeakerMediaVolumeCurveDrc[AudioPolicyManager::VOLCNT]; - // volume curve for sonification strategy on speakers - static const VolumeCurvePoint sSpeakerSonificationVolumeCurve[AudioPolicyManager::VOLCNT]; - static const VolumeCurvePoint sSpeakerSonificationVolumeCurveDrc[AudioPolicyManager::VOLCNT]; - static const VolumeCurvePoint sDefaultSystemVolumeCurve[AudioPolicyManager::VOLCNT]; - static const VolumeCurvePoint sDefaultSystemVolumeCurveDrc[AudioPolicyManager::VOLCNT]; - static const VolumeCurvePoint sHeadsetSystemVolumeCurve[AudioPolicyManager::VOLCNT]; - static const VolumeCurvePoint sDefaultVoiceVolumeCurve[AudioPolicyManager::VOLCNT]; - static const VolumeCurvePoint sSpeakerVoiceVolumeCurve[AudioPolicyManager::VOLCNT]; - static const VolumeCurvePoint sLinearVolumeCurve[AudioPolicyManager::VOLCNT]; - static const VolumeCurvePoint sSilentVolumeCurve[AudioPolicyManager::VOLCNT]; - static const VolumeCurvePoint sFullScaleVolumeCurve[AudioPolicyManager::VOLCNT]; - // default volume curves per stream and device category. See initializeVolumeCurves() - static const VolumeCurvePoint *sVolumeProfiles[AUDIO_STREAM_CNT][DEVICE_CATEGORY_CNT]; - - // descriptor for audio outputs. Used to maintain current configuration of each opened audio output - // and keep track of the usage of this output by each audio stream type. - class AudioOutputDescriptor: public AudioPortConfig - { - public: - AudioOutputDescriptor(const sp& profile); - - status_t dump(int fd); - - audio_devices_t device() const; - void changeRefCount(audio_stream_type_t stream, int delta); - - bool isDuplicated() const { return (mOutput1 != NULL && mOutput2 != NULL); } - audio_devices_t supportedDevices(); - uint32_t latency(); - bool sharesHwModuleWith(const sp outputDesc); - bool isActive(uint32_t inPastMs = 0) const; - bool isStreamActive(audio_stream_type_t stream, - uint32_t inPastMs = 0, - nsecs_t sysTime = 0) const; - bool isStrategyActive(routing_strategy strategy, - uint32_t inPastMs = 0, - nsecs_t sysTime = 0) const; - - virtual void toAudioPortConfig(struct audio_port_config *dstConfig, - const struct audio_port_config *srcConfig = NULL) const; - virtual sp getAudioPort() const { return mProfile; } - void toAudioPort(struct audio_port *port) const; - - audio_port_handle_t mId; - audio_io_handle_t mIoHandle; // output handle - uint32_t mLatency; // - audio_output_flags_t mFlags; // - audio_devices_t mDevice; // current device this output is routed to - AudioMix *mPolicyMix; // non NULL when used by a dynamic policy - audio_patch_handle_t mPatchHandle; - uint32_t mRefCount[AUDIO_STREAM_CNT]; // number of streams of each type using this output - nsecs_t mStopTime[AUDIO_STREAM_CNT]; - sp mOutput1; // used by duplicated outputs: first output - sp mOutput2; // used by duplicated outputs: second output - float mCurVolume[AUDIO_STREAM_CNT]; // current stream volume - int mMuteCount[AUDIO_STREAM_CNT]; // mute request counter - const sp mProfile; // I/O profile this output derives from - bool mStrategyMutedByDevice[NUM_STRATEGIES]; // strategies muted because of incompatible - // device selection. See checkDeviceMuteStrategies() - uint32_t mDirectOpenCount; // number of clients using this output (direct outputs only) - }; - - // descriptor for audio inputs. Used to maintain current configuration of each opened audio input - // and keep track of the usage of this input. - class AudioInputDescriptor: public AudioPortConfig - { - public: - AudioInputDescriptor(const sp& profile); - - status_t dump(int fd); - - audio_port_handle_t mId; - audio_io_handle_t mIoHandle; // input handle - audio_devices_t mDevice; // current device this input is routed to - AudioMix *mPolicyMix; // non NULL when used by a dynamic policy - audio_patch_handle_t mPatchHandle; - uint32_t mRefCount; // number of AudioRecord clients using - // this input - uint32_t mOpenRefCount; - audio_source_t mInputSource; // input source selected by application - //(mediarecorder.h) - const sp mProfile; // I/O profile this output derives from - SortedVector mSessions; // audio sessions attached to this input - bool mIsSoundTrigger; // used by a soundtrigger capture - - virtual void toAudioPortConfig(struct audio_port_config *dstConfig, - const struct audio_port_config *srcConfig = NULL) const; - virtual sp getAudioPort() const { return mProfile; } - void toAudioPort(struct audio_port *port) const; - }; - - // stream descriptor used for volume control - class StreamDescriptor - { - public: - StreamDescriptor(); - - int getVolumeIndex(audio_devices_t device); - void dump(int fd); - - int mIndexMin; // min volume index - int mIndexMax; // max volume index - KeyedVector mIndexCur; // current volume index per device - bool mCanBeMuted; // true is the stream can be muted - - const VolumeCurvePoint *mVolumeCurve[DEVICE_CATEGORY_CNT]; - }; - - // stream descriptor used for volume control - class EffectDescriptor : public RefBase - { - public: - - status_t dump(int fd); - - int mIo; // io the effect is attached to - routing_strategy mStrategy; // routing strategy the effect is associated to - int mSession; // audio session the effect is on - effect_descriptor_t mDesc; // effect descriptor - bool mEnabled; // enabled state: CPU load being used or not - }; - - void addOutput(audio_io_handle_t output, sp outputDesc); - void addInput(audio_io_handle_t input, sp inputDesc); - - // return the strategy corresponding to a given stream type - static routing_strategy getStrategy(audio_stream_type_t stream); - - // return appropriate device for streams handled by the specified strategy according to current - // phone state, connected devices... - // if fromCache is true, the device is returned from mDeviceForStrategy[], - // otherwise it is determine by current state - // (device connected,phone state, force use, a2dp output...) - // This allows to: - // 1 speed up process when the state is stable (when starting or stopping an output) - // 2 access to either current device selection (fromCache == true) or - // "future" device selection (fromCache == false) when called from a context - // where conditions are changing (setDeviceConnectionState(), setPhoneState()...) AND - // before updateDevicesAndOutputs() is called. - virtual audio_devices_t getDeviceForStrategy(routing_strategy strategy, - bool fromCache); - - // change the route of the specified output. Returns the number of ms we have slept to - // allow new routing to take effect in certain cases. - virtual uint32_t setOutputDevice(audio_io_handle_t output, - audio_devices_t device, - bool force = false, - int delayMs = 0, - audio_patch_handle_t *patchHandle = NULL, - const char* address = NULL); - status_t resetOutputDevice(audio_io_handle_t output, - int delayMs = 0, - audio_patch_handle_t *patchHandle = NULL); - status_t setInputDevice(audio_io_handle_t input, - audio_devices_t device, - bool force = false, - audio_patch_handle_t *patchHandle = NULL); - status_t resetInputDevice(audio_io_handle_t input, - audio_patch_handle_t *patchHandle = NULL); - - // select input device corresponding to requested audio source - virtual audio_devices_t getDeviceForInputSource(audio_source_t inputSource); - - // return io handle of active input or 0 if no input is active - // Only considers inputs from physical devices (e.g. main mic, headset mic) when - // ignoreVirtualInputs is true. - audio_io_handle_t getActiveInput(bool ignoreVirtualInputs = true); - - uint32_t activeInputsCount() const; - - // initialize volume curves for each strategy and device category - void initializeVolumeCurves(); - - // compute the actual volume for a given stream according to the requested index and a particular - // device - virtual float computeVolume(audio_stream_type_t stream, int index, - audio_io_handle_t output, audio_devices_t device); - - // check that volume change is permitted, compute and send new volume to audio hardware - virtual status_t checkAndSetVolume(audio_stream_type_t stream, int index, - audio_io_handle_t output, - audio_devices_t device, - int delayMs = 0, bool force = false); - - // apply all stream volumes to the specified output and device - void applyStreamVolumes(audio_io_handle_t output, audio_devices_t device, int delayMs = 0, bool force = false); - - // Mute or unmute all streams handled by the specified strategy on the specified output - void setStrategyMute(routing_strategy strategy, - bool on, - audio_io_handle_t output, - int delayMs = 0, - audio_devices_t device = (audio_devices_t)0); - - // Mute or unmute the stream on the specified output - void setStreamMute(audio_stream_type_t stream, - bool on, - audio_io_handle_t output, - int delayMs = 0, - audio_devices_t device = (audio_devices_t)0); - - // handle special cases for sonification strategy while in call: mute streams or replace by - // a special tone in the device used for communication - void handleIncallSonification(audio_stream_type_t stream, bool starting, bool stateChange); - - // true if device is in a telephony or VoIP call - virtual bool isInCall(); - - // true if given state represents a device in a telephony or VoIP call - virtual bool isStateInCall(int state); - - // when a device is connected, checks if an open output can be routed - // to this device. If none is open, tries to open one of the available outputs. - // Returns an output suitable to this device or 0. - // when a device is disconnected, checks if an output is not used any more and - // returns its handle if any. - // transfers the audio tracks and effects from one output thread to another accordingly. - status_t checkOutputsForDevice(const sp devDesc, - audio_policy_dev_state_t state, - SortedVector& outputs, - const String8 address); - - status_t checkInputsForDevice(audio_devices_t device, - audio_policy_dev_state_t state, - SortedVector& inputs, - const String8 address); - - // close an output and its companion duplicating output. - void closeOutput(audio_io_handle_t output); - - // close an input. - void closeInput(audio_io_handle_t input); - - // checks and if necessary changes outputs used for all strategies. - // must be called every time a condition that affects the output choice for a given strategy - // changes: connected device, phone state, force use... - // Must be called before updateDevicesAndOutputs() - void checkOutputForStrategy(routing_strategy strategy); - - // Same as checkOutputForStrategy() but for a all strategies in order of priority - void checkOutputForAllStrategies(); - - // manages A2DP output suspend/restore according to phone state and BT SCO usage - void checkA2dpSuspend(); - - // returns the A2DP output handle if it is open or 0 otherwise - audio_io_handle_t getA2dpOutput(); - - // selects the most appropriate device on output for current state - // must be called every time a condition that affects the device choice for a given output is - // changed: connected device, phone state, force use, output start, output stop.. - // see getDeviceForStrategy() for the use of fromCache parameter - audio_devices_t getNewOutputDevice(audio_io_handle_t output, bool fromCache); - - // updates cache of device used by all strategies (mDeviceForStrategy[]) - // must be called every time a condition that affects the device choice for a given strategy is - // changed: connected device, phone state, force use... - // cached values are used by getDeviceForStrategy() if parameter fromCache is true. - // Must be called after checkOutputForAllStrategies() - void updateDevicesAndOutputs(); - - // selects the most appropriate device on input for current state - audio_devices_t getNewInputDevice(audio_io_handle_t input); - - virtual uint32_t getMaxEffectsCpuLoad(); - virtual uint32_t getMaxEffectsMemory(); -#ifdef AUDIO_POLICY_TEST - virtual bool threadLoop(); - void exit(); - int testOutputIndex(audio_io_handle_t output); -#endif //AUDIO_POLICY_TEST - - status_t setEffectEnabled(const sp& effectDesc, bool enabled); - - // returns the category the device belongs to with regard to volume curve management - static device_category getDeviceCategory(audio_devices_t device); - - // extract one device relevant for volume control from multiple device selection - static audio_devices_t getDeviceForVolume(audio_devices_t device); - - SortedVector getOutputsForDevice(audio_devices_t device, - DefaultKeyedVector > openOutputs); - bool vectorsEqual(SortedVector& outputs1, - SortedVector& outputs2); - - // mute/unmute strategies using an incompatible device combination - // if muting, wait for the audio in pcm buffer to be drained before proceeding - // if unmuting, unmute only after the specified delay - // Returns the number of ms waited - virtual uint32_t checkDeviceMuteStrategies(sp outputDesc, - audio_devices_t prevDevice, - uint32_t delayMs); - - audio_io_handle_t selectOutput(const SortedVector& outputs, - audio_output_flags_t flags, - audio_format_t format); - // samplingRate parameter is an in/out and so may be modified - sp getInputProfile(audio_devices_t device, - String8 address, - uint32_t& samplingRate, - audio_format_t format, - audio_channel_mask_t channelMask, - audio_input_flags_t flags); - sp getProfileForDirectOutput(audio_devices_t device, - uint32_t samplingRate, - audio_format_t format, - audio_channel_mask_t channelMask, - audio_output_flags_t flags); - - audio_io_handle_t selectOutputForEffects(const SortedVector& outputs); - - bool isNonOffloadableEffectEnabled(); - - virtual status_t addAudioPatch(audio_patch_handle_t handle, - const sp& patch); - virtual status_t removeAudioPatch(audio_patch_handle_t handle); - - sp getOutputFromId(audio_port_handle_t id) const; - sp getInputFromId(audio_port_handle_t id) const; - sp getModuleForDevice(audio_devices_t device) const; - sp getModuleFromName(const char *name) const; - audio_devices_t availablePrimaryOutputDevices(); - audio_devices_t availablePrimaryInputDevices(); - - void updateCallRouting(audio_devices_t rxDevice, int delayMs = 0); - - // - // Audio policy configuration file parsing (audio_policy.conf) - // - static uint32_t stringToEnum(const struct StringToEnum *table, - size_t size, - const char *name); - static const char *enumToString(const struct StringToEnum *table, - size_t size, - uint32_t value); - static bool stringToBool(const char *value); - static uint32_t parseOutputFlagNames(char *name); - static uint32_t parseInputFlagNames(char *name); - static audio_devices_t parseDeviceNames(char *name); - void loadHwModule(cnode *root); - void loadHwModules(cnode *root); - void loadGlobalConfig(cnode *root, const sp& module); - status_t loadAudioPolicyConfig(const char *path); - void defaultAudioPolicyConfig(void); - - - uid_t mUidCached; - AudioPolicyClientInterface *mpClientInterface; // audio policy client interface - audio_io_handle_t mPrimaryOutput; // primary output handle - // list of descriptors for outputs currently opened - DefaultKeyedVector > mOutputs; - // copy of mOutputs before setDeviceConnectionState() opens new outputs - // reset to mOutputs when updateDevicesAndOutputs() is called. - DefaultKeyedVector > mPreviousOutputs; - DefaultKeyedVector > mInputs; // list of input descriptors - DeviceVector mAvailableOutputDevices; // all available output devices - DeviceVector mAvailableInputDevices; // all available input devices - int mPhoneState; // current phone state - audio_policy_forced_cfg_t mForceUse[AUDIO_POLICY_FORCE_USE_CNT]; // current forced use configuration - - StreamDescriptor mStreams[AUDIO_STREAM_CNT]; // stream descriptors for volume control - bool mLimitRingtoneVolume; // limit ringtone volume to music volume if headset connected - audio_devices_t mDeviceForStrategy[NUM_STRATEGIES]; - float mLastVoiceVolume; // last voice volume value sent to audio HAL - - // Maximum CPU load allocated to audio effects in 0.1 MIPS (ARMv5TE, 0 WS memory) units - static const uint32_t MAX_EFFECTS_CPU_LOAD = 1000; - // Maximum memory allocated to audio effects in KB - static const uint32_t MAX_EFFECTS_MEMORY = 512; - uint32_t mTotalEffectsCpuLoad; // current CPU load used by effects - uint32_t mTotalEffectsMemory; // current memory used by effects - KeyedVector > mEffects; // list of registered audio effects - bool mA2dpSuspended; // true if A2DP output is suspended - sp mDefaultOutputDevice; // output device selected by default at boot time - bool mSpeakerDrcEnabled;// true on devices that use DRC on the DEVICE_CATEGORY_SPEAKER path - // to boost soft sounds, used to adjust volume curves accordingly - - Vector < sp > mHwModules; - static volatile int32_t mNextUniqueId; - volatile int32_t mAudioPortGeneration; - - DefaultKeyedVector > mAudioPatches; - - DefaultKeyedVector mSoundTriggerSessions; - - sp mCallTxPatch; - sp mCallRxPatch; - - // for supporting "beacon" streams, i.e. streams that only play on speaker, and never - // when something other than STREAM_TTS (a.k.a. "Transmitted Through Speaker") is playing - enum { - STARTING_OUTPUT, - STARTING_BEACON, - STOPPING_OUTPUT, - STOPPING_BEACON - }; - uint32_t mBeaconMuteRefCount; // ref count for stream that would mute beacon - uint32_t mBeaconPlayingRefCount;// ref count for the playing beacon streams - bool mBeaconMuted; // has STREAM_TTS been muted - - // custom mix entry in mPolicyMixes - class AudioPolicyMix : public RefBase { - public: - AudioPolicyMix() {} - - AudioMix mMix; // Audio policy mix descriptor - sp mOutput; // Corresponding output stream - }; - DefaultKeyedVector > mPolicyMixes; // list of registered mixes - - -#ifdef AUDIO_POLICY_TEST - Mutex mLock; - Condition mWaitWorkCV; - - int mCurOutput; - bool mDirectOutput; - audio_io_handle_t mTestOutputs[NUM_TEST_OUTPUTS]; - int mTestInput; - uint32_t mTestDevice; - uint32_t mTestSamplingRate; - uint32_t mTestFormat; - uint32_t mTestChannels; - uint32_t mTestLatencyMs; -#endif //AUDIO_POLICY_TEST - static float volIndexToAmpl(audio_devices_t device, const StreamDescriptor& streamDesc, - int indexInUi); - static bool isVirtualInputDevice(audio_devices_t device); - static uint32_t nextUniqueId(); - uint32_t nextAudioPortGeneration(); -private: - // updates device caching and output for streams that can influence the - // routing of notifications - void handleNotificationRoutingForStream(audio_stream_type_t stream); - static bool deviceDistinguishesOnAddress(audio_devices_t device); - // find the outputs on a given output descriptor that have the given address. - // to be called on an AudioOutputDescriptor whose supported devices (as defined - // in mProfile->mSupportedDevices) matches the device whose address is to be matched. - // see deviceDistinguishesOnAddress(audio_devices_t) for whether the device type is one - // where addresses are used to distinguish between one connected device and another. - void findIoHandlesByAddress(sp desc /*in*/, - const audio_devices_t device /*in*/, - const String8 address /*in*/, - SortedVector& outputs /*out*/); - uint32_t curAudioPortGeneration() const { return mAudioPortGeneration; } - // internal method to return the output handle for the given device and format - audio_io_handle_t getOutputForDevice( - audio_devices_t device, - audio_session_t session, - audio_stream_type_t stream, - uint32_t samplingRate, - audio_format_t format, - audio_channel_mask_t channelMask, - audio_output_flags_t flags, - const audio_offload_info_t *offloadInfo); - // internal function to derive a stream type value from audio attributes - audio_stream_type_t streamTypefromAttributesInt(const audio_attributes_t *attr); - // return true if any output is playing anything besides the stream to ignore - bool isAnyOutputActive(audio_stream_type_t streamToIgnore); - // event is one of STARTING_OUTPUT, STARTING_BEACON, STOPPING_OUTPUT, STOPPING_BEACON - // returns 0 if no mute/unmute event happened, the largest latency of the device where - // the mute/unmute happened - uint32_t handleEventForBeacon(int event); - uint32_t setBeaconMute(bool mute); - bool isValidAttributes(const audio_attributes_t *paa); - - // select input device corresponding to requested audio source and return associated policy - // mix if any. Calls getDeviceForInputSource(). - audio_devices_t getDeviceAndMixForInputSource(audio_source_t inputSource, - AudioMix **policyMix = NULL); - - // Called by setDeviceConnectionState(). - status_t setDeviceConnectionStateInt(audio_devices_t device, - audio_policy_dev_state_t state, - const char *device_address, - const char *device_name); - sp getDeviceDescriptor(const audio_devices_t device, - const char *device_address, - const char *device_name); -}; - -}; diff --git a/services/audiopolicy/AudioPolicyService.cpp b/services/audiopolicy/AudioPolicyService.cpp deleted file mode 100644 index eb9116d..0000000 --- a/services/audiopolicy/AudioPolicyService.cpp +++ /dev/null @@ -1,1068 +0,0 @@ -/* - * Copyright (C) 2009 The Android Open Source Project - * - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - */ - -#define LOG_TAG "AudioPolicyService" -//#define LOG_NDEBUG 0 - -#include "Configuration.h" -#undef __STRICT_ANSI__ -#define __STDINT_LIMITS -#define __STDC_LIMIT_MACROS -#include - -#include -#include -#include -#include -#include -#include -#include -#include "AudioPolicyService.h" -#include "ServiceUtilities.h" -#include -#include -#include -#include - -#include -#include -#include -#include - -namespace android { - -static const char kDeadlockedString[] = "AudioPolicyService may be deadlocked\n"; -static const char kCmdDeadlockedString[] = "AudioPolicyService command thread may be deadlocked\n"; - -static const int kDumpLockRetries = 50; -static const int kDumpLockSleepUs = 20000; - -static const nsecs_t kAudioCommandTimeoutNs = seconds(3); // 3 seconds - -namespace { - extern struct audio_policy_service_ops aps_ops; -}; - -// ---------------------------------------------------------------------------- - -AudioPolicyService::AudioPolicyService() - : BnAudioPolicyService(), mpAudioPolicyDev(NULL), mpAudioPolicy(NULL), - mAudioPolicyManager(NULL), mAudioPolicyClient(NULL), mPhoneState(AUDIO_MODE_INVALID) -{ -} - -void AudioPolicyService::onFirstRef() -{ - char value[PROPERTY_VALUE_MAX]; - const struct hw_module_t *module; - int forced_val; - int rc; - - { - Mutex::Autolock _l(mLock); - - // start tone playback thread - mTonePlaybackThread = new AudioCommandThread(String8("ApmTone"), this); - // start audio commands thread - mAudioCommandThread = new AudioCommandThread(String8("ApmAudio"), this); - // start output activity command thread - mOutputCommandThread = new AudioCommandThread(String8("ApmOutput"), this); - -#ifdef USE_LEGACY_AUDIO_POLICY - ALOGI("AudioPolicyService CSTOR in legacy mode"); - - /* instantiate the audio policy manager */ - rc = hw_get_module(AUDIO_POLICY_HARDWARE_MODULE_ID, &module); - if (rc) { - return; - } - rc = audio_policy_dev_open(module, &mpAudioPolicyDev); - ALOGE_IF(rc, "couldn't open audio policy device (%s)", strerror(-rc)); - if (rc) { - return; - } - - rc = mpAudioPolicyDev->create_audio_policy(mpAudioPolicyDev, &aps_ops, this, - &mpAudioPolicy); - ALOGE_IF(rc, "couldn't create audio policy (%s)", strerror(-rc)); - if (rc) { - return; - } - - rc = mpAudioPolicy->init_check(mpAudioPolicy); - ALOGE_IF(rc, "couldn't init_check the audio policy (%s)", strerror(-rc)); - if (rc) { - return; - } - ALOGI("Loaded audio policy from %s (%s)", module->name, module->id); -#else - ALOGI("AudioPolicyService CSTOR in new mode"); - - mAudioPolicyClient = new AudioPolicyClient(this); - mAudioPolicyManager = createAudioPolicyManager(mAudioPolicyClient); -#endif - } - // load audio processing modules - spaudioPolicyEffects = new AudioPolicyEffects(); - { - Mutex::Autolock _l(mLock); - mAudioPolicyEffects = audioPolicyEffects; - } -} - -AudioPolicyService::~AudioPolicyService() -{ - mTonePlaybackThread->exit(); - mAudioCommandThread->exit(); - mOutputCommandThread->exit(); - -#ifdef USE_LEGACY_AUDIO_POLICY - if (mpAudioPolicy != NULL && mpAudioPolicyDev != NULL) { - mpAudioPolicyDev->destroy_audio_policy(mpAudioPolicyDev, mpAudioPolicy); - } - if (mpAudioPolicyDev != NULL) { - audio_policy_dev_close(mpAudioPolicyDev); - } -#else - destroyAudioPolicyManager(mAudioPolicyManager); - delete mAudioPolicyClient; -#endif - - mNotificationClients.clear(); - mAudioPolicyEffects.clear(); -} - -// A notification client is always registered by AudioSystem when the client process -// connects to AudioPolicyService. -void AudioPolicyService::registerClient(const sp& client) -{ - - Mutex::Autolock _l(mNotificationClientsLock); - - uid_t uid = IPCThreadState::self()->getCallingUid(); - if (mNotificationClients.indexOfKey(uid) < 0) { - sp notificationClient = new NotificationClient(this, - client, - uid); - ALOGV("registerClient() client %p, uid %d", client.get(), uid); - - mNotificationClients.add(uid, notificationClient); - - sp binder = IInterface::asBinder(client); - binder->linkToDeath(notificationClient); - } -} - -// removeNotificationClient() is called when the client process dies. -void AudioPolicyService::removeNotificationClient(uid_t uid) -{ - { - Mutex::Autolock _l(mNotificationClientsLock); - mNotificationClients.removeItem(uid); - } -#ifndef USE_LEGACY_AUDIO_POLICY - { - Mutex::Autolock _l(mLock); - if (mAudioPolicyManager) { - mAudioPolicyManager->clearAudioPatches(uid); - } - } -#endif -} - -void AudioPolicyService::onAudioPortListUpdate() -{ - mOutputCommandThread->updateAudioPortListCommand(); -} - -void AudioPolicyService::doOnAudioPortListUpdate() -{ - Mutex::Autolock _l(mNotificationClientsLock); - for (size_t i = 0; i < mNotificationClients.size(); i++) { - mNotificationClients.valueAt(i)->onAudioPortListUpdate(); - } -} - -void AudioPolicyService::onAudioPatchListUpdate() -{ - mOutputCommandThread->updateAudioPatchListCommand(); -} - -status_t AudioPolicyService::clientCreateAudioPatch(const struct audio_patch *patch, - audio_patch_handle_t *handle, - int delayMs) -{ - return mAudioCommandThread->createAudioPatchCommand(patch, handle, delayMs); -} - -status_t AudioPolicyService::clientReleaseAudioPatch(audio_patch_handle_t handle, - int delayMs) -{ - return mAudioCommandThread->releaseAudioPatchCommand(handle, delayMs); -} - -void AudioPolicyService::doOnAudioPatchListUpdate() -{ - Mutex::Autolock _l(mNotificationClientsLock); - for (size_t i = 0; i < mNotificationClients.size(); i++) { - mNotificationClients.valueAt(i)->onAudioPatchListUpdate(); - } -} - -status_t AudioPolicyService::clientSetAudioPortConfig(const struct audio_port_config *config, - int delayMs) -{ - return mAudioCommandThread->setAudioPortConfigCommand(config, delayMs); -} - -AudioPolicyService::NotificationClient::NotificationClient(const sp& service, - const sp& client, - uid_t uid) - : mService(service), mUid(uid), mAudioPolicyServiceClient(client) -{ -} - -AudioPolicyService::NotificationClient::~NotificationClient() -{ -} - -void AudioPolicyService::NotificationClient::binderDied(const wp& who __unused) -{ - sp keep(this); - sp service = mService.promote(); - if (service != 0) { - service->removeNotificationClient(mUid); - } -} - -void AudioPolicyService::NotificationClient::onAudioPortListUpdate() -{ - if (mAudioPolicyServiceClient != 0) { - mAudioPolicyServiceClient->onAudioPortListUpdate(); - } -} - -void AudioPolicyService::NotificationClient::onAudioPatchListUpdate() -{ - if (mAudioPolicyServiceClient != 0) { - mAudioPolicyServiceClient->onAudioPatchListUpdate(); - } -} - -void AudioPolicyService::binderDied(const wp& who) { - ALOGW("binderDied() %p, calling pid %d", who.unsafe_get(), - IPCThreadState::self()->getCallingPid()); -} - -static bool tryLock(Mutex& mutex) -{ - bool locked = false; - for (int i = 0; i < kDumpLockRetries; ++i) { - if (mutex.tryLock() == NO_ERROR) { - locked = true; - break; - } - usleep(kDumpLockSleepUs); - } - return locked; -} - -status_t AudioPolicyService::dumpInternals(int fd) -{ - const size_t SIZE = 256; - char buffer[SIZE]; - String8 result; - -#ifdef USE_LEGACY_AUDIO_POLICY - snprintf(buffer, SIZE, "PolicyManager Interface: %p\n", mpAudioPolicy); -#else - snprintf(buffer, SIZE, "AudioPolicyManager: %p\n", mAudioPolicyManager); -#endif - result.append(buffer); - snprintf(buffer, SIZE, "Command Thread: %p\n", mAudioCommandThread.get()); - result.append(buffer); - snprintf(buffer, SIZE, "Tones Thread: %p\n", mTonePlaybackThread.get()); - result.append(buffer); - - write(fd, result.string(), result.size()); - return NO_ERROR; -} - -status_t AudioPolicyService::dump(int fd, const Vector& args __unused) -{ - if (!dumpAllowed()) { - dumpPermissionDenial(fd); - } else { - bool locked = tryLock(mLock); - if (!locked) { - String8 result(kDeadlockedString); - write(fd, result.string(), result.size()); - } - - dumpInternals(fd); - if (mAudioCommandThread != 0) { - mAudioCommandThread->dump(fd); - } - if (mTonePlaybackThread != 0) { - mTonePlaybackThread->dump(fd); - } - -#ifdef USE_LEGACY_AUDIO_POLICY - if (mpAudioPolicy) { - mpAudioPolicy->dump(mpAudioPolicy, fd); - } -#else - if (mAudioPolicyManager) { - mAudioPolicyManager->dump(fd); - } -#endif - - if (locked) mLock.unlock(); - } - return NO_ERROR; -} - -status_t AudioPolicyService::dumpPermissionDenial(int fd) -{ - const size_t SIZE = 256; - char buffer[SIZE]; - String8 result; - snprintf(buffer, SIZE, "Permission Denial: " - "can't dump AudioPolicyService from pid=%d, uid=%d\n", - IPCThreadState::self()->getCallingPid(), - IPCThreadState::self()->getCallingUid()); - result.append(buffer); - write(fd, result.string(), result.size()); - return NO_ERROR; -} - -status_t AudioPolicyService::onTransact( - uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) -{ - return BnAudioPolicyService::onTransact(code, data, reply, flags); -} - - -// ----------- AudioPolicyService::AudioCommandThread implementation ---------- - -AudioPolicyService::AudioCommandThread::AudioCommandThread(String8 name, - const wp& service) - : Thread(false), mName(name), mService(service) -{ - mpToneGenerator = NULL; -} - - -AudioPolicyService::AudioCommandThread::~AudioCommandThread() -{ - if (!mAudioCommands.isEmpty()) { - release_wake_lock(mName.string()); - } - mAudioCommands.clear(); - delete mpToneGenerator; -} - -void AudioPolicyService::AudioCommandThread::onFirstRef() -{ - run(mName.string(), ANDROID_PRIORITY_AUDIO); -} - -bool AudioPolicyService::AudioCommandThread::threadLoop() -{ - nsecs_t waitTime = INT64_MAX; - - mLock.lock(); - while (!exitPending()) - { - sp svc; - while (!mAudioCommands.isEmpty() && !exitPending()) { - nsecs_t curTime = systemTime(); - // commands are sorted by increasing time stamp: execute them from index 0 and up - if (mAudioCommands[0]->mTime <= curTime) { - sp command = mAudioCommands[0]; - mAudioCommands.removeAt(0); - mLastCommand = command; - - switch (command->mCommand) { - case START_TONE: { - mLock.unlock(); - ToneData *data = (ToneData *)command->mParam.get(); - ALOGV("AudioCommandThread() processing start tone %d on stream %d", - data->mType, data->mStream); - delete mpToneGenerator; - mpToneGenerator = new ToneGenerator(data->mStream, 1.0); - mpToneGenerator->startTone(data->mType); - mLock.lock(); - }break; - case STOP_TONE: { - mLock.unlock(); - ALOGV("AudioCommandThread() processing stop tone"); - if (mpToneGenerator != NULL) { - mpToneGenerator->stopTone(); - delete mpToneGenerator; - mpToneGenerator = NULL; - } - mLock.lock(); - }break; - case SET_VOLUME: { - VolumeData *data = (VolumeData *)command->mParam.get(); - ALOGV("AudioCommandThread() processing set volume stream %d, \ - volume %f, output %d", data->mStream, data->mVolume, data->mIO); - command->mStatus = AudioSystem::setStreamVolume(data->mStream, - data->mVolume, - data->mIO); - }break; - case SET_PARAMETERS: { - ParametersData *data = (ParametersData *)command->mParam.get(); - ALOGV("AudioCommandThread() processing set parameters string %s, io %d", - data->mKeyValuePairs.string(), data->mIO); - command->mStatus = AudioSystem::setParameters(data->mIO, data->mKeyValuePairs); - }break; - case SET_VOICE_VOLUME: { - VoiceVolumeData *data = (VoiceVolumeData *)command->mParam.get(); - ALOGV("AudioCommandThread() processing set voice volume volume %f", - data->mVolume); - command->mStatus = AudioSystem::setVoiceVolume(data->mVolume); - }break; - case STOP_OUTPUT: { - StopOutputData *data = (StopOutputData *)command->mParam.get(); - ALOGV("AudioCommandThread() processing stop output %d", - data->mIO); - svc = mService.promote(); - if (svc == 0) { - break; - } - mLock.unlock(); - svc->doStopOutput(data->mIO, data->mStream, data->mSession); - mLock.lock(); - }break; - case RELEASE_OUTPUT: { - ReleaseOutputData *data = (ReleaseOutputData *)command->mParam.get(); - ALOGV("AudioCommandThread() processing release output %d", - data->mIO); - svc = mService.promote(); - if (svc == 0) { - break; - } - mLock.unlock(); - svc->doReleaseOutput(data->mIO, data->mStream, data->mSession); - mLock.lock(); - }break; - case CREATE_AUDIO_PATCH: { - CreateAudioPatchData *data = (CreateAudioPatchData *)command->mParam.get(); - ALOGV("AudioCommandThread() processing create audio patch"); - sp af = AudioSystem::get_audio_flinger(); - if (af == 0) { - command->mStatus = PERMISSION_DENIED; - } else { - command->mStatus = af->createAudioPatch(&data->mPatch, &data->mHandle); - } - } break; - case RELEASE_AUDIO_PATCH: { - ReleaseAudioPatchData *data = (ReleaseAudioPatchData *)command->mParam.get(); - ALOGV("AudioCommandThread() processing release audio patch"); - sp af = AudioSystem::get_audio_flinger(); - if (af == 0) { - command->mStatus = PERMISSION_DENIED; - } else { - command->mStatus = af->releaseAudioPatch(data->mHandle); - } - } break; - case UPDATE_AUDIOPORT_LIST: { - ALOGV("AudioCommandThread() processing update audio port list"); - svc = mService.promote(); - if (svc == 0) { - break; - } - mLock.unlock(); - svc->doOnAudioPortListUpdate(); - mLock.lock(); - }break; - case UPDATE_AUDIOPATCH_LIST: { - ALOGV("AudioCommandThread() processing update audio patch list"); - svc = mService.promote(); - if (svc == 0) { - break; - } - mLock.unlock(); - svc->doOnAudioPatchListUpdate(); - mLock.lock(); - }break; - case SET_AUDIOPORT_CONFIG: { - SetAudioPortConfigData *data = (SetAudioPortConfigData *)command->mParam.get(); - ALOGV("AudioCommandThread() processing set port config"); - sp af = AudioSystem::get_audio_flinger(); - if (af == 0) { - command->mStatus = PERMISSION_DENIED; - } else { - command->mStatus = af->setAudioPortConfig(&data->mConfig); - } - } break; - default: - ALOGW("AudioCommandThread() unknown command %d", command->mCommand); - } - { - Mutex::Autolock _l(command->mLock); - if (command->mWaitStatus) { - command->mWaitStatus = false; - command->mCond.signal(); - } - } - waitTime = INT64_MAX; - } else { - waitTime = mAudioCommands[0]->mTime - curTime; - break; - } - } - // release mLock before releasing strong reference on the service as - // AudioPolicyService destructor calls AudioCommandThread::exit() which acquires mLock. - mLock.unlock(); - svc.clear(); - mLock.lock(); - if (!exitPending() && mAudioCommands.isEmpty()) { - // release delayed commands wake lock - release_wake_lock(mName.string()); - ALOGV("AudioCommandThread() going to sleep"); - mWaitWorkCV.waitRelative(mLock, waitTime); - ALOGV("AudioCommandThread() waking up"); - } - } - // release delayed commands wake lock before quitting - if (!mAudioCommands.isEmpty()) { - release_wake_lock(mName.string()); - } - mLock.unlock(); - return false; -} - -status_t AudioPolicyService::AudioCommandThread::dump(int fd) -{ - const size_t SIZE = 256; - char buffer[SIZE]; - String8 result; - - snprintf(buffer, SIZE, "AudioCommandThread %p Dump\n", this); - result.append(buffer); - write(fd, result.string(), result.size()); - - bool locked = tryLock(mLock); - if (!locked) { - String8 result2(kCmdDeadlockedString); - write(fd, result2.string(), result2.size()); - } - - snprintf(buffer, SIZE, "- Commands:\n"); - result = String8(buffer); - result.append(" Command Time Wait pParam\n"); - for (size_t i = 0; i < mAudioCommands.size(); i++) { - mAudioCommands[i]->dump(buffer, SIZE); - result.append(buffer); - } - result.append(" Last Command\n"); - if (mLastCommand != 0) { - mLastCommand->dump(buffer, SIZE); - result.append(buffer); - } else { - result.append(" none\n"); - } - - write(fd, result.string(), result.size()); - - if (locked) mLock.unlock(); - - return NO_ERROR; -} - -void AudioPolicyService::AudioCommandThread::startToneCommand(ToneGenerator::tone_type type, - audio_stream_type_t stream) -{ - sp command = new AudioCommand(); - command->mCommand = START_TONE; - sp data = new ToneData(); - data->mType = type; - data->mStream = stream; - command->mParam = data; - ALOGV("AudioCommandThread() adding tone start type %d, stream %d", type, stream); - sendCommand(command); -} - -void AudioPolicyService::AudioCommandThread::stopToneCommand() -{ - sp command = new AudioCommand(); - command->mCommand = STOP_TONE; - ALOGV("AudioCommandThread() adding tone stop"); - sendCommand(command); -} - -status_t AudioPolicyService::AudioCommandThread::volumeCommand(audio_stream_type_t stream, - float volume, - audio_io_handle_t output, - int delayMs) -{ - sp command = new AudioCommand(); - command->mCommand = SET_VOLUME; - sp data = new VolumeData(); - data->mStream = stream; - data->mVolume = volume; - data->mIO = output; - command->mParam = data; - command->mWaitStatus = true; - ALOGV("AudioCommandThread() adding set volume stream %d, volume %f, output %d", - stream, volume, output); - return sendCommand(command, delayMs); -} - -status_t AudioPolicyService::AudioCommandThread::parametersCommand(audio_io_handle_t ioHandle, - const char *keyValuePairs, - int delayMs) -{ - sp command = new AudioCommand(); - command->mCommand = SET_PARAMETERS; - sp data = new ParametersData(); - data->mIO = ioHandle; - data->mKeyValuePairs = String8(keyValuePairs); - command->mParam = data; - command->mWaitStatus = true; - ALOGV("AudioCommandThread() adding set parameter string %s, io %d ,delay %d", - keyValuePairs, ioHandle, delayMs); - return sendCommand(command, delayMs); -} - -status_t AudioPolicyService::AudioCommandThread::voiceVolumeCommand(float volume, int delayMs) -{ - sp command = new AudioCommand(); - command->mCommand = SET_VOICE_VOLUME; - sp data = new VoiceVolumeData(); - data->mVolume = volume; - command->mParam = data; - command->mWaitStatus = true; - ALOGV("AudioCommandThread() adding set voice volume volume %f", volume); - return sendCommand(command, delayMs); -} - -void AudioPolicyService::AudioCommandThread::stopOutputCommand(audio_io_handle_t output, - audio_stream_type_t stream, - audio_session_t session) -{ - sp command = new AudioCommand(); - command->mCommand = STOP_OUTPUT; - sp data = new StopOutputData(); - data->mIO = output; - data->mStream = stream; - data->mSession = session; - command->mParam = data; - ALOGV("AudioCommandThread() adding stop output %d", output); - sendCommand(command); -} - -void AudioPolicyService::AudioCommandThread::releaseOutputCommand(audio_io_handle_t output, - audio_stream_type_t stream, - audio_session_t session) -{ - sp command = new AudioCommand(); - command->mCommand = RELEASE_OUTPUT; - sp data = new ReleaseOutputData(); - data->mIO = output; - data->mStream = stream; - data->mSession = session; - command->mParam = data; - ALOGV("AudioCommandThread() adding release output %d", output); - sendCommand(command); -} - -status_t AudioPolicyService::AudioCommandThread::createAudioPatchCommand( - const struct audio_patch *patch, - audio_patch_handle_t *handle, - int delayMs) -{ - status_t status = NO_ERROR; - - sp command = new AudioCommand(); - command->mCommand = CREATE_AUDIO_PATCH; - CreateAudioPatchData *data = new CreateAudioPatchData(); - data->mPatch = *patch; - data->mHandle = *handle; - command->mParam = data; - command->mWaitStatus = true; - ALOGV("AudioCommandThread() adding create patch delay %d", delayMs); - status = sendCommand(command, delayMs); - if (status == NO_ERROR) { - *handle = data->mHandle; - } - return status; -} - -status_t AudioPolicyService::AudioCommandThread::releaseAudioPatchCommand(audio_patch_handle_t handle, - int delayMs) -{ - sp command = new AudioCommand(); - command->mCommand = RELEASE_AUDIO_PATCH; - ReleaseAudioPatchData *data = new ReleaseAudioPatchData(); - data->mHandle = handle; - command->mParam = data; - command->mWaitStatus = true; - ALOGV("AudioCommandThread() adding release patch delay %d", delayMs); - return sendCommand(command, delayMs); -} - -void AudioPolicyService::AudioCommandThread::updateAudioPortListCommand() -{ - sp command = new AudioCommand(); - command->mCommand = UPDATE_AUDIOPORT_LIST; - ALOGV("AudioCommandThread() adding update audio port list"); - sendCommand(command); -} - -void AudioPolicyService::AudioCommandThread::updateAudioPatchListCommand() -{ - spcommand = new AudioCommand(); - command->mCommand = UPDATE_AUDIOPATCH_LIST; - ALOGV("AudioCommandThread() adding update audio patch list"); - sendCommand(command); -} - -status_t AudioPolicyService::AudioCommandThread::setAudioPortConfigCommand( - const struct audio_port_config *config, int delayMs) -{ - sp command = new AudioCommand(); - command->mCommand = SET_AUDIOPORT_CONFIG; - SetAudioPortConfigData *data = new SetAudioPortConfigData(); - data->mConfig = *config; - command->mParam = data; - command->mWaitStatus = true; - ALOGV("AudioCommandThread() adding set port config delay %d", delayMs); - return sendCommand(command, delayMs); -} - -status_t AudioPolicyService::AudioCommandThread::sendCommand(sp& command, int delayMs) -{ - { - Mutex::Autolock _l(mLock); - insertCommand_l(command, delayMs); - mWaitWorkCV.signal(); - } - Mutex::Autolock _l(command->mLock); - while (command->mWaitStatus) { - nsecs_t timeOutNs = kAudioCommandTimeoutNs + milliseconds(delayMs); - if (command->mCond.waitRelative(command->mLock, timeOutNs) != NO_ERROR) { - command->mStatus = TIMED_OUT; - command->mWaitStatus = false; - } - } - return command->mStatus; -} - -// insertCommand_l() must be called with mLock held -void AudioPolicyService::AudioCommandThread::insertCommand_l(sp& command, int delayMs) -{ - ssize_t i; // not size_t because i will count down to -1 - Vector < sp > removedCommands; - command->mTime = systemTime() + milliseconds(delayMs); - - // acquire wake lock to make sure delayed commands are processed - if (mAudioCommands.isEmpty()) { - acquire_wake_lock(PARTIAL_WAKE_LOCK, mName.string()); - } - - // check same pending commands with later time stamps and eliminate them - for (i = mAudioCommands.size()-1; i >= 0; i--) { - sp command2 = mAudioCommands[i]; - // commands are sorted by increasing time stamp: no need to scan the rest of mAudioCommands - if (command2->mTime <= command->mTime) break; - - // create audio patch or release audio patch commands are equivalent - // with regard to filtering - if ((command->mCommand == CREATE_AUDIO_PATCH) || - (command->mCommand == RELEASE_AUDIO_PATCH)) { - if ((command2->mCommand != CREATE_AUDIO_PATCH) && - (command2->mCommand != RELEASE_AUDIO_PATCH)) { - continue; - } - } else if (command2->mCommand != command->mCommand) continue; - - switch (command->mCommand) { - case SET_PARAMETERS: { - ParametersData *data = (ParametersData *)command->mParam.get(); - ParametersData *data2 = (ParametersData *)command2->mParam.get(); - if (data->mIO != data2->mIO) break; - ALOGV("Comparing parameter command %s to new command %s", - data2->mKeyValuePairs.string(), data->mKeyValuePairs.string()); - AudioParameter param = AudioParameter(data->mKeyValuePairs); - AudioParameter param2 = AudioParameter(data2->mKeyValuePairs); - for (size_t j = 0; j < param.size(); j++) { - String8 key; - String8 value; - param.getAt(j, key, value); - for (size_t k = 0; k < param2.size(); k++) { - String8 key2; - String8 value2; - param2.getAt(k, key2, value2); - if (key2 == key) { - param2.remove(key2); - ALOGV("Filtering out parameter %s", key2.string()); - break; - } - } - } - // if all keys have been filtered out, remove the command. - // otherwise, update the key value pairs - if (param2.size() == 0) { - removedCommands.add(command2); - } else { - data2->mKeyValuePairs = param2.toString(); - } - command->mTime = command2->mTime; - // force delayMs to non 0 so that code below does not request to wait for - // command status as the command is now delayed - delayMs = 1; - } break; - - case SET_VOLUME: { - VolumeData *data = (VolumeData *)command->mParam.get(); - VolumeData *data2 = (VolumeData *)command2->mParam.get(); - if (data->mIO != data2->mIO) break; - if (data->mStream != data2->mStream) break; - ALOGV("Filtering out volume command on output %d for stream %d", - data->mIO, data->mStream); - removedCommands.add(command2); - command->mTime = command2->mTime; - // force delayMs to non 0 so that code below does not request to wait for - // command status as the command is now delayed - delayMs = 1; - } break; - - case CREATE_AUDIO_PATCH: - case RELEASE_AUDIO_PATCH: { - audio_patch_handle_t handle; - struct audio_patch patch; - if (command->mCommand == CREATE_AUDIO_PATCH) { - handle = ((CreateAudioPatchData *)command->mParam.get())->mHandle; - patch = ((CreateAudioPatchData *)command->mParam.get())->mPatch; - } else { - handle = ((ReleaseAudioPatchData *)command->mParam.get())->mHandle; - } - audio_patch_handle_t handle2; - struct audio_patch patch2; - if (command2->mCommand == CREATE_AUDIO_PATCH) { - handle2 = ((CreateAudioPatchData *)command2->mParam.get())->mHandle; - patch2 = ((CreateAudioPatchData *)command2->mParam.get())->mPatch; - } else { - handle2 = ((ReleaseAudioPatchData *)command2->mParam.get())->mHandle; - } - if (handle != handle2) break; - /* Filter CREATE_AUDIO_PATCH commands only when they are issued for - same output. */ - if( (command->mCommand == CREATE_AUDIO_PATCH) && - (command2->mCommand == CREATE_AUDIO_PATCH) ) { - bool isOutputDiff = false; - if (patch.num_sources == patch2.num_sources) { - for (unsigned count = 0; count < patch.num_sources; count++) { - if (patch.sources[count].id != patch2.sources[count].id) { - isOutputDiff = true; - break; - } - } - if (isOutputDiff) - break; - } - } - ALOGV("Filtering out %s audio patch command for handle %d", - (command->mCommand == CREATE_AUDIO_PATCH) ? "create" : "release", handle); - removedCommands.add(command2); - command->mTime = command2->mTime; - // force delayMs to non 0 so that code below does not request to wait for - // command status as the command is now delayed - delayMs = 1; - } break; - - case START_TONE: - case STOP_TONE: - default: - break; - } - } - - // remove filtered commands - for (size_t j = 0; j < removedCommands.size(); j++) { - // removed commands always have time stamps greater than current command - for (size_t k = i + 1; k < mAudioCommands.size(); k++) { - if (mAudioCommands[k].get() == removedCommands[j].get()) { - ALOGV("suppressing command: %d", mAudioCommands[k]->mCommand); - mAudioCommands.removeAt(k); - break; - } - } - } - removedCommands.clear(); - - // Disable wait for status if delay is not 0. - // Except for create audio patch command because the returned patch handle - // is needed by audio policy manager - if (delayMs != 0 && command->mCommand != CREATE_AUDIO_PATCH) { - command->mWaitStatus = false; - } - - // insert command at the right place according to its time stamp - ALOGV("inserting command: %d at index %zd, num commands %zu", - command->mCommand, i+1, mAudioCommands.size()); - mAudioCommands.insertAt(command, i + 1); -} - -void AudioPolicyService::AudioCommandThread::exit() -{ - ALOGV("AudioCommandThread::exit"); - { - AutoMutex _l(mLock); - requestExit(); - mWaitWorkCV.signal(); - } - requestExitAndWait(); -} - -void AudioPolicyService::AudioCommandThread::AudioCommand::dump(char* buffer, size_t size) -{ - snprintf(buffer, size, " %02d %06d.%03d %01u %p\n", - mCommand, - (int)ns2s(mTime), - (int)ns2ms(mTime)%1000, - mWaitStatus, - mParam.get()); -} - -/******* helpers for the service_ops callbacks defined below *********/ -void AudioPolicyService::setParameters(audio_io_handle_t ioHandle, - const char *keyValuePairs, - int delayMs) -{ - mAudioCommandThread->parametersCommand(ioHandle, keyValuePairs, - delayMs); -} - -int AudioPolicyService::setStreamVolume(audio_stream_type_t stream, - float volume, - audio_io_handle_t output, - int delayMs) -{ - return (int)mAudioCommandThread->volumeCommand(stream, volume, - output, delayMs); -} - -int AudioPolicyService::startTone(audio_policy_tone_t tone, - audio_stream_type_t stream) -{ - if (tone != AUDIO_POLICY_TONE_IN_CALL_NOTIFICATION) { - ALOGE("startTone: illegal tone requested (%d)", tone); - } - if (stream != AUDIO_STREAM_VOICE_CALL) { - ALOGE("startTone: illegal stream (%d) requested for tone %d", stream, - tone); - } - mTonePlaybackThread->startToneCommand(ToneGenerator::TONE_SUP_CALL_WAITING, - AUDIO_STREAM_VOICE_CALL); - return 0; -} - -int AudioPolicyService::stopTone() -{ - mTonePlaybackThread->stopToneCommand(); - return 0; -} - -int AudioPolicyService::setVoiceVolume(float volume, int delayMs) -{ - return (int)mAudioCommandThread->voiceVolumeCommand(volume, delayMs); -} - -extern "C" { -audio_module_handle_t aps_load_hw_module(void *service __unused, - const char *name); -audio_io_handle_t aps_open_output(void *service __unused, - audio_devices_t *pDevices, - uint32_t *pSamplingRate, - audio_format_t *pFormat, - audio_channel_mask_t *pChannelMask, - uint32_t *pLatencyMs, - audio_output_flags_t flags); - -audio_io_handle_t aps_open_output_on_module(void *service __unused, - audio_module_handle_t module, - audio_devices_t *pDevices, - uint32_t *pSamplingRate, - audio_format_t *pFormat, - audio_channel_mask_t *pChannelMask, - uint32_t *pLatencyMs, - audio_output_flags_t flags, - const audio_offload_info_t *offloadInfo); -audio_io_handle_t aps_open_dup_output(void *service __unused, - audio_io_handle_t output1, - audio_io_handle_t output2); -int aps_close_output(void *service __unused, audio_io_handle_t output); -int aps_suspend_output(void *service __unused, audio_io_handle_t output); -int aps_restore_output(void *service __unused, audio_io_handle_t output); -audio_io_handle_t aps_open_input(void *service __unused, - audio_devices_t *pDevices, - uint32_t *pSamplingRate, - audio_format_t *pFormat, - audio_channel_mask_t *pChannelMask, - audio_in_acoustics_t acoustics __unused); -audio_io_handle_t aps_open_input_on_module(void *service __unused, - audio_module_handle_t module, - audio_devices_t *pDevices, - uint32_t *pSamplingRate, - audio_format_t *pFormat, - audio_channel_mask_t *pChannelMask); -int aps_close_input(void *service __unused, audio_io_handle_t input); -int aps_invalidate_stream(void *service __unused, audio_stream_type_t stream); -int aps_move_effects(void *service __unused, int session, - audio_io_handle_t src_output, - audio_io_handle_t dst_output); -char * aps_get_parameters(void *service __unused, audio_io_handle_t io_handle, - const char *keys); -void aps_set_parameters(void *service, audio_io_handle_t io_handle, - const char *kv_pairs, int delay_ms); -int aps_set_stream_volume(void *service, audio_stream_type_t stream, - float volume, audio_io_handle_t output, - int delay_ms); -int aps_start_tone(void *service, audio_policy_tone_t tone, - audio_stream_type_t stream); -int aps_stop_tone(void *service); -int aps_set_voice_volume(void *service, float volume, int delay_ms); -}; - -namespace { - struct audio_policy_service_ops aps_ops = { - .open_output = aps_open_output, - .open_duplicate_output = aps_open_dup_output, - .close_output = aps_close_output, - .suspend_output = aps_suspend_output, - .restore_output = aps_restore_output, - .open_input = aps_open_input, - .close_input = aps_close_input, - .set_stream_volume = aps_set_stream_volume, - .invalidate_stream = aps_invalidate_stream, - .set_parameters = aps_set_parameters, - .get_parameters = aps_get_parameters, - .start_tone = aps_start_tone, - .stop_tone = aps_stop_tone, - .set_voice_volume = aps_set_voice_volume, - .move_effects = aps_move_effects, - .load_hw_module = aps_load_hw_module, - .open_output_on_module = aps_open_output_on_module, - .open_input_on_module = aps_open_input_on_module, - }; -}; // namespace - -}; // namespace android diff --git a/services/audiopolicy/AudioPolicyService.h b/services/audiopolicy/AudioPolicyService.h deleted file mode 100644 index 7c2b59d..0000000 --- a/services/audiopolicy/AudioPolicyService.h +++ /dev/null @@ -1,524 +0,0 @@ -/* - * Copyright (C) 2009 The Android Open Source Project - * - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - */ - -#ifndef ANDROID_AUDIOPOLICYSERVICE_H -#define ANDROID_AUDIOPOLICYSERVICE_H - -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#ifdef USE_LEGACY_AUDIO_POLICY -#include -#endif -#include "AudioPolicyEffects.h" -#include "AudioPolicyManager.h" - - -namespace android { - -// ---------------------------------------------------------------------------- - -class AudioPolicyService : - public BinderService, - public BnAudioPolicyService, - public IBinder::DeathRecipient -{ - friend class BinderService; - -public: - // for BinderService - static const char *getServiceName() ANDROID_API { return "media.audio_policy"; } - - virtual status_t dump(int fd, const Vector& args); - - // - // BnAudioPolicyService (see AudioPolicyInterface for method descriptions) - // - - virtual status_t setDeviceConnectionState(audio_devices_t device, - audio_policy_dev_state_t state, - const char *device_address, - const char *device_name); - virtual audio_policy_dev_state_t getDeviceConnectionState( - audio_devices_t device, - const char *device_address); - virtual status_t setPhoneState(audio_mode_t state); - virtual status_t setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config); - virtual audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage); - virtual audio_io_handle_t getOutput(audio_stream_type_t stream, - uint32_t samplingRate = 0, - audio_format_t format = AUDIO_FORMAT_DEFAULT, - audio_channel_mask_t channelMask = 0, - audio_output_flags_t flags = - AUDIO_OUTPUT_FLAG_NONE, - const audio_offload_info_t *offloadInfo = NULL); - virtual status_t getOutputForAttr(const audio_attributes_t *attr, - audio_io_handle_t *output, - audio_session_t session, - audio_stream_type_t *stream, - uint32_t samplingRate = 0, - audio_format_t format = AUDIO_FORMAT_DEFAULT, - audio_channel_mask_t channelMask = 0, - audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, - const audio_offload_info_t *offloadInfo = NULL); - virtual status_t startOutput(audio_io_handle_t output, - audio_stream_type_t stream, - audio_session_t session); - virtual status_t stopOutput(audio_io_handle_t output, - audio_stream_type_t stream, - audio_session_t session); - virtual void releaseOutput(audio_io_handle_t output, - audio_stream_type_t stream, - audio_session_t session); - virtual status_t getInputForAttr(const audio_attributes_t *attr, - audio_io_handle_t *input, - audio_session_t session, - uint32_t samplingRate, - audio_format_t format, - audio_channel_mask_t channelMask, - audio_input_flags_t flags); - virtual status_t startInput(audio_io_handle_t input, - audio_session_t session); - virtual status_t stopInput(audio_io_handle_t input, - audio_session_t session); - virtual void releaseInput(audio_io_handle_t input, - audio_session_t session); - virtual status_t initStreamVolume(audio_stream_type_t stream, - int indexMin, - int indexMax); - virtual status_t setStreamVolumeIndex(audio_stream_type_t stream, - int index, - audio_devices_t device); - virtual status_t getStreamVolumeIndex(audio_stream_type_t stream, - int *index, - audio_devices_t device); - - virtual uint32_t getStrategyForStream(audio_stream_type_t stream); - virtual audio_devices_t getDevicesForStream(audio_stream_type_t stream); - - virtual audio_io_handle_t getOutputForEffect(const effect_descriptor_t *desc); - virtual status_t registerEffect(const effect_descriptor_t *desc, - audio_io_handle_t io, - uint32_t strategy, - int session, - int id); - virtual status_t unregisterEffect(int id); - virtual status_t setEffectEnabled(int id, bool enabled); - virtual bool isStreamActive(audio_stream_type_t stream, uint32_t inPastMs = 0) const; - virtual bool isStreamActiveRemotely(audio_stream_type_t stream, uint32_t inPastMs = 0) const; - virtual bool isSourceActive(audio_source_t source) const; - - virtual status_t queryDefaultPreProcessing(int audioSession, - effect_descriptor_t *descriptors, - uint32_t *count); - virtual status_t onTransact( - uint32_t code, - const Parcel& data, - Parcel* reply, - uint32_t flags); - - // IBinder::DeathRecipient - virtual void binderDied(const wp& who); - - // RefBase - virtual void onFirstRef(); - - // - // Helpers for the struct audio_policy_service_ops implementation. - // This is used by the audio policy manager for certain operations that - // are implemented by the policy service. - // - virtual void setParameters(audio_io_handle_t ioHandle, - const char *keyValuePairs, - int delayMs); - - virtual status_t setStreamVolume(audio_stream_type_t stream, - float volume, - audio_io_handle_t output, - int delayMs = 0); - virtual status_t startTone(audio_policy_tone_t tone, audio_stream_type_t stream); - virtual status_t stopTone(); - virtual status_t setVoiceVolume(float volume, int delayMs = 0); - virtual bool isOffloadSupported(const audio_offload_info_t &config); - - virtual status_t listAudioPorts(audio_port_role_t role, - audio_port_type_t type, - unsigned int *num_ports, - struct audio_port *ports, - unsigned int *generation); - virtual status_t getAudioPort(struct audio_port *port); - virtual status_t createAudioPatch(const struct audio_patch *patch, - audio_patch_handle_t *handle); - virtual status_t releaseAudioPatch(audio_patch_handle_t handle); - virtual status_t listAudioPatches(unsigned int *num_patches, - struct audio_patch *patches, - unsigned int *generation); - virtual status_t setAudioPortConfig(const struct audio_port_config *config); - - virtual void registerClient(const sp& client); - - virtual status_t acquireSoundTriggerSession(audio_session_t *session, - audio_io_handle_t *ioHandle, - audio_devices_t *device); - - virtual status_t releaseSoundTriggerSession(audio_session_t session); - - virtual audio_mode_t getPhoneState(); - - virtual status_t registerPolicyMixes(Vector mixes, bool registration); - - status_t doStopOutput(audio_io_handle_t output, - audio_stream_type_t stream, - audio_session_t session); - void doReleaseOutput(audio_io_handle_t output, - audio_stream_type_t stream, - audio_session_t session); - - status_t clientCreateAudioPatch(const struct audio_patch *patch, - audio_patch_handle_t *handle, - int delayMs); - status_t clientReleaseAudioPatch(audio_patch_handle_t handle, - int delayMs); - virtual status_t clientSetAudioPortConfig(const struct audio_port_config *config, - int delayMs); - - void removeNotificationClient(uid_t uid); - void onAudioPortListUpdate(); - void doOnAudioPortListUpdate(); - void onAudioPatchListUpdate(); - void doOnAudioPatchListUpdate(); - -private: - AudioPolicyService() ANDROID_API; - virtual ~AudioPolicyService(); - - status_t dumpInternals(int fd); - - // Thread used for tone playback and to send audio config commands to audio flinger - // For tone playback, using a separate thread is necessary to avoid deadlock with mLock because - // startTone() and stopTone() are normally called with mLock locked and requesting a tone start - // or stop will cause calls to AudioPolicyService and an attempt to lock mLock. - // For audio config commands, it is necessary because audio flinger requires that the calling - // process (user) has permission to modify audio settings. - class AudioCommandThread : public Thread { - class AudioCommand; - public: - - // commands for tone AudioCommand - enum { - START_TONE, - STOP_TONE, - SET_VOLUME, - SET_PARAMETERS, - SET_VOICE_VOLUME, - STOP_OUTPUT, - RELEASE_OUTPUT, - CREATE_AUDIO_PATCH, - RELEASE_AUDIO_PATCH, - UPDATE_AUDIOPORT_LIST, - UPDATE_AUDIOPATCH_LIST, - SET_AUDIOPORT_CONFIG, - }; - - AudioCommandThread (String8 name, const wp& service); - virtual ~AudioCommandThread(); - - status_t dump(int fd); - - // Thread virtuals - virtual void onFirstRef(); - virtual bool threadLoop(); - - void exit(); - void startToneCommand(ToneGenerator::tone_type type, - audio_stream_type_t stream); - void stopToneCommand(); - status_t volumeCommand(audio_stream_type_t stream, float volume, - audio_io_handle_t output, int delayMs = 0); - status_t parametersCommand(audio_io_handle_t ioHandle, - const char *keyValuePairs, int delayMs = 0); - status_t voiceVolumeCommand(float volume, int delayMs = 0); - void stopOutputCommand(audio_io_handle_t output, - audio_stream_type_t stream, - audio_session_t session); - void releaseOutputCommand(audio_io_handle_t output, - audio_stream_type_t stream, - audio_session_t session); - status_t sendCommand(sp& command, int delayMs = 0); - void insertCommand_l(sp& command, int delayMs = 0); - status_t createAudioPatchCommand(const struct audio_patch *patch, - audio_patch_handle_t *handle, - int delayMs); - status_t releaseAudioPatchCommand(audio_patch_handle_t handle, - int delayMs); - void updateAudioPortListCommand(); - void updateAudioPatchListCommand(); - status_t setAudioPortConfigCommand(const struct audio_port_config *config, - int delayMs); - void insertCommand_l(AudioCommand *command, int delayMs = 0); - - private: - class AudioCommandData; - - // descriptor for requested tone playback event - class AudioCommand: public RefBase { - - public: - AudioCommand() - : mCommand(-1), mStatus(NO_ERROR), mWaitStatus(false) {} - - void dump(char* buffer, size_t size); - - int mCommand; // START_TONE, STOP_TONE ... - nsecs_t mTime; // time stamp - Mutex mLock; // mutex associated to mCond - Condition mCond; // condition for status return - status_t mStatus; // command status - bool mWaitStatus; // true if caller is waiting for status - sp mParam; // command specific parameter data - }; - - class AudioCommandData: public RefBase { - public: - virtual ~AudioCommandData() {} - protected: - AudioCommandData() {} - }; - - class ToneData : public AudioCommandData { - public: - ToneGenerator::tone_type mType; // tone type (START_TONE only) - audio_stream_type_t mStream; // stream type (START_TONE only) - }; - - class VolumeData : public AudioCommandData { - public: - audio_stream_type_t mStream; - float mVolume; - audio_io_handle_t mIO; - }; - - class ParametersData : public AudioCommandData { - public: - audio_io_handle_t mIO; - String8 mKeyValuePairs; - }; - - class VoiceVolumeData : public AudioCommandData { - public: - float mVolume; - }; - - class StopOutputData : public AudioCommandData { - public: - audio_io_handle_t mIO; - audio_stream_type_t mStream; - audio_session_t mSession; - }; - - class ReleaseOutputData : public AudioCommandData { - public: - audio_io_handle_t mIO; - audio_stream_type_t mStream; - audio_session_t mSession; - }; - - class CreateAudioPatchData : public AudioCommandData { - public: - struct audio_patch mPatch; - audio_patch_handle_t mHandle; - }; - - class ReleaseAudioPatchData : public AudioCommandData { - public: - audio_patch_handle_t mHandle; - }; - - class SetAudioPortConfigData : public AudioCommandData { - public: - struct audio_port_config mConfig; - }; - - Mutex mLock; - Condition mWaitWorkCV; - Vector < sp > mAudioCommands; // list of pending commands - ToneGenerator *mpToneGenerator; // the tone generator - sp mLastCommand; // last processed command (used by dump) - String8 mName; // string used by wake lock fo delayed commands - wp mService; - }; - - class AudioPolicyClient : public AudioPolicyClientInterface - { - public: - AudioPolicyClient(AudioPolicyService *service) : mAudioPolicyService(service) {} - virtual ~AudioPolicyClient() {} - - // - // Audio HW module functions - // - - // loads a HW module. - virtual audio_module_handle_t loadHwModule(const char *name); - - // - // Audio output Control functions - // - - // opens an audio output with the requested parameters. The parameter values can indicate to use the default values - // in case the audio policy manager has no specific requirements for the output being opened. - // When the function returns, the parameter values reflect the actual values used by the audio hardware output stream. - // The audio policy manager can check if the proposed parameters are suitable or not and act accordingly. - virtual status_t openOutput(audio_module_handle_t module, - audio_io_handle_t *output, - audio_config_t *config, - audio_devices_t *devices, - const String8& address, - uint32_t *latencyMs, - audio_output_flags_t flags); - // creates a special output that is duplicated to the two outputs passed as arguments. The duplication is performed by - // a special mixer thread in the AudioFlinger. - virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, audio_io_handle_t output2); - // closes the output stream - virtual status_t closeOutput(audio_io_handle_t output); - // suspends the output. When an output is suspended, the corresponding audio hardware output stream is placed in - // standby and the AudioTracks attached to the mixer thread are still processed but the output mix is discarded. - virtual status_t suspendOutput(audio_io_handle_t output); - // restores a suspended output. - virtual status_t restoreOutput(audio_io_handle_t output); - - // - // Audio input Control functions - // - - // opens an audio input - virtual audio_io_handle_t openInput(audio_module_handle_t module, - audio_io_handle_t *input, - audio_config_t *config, - audio_devices_t *devices, - const String8& address, - audio_source_t source, - audio_input_flags_t flags); - // closes an audio input - virtual status_t closeInput(audio_io_handle_t input); - // - // misc control functions - // - - // set a stream volume for a particular output. For the same user setting, a given stream type can have different volumes - // for each output (destination device) it is attached to. - virtual status_t setStreamVolume(audio_stream_type_t stream, float volume, audio_io_handle_t output, int delayMs = 0); - - // invalidate a stream type, causing a reroute to an unspecified new output - virtual status_t invalidateStream(audio_stream_type_t stream); - - // function enabling to send proprietary informations directly from audio policy manager to audio hardware interface. - virtual void setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs, int delayMs = 0); - // function enabling to receive proprietary informations directly from audio hardware interface to audio policy manager. - virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys); - - // request the playback of a tone on the specified stream: used for instance to replace notification sounds when playing - // over a telephony device during a phone call. - virtual status_t startTone(audio_policy_tone_t tone, audio_stream_type_t stream); - virtual status_t stopTone(); - - // set down link audio volume. - virtual status_t setVoiceVolume(float volume, int delayMs = 0); - - // move effect to the specified output - virtual status_t moveEffects(int session, - audio_io_handle_t srcOutput, - audio_io_handle_t dstOutput); - - /* Create a patch between several source and sink ports */ - virtual status_t createAudioPatch(const struct audio_patch *patch, - audio_patch_handle_t *handle, - int delayMs); - - /* Release a patch */ - virtual status_t releaseAudioPatch(audio_patch_handle_t handle, - int delayMs); - - /* Set audio port configuration */ - virtual status_t setAudioPortConfig(const struct audio_port_config *config, int delayMs); - - virtual void onAudioPortListUpdate(); - virtual void onAudioPatchListUpdate(); - - virtual audio_unique_id_t newAudioUniqueId(); - - private: - AudioPolicyService *mAudioPolicyService; - }; - - // --- Notification Client --- - class NotificationClient : public IBinder::DeathRecipient { - public: - NotificationClient(const sp& service, - const sp& client, - uid_t uid); - virtual ~NotificationClient(); - - void onAudioPortListUpdate(); - void onAudioPatchListUpdate(); - - // IBinder::DeathRecipient - virtual void binderDied(const wp& who); - - private: - NotificationClient(const NotificationClient&); - NotificationClient& operator = (const NotificationClient&); - - const wp mService; - const uid_t mUid; - const sp mAudioPolicyServiceClient; - }; - - // Internal dump utilities. - status_t dumpPermissionDenial(int fd); - - - mutable Mutex mLock; // prevents concurrent access to AudioPolicy manager functions changing - // device connection state or routing - sp mAudioCommandThread; // audio commands thread - sp mTonePlaybackThread; // tone playback thread - sp mOutputCommandThread; // process stop and release output - struct audio_policy_device *mpAudioPolicyDev; - struct audio_policy *mpAudioPolicy; - AudioPolicyInterface *mAudioPolicyManager; - AudioPolicyClient *mAudioPolicyClient; - - DefaultKeyedVector< uid_t, sp > mNotificationClients; - Mutex mNotificationClientsLock; // protects mNotificationClients - // Manage all effects configured in audio_effects.conf - sp mAudioPolicyEffects; - audio_mode_t mPhoneState; -}; - -}; // namespace android - -#endif // ANDROID_AUDIOPOLICYSERVICE_H diff --git a/services/audiopolicy/audio_policy_conf.h b/services/audiopolicy/audio_policy_conf.h deleted file mode 100644 index 2535a67..0000000 --- a/services/audiopolicy/audio_policy_conf.h +++ /dev/null @@ -1,77 +0,0 @@ -/* - * Copyright (C) 2012 The Android Open Source Project - * - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - */ - - -#ifndef ANDROID_AUDIO_POLICY_CONF_H -#define ANDROID_AUDIO_POLICY_CONF_H - - -///////////////////////////////////////////////// -// Definitions for audio policy configuration file (audio_policy.conf) -///////////////////////////////////////////////// - -#define AUDIO_HARDWARE_MODULE_ID_MAX_LEN 32 - -#define AUDIO_POLICY_CONFIG_FILE "/system/etc/audio_policy.conf" -#define AUDIO_POLICY_VENDOR_CONFIG_FILE "/vendor/etc/audio_policy.conf" - -// global configuration -#define GLOBAL_CONFIG_TAG "global_configuration" - -#define ATTACHED_OUTPUT_DEVICES_TAG "attached_output_devices" -#define DEFAULT_OUTPUT_DEVICE_TAG "default_output_device" -#define ATTACHED_INPUT_DEVICES_TAG "attached_input_devices" -#define SPEAKER_DRC_ENABLED_TAG "speaker_drc_enabled" -#define AUDIO_HAL_VERSION_TAG "audio_hal_version" - -// hw modules descriptions -#define AUDIO_HW_MODULE_TAG "audio_hw_modules" - -#define OUTPUTS_TAG "outputs" -#define INPUTS_TAG "inputs" - -#define SAMPLING_RATES_TAG "sampling_rates" -#define FORMATS_TAG "formats" -#define CHANNELS_TAG "channel_masks" -#define DEVICES_TAG "devices" -#define FLAGS_TAG "flags" - -#define DYNAMIC_VALUE_TAG "dynamic" // special value for "channel_masks", "sampling_rates" and - // "formats" in outputs descriptors indicating that supported - // values should be queried after opening the output. - -#define DEVICES_TAG "devices" -#define DEVICE_TYPE "type" -#define DEVICE_ADDRESS "address" - -#define MIXERS_TAG "mixers" -#define MIXER_TYPE "type" -#define MIXER_TYPE_MUX "mux" -#define MIXER_TYPE_MIX "mix" - -#define GAINS_TAG "gains" -#define GAIN_MODE "mode" -#define GAIN_CHANNELS "channel_mask" -#define GAIN_MIN_VALUE "min_value_mB" -#define GAIN_MAX_VALUE "max_value_mB" -#define GAIN_DEFAULT_VALUE "default_value_mB" -#define GAIN_STEP_VALUE "step_value_mB" -#define GAIN_MIN_RAMP_MS "min_ramp_ms" -#define GAIN_MAX_RAMP_MS "max_ramp_ms" - - - -#endif // ANDROID_AUDIO_POLICY_CONF_H diff --git a/services/audiopolicy/manager/AudioPolicyFactory.cpp b/services/audiopolicy/manager/AudioPolicyFactory.cpp new file mode 100644 index 0000000..9910a1f --- /dev/null +++ b/services/audiopolicy/manager/AudioPolicyFactory.cpp @@ -0,0 +1,32 @@ +/* + * Copyright (C) 2014 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#include "managerdefault/AudioPolicyManager.h" + +namespace android { + +extern "C" AudioPolicyInterface* createAudioPolicyManager( + AudioPolicyClientInterface *clientInterface) +{ + return new AudioPolicyManager(clientInterface); +} + +extern "C" void destroyAudioPolicyManager(AudioPolicyInterface *interface) +{ + delete interface; +} + +}; // namespace android diff --git a/services/audiopolicy/managerdefault/ApmImplDefinitions.h b/services/audiopolicy/managerdefault/ApmImplDefinitions.h new file mode 100644 index 0000000..620979b --- /dev/null +++ b/services/audiopolicy/managerdefault/ApmImplDefinitions.h @@ -0,0 +1,32 @@ +/* + * Copyright (C) 2015 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +namespace android { + +enum routing_strategy { + STRATEGY_MEDIA, + STRATEGY_PHONE, + STRATEGY_SONIFICATION, + STRATEGY_SONIFICATION_RESPECTFUL, + STRATEGY_DTMF, + STRATEGY_ENFORCED_AUDIBLE, + STRATEGY_TRANSMITTED_THROUGH_SPEAKER, + STRATEGY_ACCESSIBILITY, + STRATEGY_REROUTING, + NUM_STRATEGIES +}; + +}; //namespace android diff --git a/services/audiopolicy/managerdefault/AudioInputDescriptor.cpp b/services/audiopolicy/managerdefault/AudioInputDescriptor.cpp new file mode 100644 index 0000000..f4054c8 --- /dev/null +++ b/services/audiopolicy/managerdefault/AudioInputDescriptor.cpp @@ -0,0 +1,100 @@ +/* + * Copyright (C) 2015 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#define LOG_TAG "APM::AudioInputDescriptor" +//#define LOG_NDEBUG 0 + +#include "AudioPolicyManager.h" + +namespace android { + +AudioInputDescriptor::AudioInputDescriptor(const sp& profile) + : mId(0), mIoHandle(0), + mDevice(AUDIO_DEVICE_NONE), mPolicyMix(NULL), mPatchHandle(0), mRefCount(0), + mInputSource(AUDIO_SOURCE_DEFAULT), mProfile(profile), mIsSoundTrigger(false) +{ + if (profile != NULL) { + mSamplingRate = profile->pickSamplingRate(); + mFormat = profile->pickFormat(); + mChannelMask = profile->pickChannelMask(); + if (profile->mGains.size() > 0) { + profile->mGains[0]->getDefaultConfig(&mGain); + } + } +} + +void AudioInputDescriptor::toAudioPortConfig( + struct audio_port_config *dstConfig, + const struct audio_port_config *srcConfig) const +{ + ALOG_ASSERT(mProfile != 0, + "toAudioPortConfig() called on input with null profile %d", mIoHandle); + dstConfig->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK| + AUDIO_PORT_CONFIG_FORMAT|AUDIO_PORT_CONFIG_GAIN; + if (srcConfig != NULL) { + dstConfig->config_mask |= srcConfig->config_mask; + } + + AudioPortConfig::toAudioPortConfig(dstConfig, srcConfig); + + dstConfig->id = mId; + dstConfig->role = AUDIO_PORT_ROLE_SINK; + dstConfig->type = AUDIO_PORT_TYPE_MIX; + dstConfig->ext.mix.hw_module = mProfile->mModule->mHandle; + dstConfig->ext.mix.handle = mIoHandle; + dstConfig->ext.mix.usecase.source = mInputSource; +} + +void AudioInputDescriptor::toAudioPort( + struct audio_port *port) const +{ + ALOG_ASSERT(mProfile != 0, "toAudioPort() called on input with null profile %d", mIoHandle); + + mProfile->toAudioPort(port); + port->id = mId; + toAudioPortConfig(&port->active_config); + port->ext.mix.hw_module = mProfile->mModule->mHandle; + port->ext.mix.handle = mIoHandle; + port->ext.mix.latency_class = AUDIO_LATENCY_NORMAL; +} + +status_t AudioInputDescriptor::dump(int fd) +{ + const size_t SIZE = 256; + char buffer[SIZE]; + String8 result; + + snprintf(buffer, SIZE, " ID: %d\n", mId); + result.append(buffer); + snprintf(buffer, SIZE, " Sampling rate: %d\n", mSamplingRate); + result.append(buffer); + snprintf(buffer, SIZE, " Format: %d\n", mFormat); + result.append(buffer); + snprintf(buffer, SIZE, " Channels: %08x\n", mChannelMask); + result.append(buffer); + snprintf(buffer, SIZE, " Devices %08x\n", mDevice); + result.append(buffer); + snprintf(buffer, SIZE, " Ref Count %d\n", mRefCount); + result.append(buffer); + snprintf(buffer, SIZE, " Open Ref Count %d\n", mOpenRefCount); + result.append(buffer); + + write(fd, result.string(), result.size()); + + return NO_ERROR; +} + +}; //namespace android diff --git a/services/audiopolicy/managerdefault/AudioInputDescriptor.h b/services/audiopolicy/managerdefault/AudioInputDescriptor.h new file mode 100644 index 0000000..02579e6 --- /dev/null +++ b/services/audiopolicy/managerdefault/AudioInputDescriptor.h @@ -0,0 +1,48 @@ +/* + * Copyright (C) 2015 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +namespace android { + +// descriptor for audio inputs. Used to maintain current configuration of each opened audio input +// and keep track of the usage of this input. +class AudioInputDescriptor: public AudioPortConfig +{ +public: + AudioInputDescriptor(const sp& profile); + + status_t dump(int fd); + + audio_port_handle_t mId; + audio_io_handle_t mIoHandle; // input handle + audio_devices_t mDevice; // current device this input is routed to + AudioMix *mPolicyMix; // non NULL when used by a dynamic policy + audio_patch_handle_t mPatchHandle; + uint32_t mRefCount; // number of AudioRecord clients using + // this input + uint32_t mOpenRefCount; + audio_source_t mInputSource; // input source selected by application + //(mediarecorder.h) + const sp mProfile; // I/O profile this output derives from + SortedVector mSessions; // audio sessions attached to this input + bool mIsSoundTrigger; // used by a soundtrigger capture + + virtual void toAudioPortConfig(struct audio_port_config *dstConfig, + const struct audio_port_config *srcConfig = NULL) const; + virtual sp getAudioPort() const { return mProfile; } + void toAudioPort(struct audio_port *port) const; +}; + +}; // namespace android diff --git a/services/audiopolicy/managerdefault/AudioOutputDescriptor.cpp b/services/audiopolicy/managerdefault/AudioOutputDescriptor.cpp new file mode 100644 index 0000000..4b85972 --- /dev/null +++ b/services/audiopolicy/managerdefault/AudioOutputDescriptor.cpp @@ -0,0 +1,221 @@ +/* + * Copyright (C) 2015 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#define LOG_TAG "APM::AudioOutputDescriptor" +//#define LOG_NDEBUG 0 + +#include "AudioPolicyManager.h" + +namespace android { + +AudioOutputDescriptor::AudioOutputDescriptor( + const sp& profile) + : mId(0), mIoHandle(0), mLatency(0), + mFlags((audio_output_flags_t)0), mDevice(AUDIO_DEVICE_NONE), mPolicyMix(NULL), + mPatchHandle(0), + mOutput1(0), mOutput2(0), mProfile(profile), mDirectOpenCount(0) +{ + // clear usage count for all stream types + for (int i = 0; i < AUDIO_STREAM_CNT; i++) { + mRefCount[i] = 0; + mCurVolume[i] = -1.0; + mMuteCount[i] = 0; + mStopTime[i] = 0; + } + for (int i = 0; i < NUM_STRATEGIES; i++) { + mStrategyMutedByDevice[i] = false; + } + if (profile != NULL) { + mFlags = (audio_output_flags_t)profile->mFlags; + mSamplingRate = profile->pickSamplingRate(); + mFormat = profile->pickFormat(); + mChannelMask = profile->pickChannelMask(); + if (profile->mGains.size() > 0) { + profile->mGains[0]->getDefaultConfig(&mGain); + } + } +} + +audio_devices_t AudioOutputDescriptor::device() const +{ + if (isDuplicated()) { + return (audio_devices_t)(mOutput1->mDevice | mOutput2->mDevice); + } else { + return mDevice; + } +} + +uint32_t AudioOutputDescriptor::latency() +{ + if (isDuplicated()) { + return (mOutput1->mLatency > mOutput2->mLatency) ? mOutput1->mLatency : mOutput2->mLatency; + } else { + return mLatency; + } +} + +bool AudioOutputDescriptor::sharesHwModuleWith( + const sp outputDesc) +{ + if (isDuplicated()) { + return mOutput1->sharesHwModuleWith(outputDesc) || mOutput2->sharesHwModuleWith(outputDesc); + } else if (outputDesc->isDuplicated()){ + return sharesHwModuleWith(outputDesc->mOutput1) || sharesHwModuleWith(outputDesc->mOutput2); + } else { + return (mProfile->mModule == outputDesc->mProfile->mModule); + } +} + +void AudioOutputDescriptor::changeRefCount(audio_stream_type_t stream, + int delta) +{ + // forward usage count change to attached outputs + if (isDuplicated()) { + mOutput1->changeRefCount(stream, delta); + mOutput2->changeRefCount(stream, delta); + } + if ((delta + (int)mRefCount[stream]) < 0) { + ALOGW("changeRefCount() invalid delta %d for stream %d, refCount %d", + delta, stream, mRefCount[stream]); + mRefCount[stream] = 0; + return; + } + mRefCount[stream] += delta; + ALOGV("changeRefCount() stream %d, count %d", stream, mRefCount[stream]); +} + +audio_devices_t AudioOutputDescriptor::supportedDevices() +{ + if (isDuplicated()) { + return (audio_devices_t)(mOutput1->supportedDevices() | mOutput2->supportedDevices()); + } else { + return mProfile->mSupportedDevices.types() ; + } +} + +bool AudioOutputDescriptor::isActive(uint32_t inPastMs) const +{ + return isStrategyActive(NUM_STRATEGIES, inPastMs); +} + +bool AudioOutputDescriptor::isStrategyActive(routing_strategy strategy, + uint32_t inPastMs, + nsecs_t sysTime) const +{ + if ((sysTime == 0) && (inPastMs != 0)) { + sysTime = systemTime(); + } + for (int i = 0; i < (int)AUDIO_STREAM_CNT; i++) { + if (i == AUDIO_STREAM_PATCH) { + continue; + } + if (((AudioPolicyManager::getStrategy((audio_stream_type_t)i) == strategy) || + (NUM_STRATEGIES == strategy)) && + isStreamActive((audio_stream_type_t)i, inPastMs, sysTime)) { + return true; + } + } + return false; +} + +bool AudioOutputDescriptor::isStreamActive(audio_stream_type_t stream, + uint32_t inPastMs, + nsecs_t sysTime) const +{ + if (mRefCount[stream] != 0) { + return true; + } + if (inPastMs == 0) { + return false; + } + if (sysTime == 0) { + sysTime = systemTime(); + } + if (ns2ms(sysTime - mStopTime[stream]) < inPastMs) { + return true; + } + return false; +} + +void AudioOutputDescriptor::toAudioPortConfig( + struct audio_port_config *dstConfig, + const struct audio_port_config *srcConfig) const +{ + ALOG_ASSERT(!isDuplicated(), "toAudioPortConfig() called on duplicated output %d", mIoHandle); + + dstConfig->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK| + AUDIO_PORT_CONFIG_FORMAT|AUDIO_PORT_CONFIG_GAIN; + if (srcConfig != NULL) { + dstConfig->config_mask |= srcConfig->config_mask; + } + AudioPortConfig::toAudioPortConfig(dstConfig, srcConfig); + + dstConfig->id = mId; + dstConfig->role = AUDIO_PORT_ROLE_SOURCE; + dstConfig->type = AUDIO_PORT_TYPE_MIX; + dstConfig->ext.mix.hw_module = mProfile->mModule->mHandle; + dstConfig->ext.mix.handle = mIoHandle; + dstConfig->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; +} + +void AudioOutputDescriptor::toAudioPort( + struct audio_port *port) const +{ + ALOG_ASSERT(!isDuplicated(), "toAudioPort() called on duplicated output %d", mIoHandle); + mProfile->toAudioPort(port); + port->id = mId; + toAudioPortConfig(&port->active_config); + port->ext.mix.hw_module = mProfile->mModule->mHandle; + port->ext.mix.handle = mIoHandle; + port->ext.mix.latency_class = + mFlags & AUDIO_OUTPUT_FLAG_FAST ? AUDIO_LATENCY_LOW : AUDIO_LATENCY_NORMAL; +} + +status_t AudioOutputDescriptor::dump(int fd) +{ + const size_t SIZE = 256; + char buffer[SIZE]; + String8 result; + + snprintf(buffer, SIZE, " ID: %d\n", mId); + result.append(buffer); + snprintf(buffer, SIZE, " Sampling rate: %d\n", mSamplingRate); + result.append(buffer); + snprintf(buffer, SIZE, " Format: %08x\n", mFormat); + result.append(buffer); + snprintf(buffer, SIZE, " Channels: %08x\n", mChannelMask); + result.append(buffer); + snprintf(buffer, SIZE, " Latency: %d\n", mLatency); + result.append(buffer); + snprintf(buffer, SIZE, " Flags %08x\n", mFlags); + result.append(buffer); + snprintf(buffer, SIZE, " Devices %08x\n", device()); + result.append(buffer); + snprintf(buffer, SIZE, " Stream volume refCount muteCount\n"); + result.append(buffer); + for (int i = 0; i < (int)AUDIO_STREAM_CNT; i++) { + snprintf(buffer, SIZE, " %02d %.03f %02d %02d\n", + i, mCurVolume[i], mRefCount[i], mMuteCount[i]); + result.append(buffer); + } + write(fd, result.string(), result.size()); + + return NO_ERROR; +} + + + +}; //namespace android diff --git a/services/audiopolicy/managerdefault/AudioOutputDescriptor.h b/services/audiopolicy/managerdefault/AudioOutputDescriptor.h new file mode 100644 index 0000000..32f46e4 --- /dev/null +++ b/services/audiopolicy/managerdefault/AudioOutputDescriptor.h @@ -0,0 +1,69 @@ +/* + * Copyright (C) 2015 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#include "ApmImplDefinitions.h" + +namespace android { + +// descriptor for audio outputs. Used to maintain current configuration of each opened audio output +// and keep track of the usage of this output by each audio stream type. +class AudioOutputDescriptor: public AudioPortConfig +{ +public: + AudioOutputDescriptor(const sp& profile); + + status_t dump(int fd); + + audio_devices_t device() const; + void changeRefCount(audio_stream_type_t stream, int delta); + + bool isDuplicated() const { return (mOutput1 != NULL && mOutput2 != NULL); } + audio_devices_t supportedDevices(); + uint32_t latency(); + bool sharesHwModuleWith(const sp outputDesc); + bool isActive(uint32_t inPastMs = 0) const; + bool isStreamActive(audio_stream_type_t stream, + uint32_t inPastMs = 0, + nsecs_t sysTime = 0) const; + bool isStrategyActive(routing_strategy strategy, + uint32_t inPastMs = 0, + nsecs_t sysTime = 0) const; + + virtual void toAudioPortConfig(struct audio_port_config *dstConfig, + const struct audio_port_config *srcConfig = NULL) const; + virtual sp getAudioPort() const { return mProfile; } + void toAudioPort(struct audio_port *port) const; + + audio_port_handle_t mId; + audio_io_handle_t mIoHandle; // output handle + uint32_t mLatency; // + audio_output_flags_t mFlags; // + audio_devices_t mDevice; // current device this output is routed to + AudioMix *mPolicyMix; // non NULL when used by a dynamic policy + audio_patch_handle_t mPatchHandle; + uint32_t mRefCount[AUDIO_STREAM_CNT]; // number of streams of each type using this output + nsecs_t mStopTime[AUDIO_STREAM_CNT]; + sp mOutput1; // used by duplicated outputs: first output + sp mOutput2; // used by duplicated outputs: second output + float mCurVolume[AUDIO_STREAM_CNT]; // current stream volume + int mMuteCount[AUDIO_STREAM_CNT]; // mute request counter + const sp mProfile; // I/O profile this output derives from + bool mStrategyMutedByDevice[NUM_STRATEGIES]; // strategies muted because of incompatible + // device selection. See checkDeviceMuteStrategies() + uint32_t mDirectOpenCount; // number of clients using this output (direct outputs only) +}; + +}; // namespace android diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp new file mode 100644 index 0000000..b48dc80 --- /dev/null +++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp @@ -0,0 +1,5766 @@ +/* + * Copyright (C) 2009 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#define LOG_TAG "APM::AudioPolicyManager" +//#define LOG_NDEBUG 0 + +//#define VERY_VERBOSE_LOGGING +#ifdef VERY_VERBOSE_LOGGING +#define ALOGVV ALOGV +#else +#define ALOGVV(a...) do { } while(0) +#endif + +// A device mask for all audio input devices that are considered "virtual" when evaluating +// active inputs in getActiveInput() +#define APM_AUDIO_IN_DEVICE_VIRTUAL_ALL (AUDIO_DEVICE_IN_REMOTE_SUBMIX|AUDIO_DEVICE_IN_FM_TUNER) +// A device mask for all audio output devices that are considered "remote" when evaluating +// active output devices in isStreamActiveRemotely() +#define APM_AUDIO_OUT_DEVICE_REMOTE_ALL AUDIO_DEVICE_OUT_REMOTE_SUBMIX +// A device mask for all audio input and output devices where matching inputs/outputs on device +// type alone is not enough: the address must match too +#define APM_AUDIO_DEVICE_MATCH_ADDRESS_ALL (AUDIO_DEVICE_IN_REMOTE_SUBMIX | \ + AUDIO_DEVICE_OUT_REMOTE_SUBMIX) + +#include +#include + +#include +#include +#include +#include +#include +#include +#include +#include "AudioPolicyManager.h" +#include "audio_policy_conf.h" + +namespace android { + +// ---------------------------------------------------------------------------- +// AudioPolicyInterface implementation +// ---------------------------------------------------------------------------- + +status_t AudioPolicyManager::setDeviceConnectionState(audio_devices_t device, + audio_policy_dev_state_t state, + const char *device_address, + const char *device_name) +{ + return setDeviceConnectionStateInt(device, state, device_address, device_name); +} + +status_t AudioPolicyManager::setDeviceConnectionStateInt(audio_devices_t device, + audio_policy_dev_state_t state, + const char *device_address, + const char *device_name) +{ + ALOGV("setDeviceConnectionStateInt() device: 0x%X, state %d, address %s name %s", +- device, state, device_address, device_name); + + // connect/disconnect only 1 device at a time + if (!audio_is_output_device(device) && !audio_is_input_device(device)) return BAD_VALUE; + + sp devDesc = getDeviceDescriptor(device, device_address, device_name); + + // handle output devices + if (audio_is_output_device(device)) { + SortedVector outputs; + + ssize_t index = mAvailableOutputDevices.indexOf(devDesc); + + // save a copy of the opened output descriptors before any output is opened or closed + // by checkOutputsForDevice(). This will be needed by checkOutputForAllStrategies() + mPreviousOutputs = mOutputs; + switch (state) + { + // handle output device connection + case AUDIO_POLICY_DEVICE_STATE_AVAILABLE: { + if (index >= 0) { + ALOGW("setDeviceConnectionState() device already connected: %x", device); + return INVALID_OPERATION; + } + ALOGV("setDeviceConnectionState() connecting device %x", device); + + // register new device as available + index = mAvailableOutputDevices.add(devDesc); + if (index >= 0) { + sp module = getModuleForDevice(device); + if (module == 0) { + ALOGD("setDeviceConnectionState() could not find HW module for device %08x", + device); + mAvailableOutputDevices.remove(devDesc); + return INVALID_OPERATION; + } + mAvailableOutputDevices[index]->attach(module); + } else { + return NO_MEMORY; + } + + if (checkOutputsForDevice(devDesc, state, outputs, devDesc->mAddress) != NO_ERROR) { + mAvailableOutputDevices.remove(devDesc); + return INVALID_OPERATION; + } + // outputs should never be empty here + ALOG_ASSERT(outputs.size() != 0, "setDeviceConnectionState():" + "checkOutputsForDevice() returned no outputs but status OK"); + ALOGV("setDeviceConnectionState() checkOutputsForDevice() returned %zu outputs", + outputs.size()); + + // Send connect to HALs + AudioParameter param = AudioParameter(devDesc->mAddress); + param.addInt(String8(AUDIO_PARAMETER_DEVICE_CONNECT), device); + mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString()); + + } break; + // handle output device disconnection + case AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE: { + if (index < 0) { + ALOGW("setDeviceConnectionState() device not connected: %x", device); + return INVALID_OPERATION; + } + + ALOGV("setDeviceConnectionState() disconnecting output device %x", device); + + // Send Disconnect to HALs + AudioParameter param = AudioParameter(devDesc->mAddress); + param.addInt(String8(AUDIO_PARAMETER_DEVICE_DISCONNECT), device); + mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString()); + + // remove device from available output devices + mAvailableOutputDevices.remove(devDesc); + + checkOutputsForDevice(devDesc, state, outputs, devDesc->mAddress); + } break; + + default: + ALOGE("setDeviceConnectionState() invalid state: %x", state); + return BAD_VALUE; + } + + // checkA2dpSuspend must run before checkOutputForAllStrategies so that A2DP + // output is suspended before any tracks are moved to it + checkA2dpSuspend(); + checkOutputForAllStrategies(); + // outputs must be closed after checkOutputForAllStrategies() is executed + if (!outputs.isEmpty()) { + for (size_t i = 0; i < outputs.size(); i++) { + sp desc = mOutputs.valueFor(outputs[i]); + // close unused outputs after device disconnection or direct outputs that have been + // opened by checkOutputsForDevice() to query dynamic parameters + if ((state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) || + (((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) && + (desc->mDirectOpenCount == 0))) { + closeOutput(outputs[i]); + } + } + // check again after closing A2DP output to reset mA2dpSuspended if needed + checkA2dpSuspend(); + } + + updateDevicesAndOutputs(); + if (mPhoneState == AUDIO_MODE_IN_CALL) { + audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/); + updateCallRouting(newDevice); + } + for (size_t i = 0; i < mOutputs.size(); i++) { + audio_io_handle_t output = mOutputs.keyAt(i); + if ((mPhoneState != AUDIO_MODE_IN_CALL) || (output != mPrimaryOutput)) { + audio_devices_t newDevice = getNewOutputDevice(mOutputs.keyAt(i), + true /*fromCache*/); + // do not force device change on duplicated output because if device is 0, it will + // also force a device 0 for the two outputs it is duplicated to which may override + // a valid device selection on those outputs. + bool force = !mOutputs.valueAt(i)->isDuplicated() + && (!deviceDistinguishesOnAddress(device) + // always force when disconnecting (a non-duplicated device) + || (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE)); + setOutputDevice(output, newDevice, force, 0); + } + } + + mpClientInterface->onAudioPortListUpdate(); + return NO_ERROR; + } // end if is output device + + // handle input devices + if (audio_is_input_device(device)) { + SortedVector inputs; + + ssize_t index = mAvailableInputDevices.indexOf(devDesc); + switch (state) + { + // handle input device connection + case AUDIO_POLICY_DEVICE_STATE_AVAILABLE: { + if (index >= 0) { + ALOGW("setDeviceConnectionState() device already connected: %d", device); + return INVALID_OPERATION; + } + sp module = getModuleForDevice(device); + if (module == NULL) { + ALOGW("setDeviceConnectionState(): could not find HW module for device %08x", + device); + return INVALID_OPERATION; + } + if (checkInputsForDevice(device, state, inputs, devDesc->mAddress) != NO_ERROR) { + return INVALID_OPERATION; + } + + index = mAvailableInputDevices.add(devDesc); + if (index >= 0) { + mAvailableInputDevices[index]->attach(module); + } else { + return NO_MEMORY; + } + + // Set connect to HALs + AudioParameter param = AudioParameter(devDesc->mAddress); + param.addInt(String8(AUDIO_PARAMETER_DEVICE_CONNECT), device); + mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString()); + + } break; + + // handle input device disconnection + case AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE: { + if (index < 0) { + ALOGW("setDeviceConnectionState() device not connected: %d", device); + return INVALID_OPERATION; + } + + ALOGV("setDeviceConnectionState() disconnecting input device %x", device); + + // Set Disconnect to HALs + AudioParameter param = AudioParameter(devDesc->mAddress); + param.addInt(String8(AUDIO_PARAMETER_DEVICE_DISCONNECT), device); + mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString()); + + checkInputsForDevice(device, state, inputs, devDesc->mAddress); + mAvailableInputDevices.remove(devDesc); + + } break; + + default: + ALOGE("setDeviceConnectionState() invalid state: %x", state); + return BAD_VALUE; + } + + closeAllInputs(); + + if (mPhoneState == AUDIO_MODE_IN_CALL) { + audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/); + updateCallRouting(newDevice); + } + + mpClientInterface->onAudioPortListUpdate(); + return NO_ERROR; + } // end if is input device + + ALOGW("setDeviceConnectionState() invalid device: %x", device); + return BAD_VALUE; +} + +audio_policy_dev_state_t AudioPolicyManager::getDeviceConnectionState(audio_devices_t device, + const char *device_address) +{ + sp devDesc = getDeviceDescriptor(device, device_address, ""); + DeviceVector *deviceVector; + + if (audio_is_output_device(device)) { + deviceVector = &mAvailableOutputDevices; + } else if (audio_is_input_device(device)) { + deviceVector = &mAvailableInputDevices; + } else { + ALOGW("getDeviceConnectionState() invalid device type %08x", device); + return AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE; + } + + ssize_t index = deviceVector->indexOf(devDesc); + if (index >= 0) { + return AUDIO_POLICY_DEVICE_STATE_AVAILABLE; + } else { + return AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE; + } +} + +sp AudioPolicyManager::getDeviceDescriptor(const audio_devices_t device, + const char *device_address, + const char *device_name) +{ + String8 address = (device_address == NULL) ? String8("") : String8(device_address); + // handle legacy remote submix case where the address was not always specified + if (deviceDistinguishesOnAddress(device) && (address.length() == 0)) { + address = String8("0"); + } + + for (size_t i = 0; i < mHwModules.size(); i++) { + if (mHwModules[i]->mHandle == 0) { + continue; + } + DeviceVector deviceList = + mHwModules[i]->mDeclaredDevices.getDevicesFromTypeAddr(device, address); + if (!deviceList.isEmpty()) { + return deviceList.itemAt(0); + } + deviceList = mHwModules[i]->mDeclaredDevices.getDevicesFromType(device); + if (!deviceList.isEmpty()) { + return deviceList.itemAt(0); + } + } + + sp devDesc = + new DeviceDescriptor(String8(device_name != NULL ? device_name : ""), device); + devDesc->mAddress = address; + return devDesc; +} + +void AudioPolicyManager::updateCallRouting(audio_devices_t rxDevice, int delayMs) +{ + bool createTxPatch = false; + struct audio_patch patch; + patch.num_sources = 1; + patch.num_sinks = 1; + status_t status; + audio_patch_handle_t afPatchHandle; + DeviceVector deviceList; + + audio_devices_t txDevice = getDeviceAndMixForInputSource(AUDIO_SOURCE_VOICE_COMMUNICATION); + ALOGV("updateCallRouting device rxDevice %08x txDevice %08x", rxDevice, txDevice); + + // release existing RX patch if any + if (mCallRxPatch != 0) { + mpClientInterface->releaseAudioPatch(mCallRxPatch->mAfPatchHandle, 0); + mCallRxPatch.clear(); + } + // release TX patch if any + if (mCallTxPatch != 0) { + mpClientInterface->releaseAudioPatch(mCallTxPatch->mAfPatchHandle, 0); + mCallTxPatch.clear(); + } + + // If the RX device is on the primary HW module, then use legacy routing method for voice calls + // via setOutputDevice() on primary output. + // Otherwise, create two audio patches for TX and RX path. + if (availablePrimaryOutputDevices() & rxDevice) { + setOutputDevice(mPrimaryOutput, rxDevice, true, delayMs); + // If the TX device is also on the primary HW module, setOutputDevice() will take care + // of it due to legacy implementation. If not, create a patch. + if ((availablePrimaryInputDevices() & txDevice & ~AUDIO_DEVICE_BIT_IN) + == AUDIO_DEVICE_NONE) { + createTxPatch = true; + } + } else { + // create RX path audio patch + deviceList = mAvailableOutputDevices.getDevicesFromType(rxDevice); + ALOG_ASSERT(!deviceList.isEmpty(), + "updateCallRouting() selected device not in output device list"); + sp rxSinkDeviceDesc = deviceList.itemAt(0); + deviceList = mAvailableInputDevices.getDevicesFromType(AUDIO_DEVICE_IN_TELEPHONY_RX); + ALOG_ASSERT(!deviceList.isEmpty(), + "updateCallRouting() no telephony RX device"); + sp rxSourceDeviceDesc = deviceList.itemAt(0); + + rxSourceDeviceDesc->toAudioPortConfig(&patch.sources[0]); + rxSinkDeviceDesc->toAudioPortConfig(&patch.sinks[0]); + + // request to reuse existing output stream if one is already opened to reach the RX device + SortedVector outputs = + getOutputsForDevice(rxDevice, mOutputs); + audio_io_handle_t output = selectOutput(outputs, + AUDIO_OUTPUT_FLAG_NONE, + AUDIO_FORMAT_INVALID); + if (output != AUDIO_IO_HANDLE_NONE) { + sp outputDesc = mOutputs.valueFor(output); + ALOG_ASSERT(!outputDesc->isDuplicated(), + "updateCallRouting() RX device output is duplicated"); + outputDesc->toAudioPortConfig(&patch.sources[1]); + patch.num_sources = 2; + } + + afPatchHandle = AUDIO_PATCH_HANDLE_NONE; + status = mpClientInterface->createAudioPatch(&patch, &afPatchHandle, 0); + ALOGW_IF(status != NO_ERROR, "updateCallRouting() error %d creating RX audio patch", + status); + if (status == NO_ERROR) { + mCallRxPatch = new AudioPatch((audio_patch_handle_t)nextUniqueId(), + &patch, mUidCached); + mCallRxPatch->mAfPatchHandle = afPatchHandle; + mCallRxPatch->mUid = mUidCached; + } + createTxPatch = true; + } + if (createTxPatch) { + + struct audio_patch patch; + patch.num_sources = 1; + patch.num_sinks = 1; + deviceList = mAvailableInputDevices.getDevicesFromType(txDevice); + ALOG_ASSERT(!deviceList.isEmpty(), + "updateCallRouting() selected device not in input device list"); + sp txSourceDeviceDesc = deviceList.itemAt(0); + txSourceDeviceDesc->toAudioPortConfig(&patch.sources[0]); + deviceList = mAvailableOutputDevices.getDevicesFromType(AUDIO_DEVICE_OUT_TELEPHONY_TX); + ALOG_ASSERT(!deviceList.isEmpty(), + "updateCallRouting() no telephony TX device"); + sp txSinkDeviceDesc = deviceList.itemAt(0); + txSinkDeviceDesc->toAudioPortConfig(&patch.sinks[0]); + + SortedVector outputs = + getOutputsForDevice(AUDIO_DEVICE_OUT_TELEPHONY_TX, mOutputs); + audio_io_handle_t output = selectOutput(outputs, + AUDIO_OUTPUT_FLAG_NONE, + AUDIO_FORMAT_INVALID); + // request to reuse existing output stream if one is already opened to reach the TX + // path output device + if (output != AUDIO_IO_HANDLE_NONE) { + sp outputDesc = mOutputs.valueFor(output); + ALOG_ASSERT(!outputDesc->isDuplicated(), + "updateCallRouting() RX device output is duplicated"); + outputDesc->toAudioPortConfig(&patch.sources[1]); + patch.num_sources = 2; + } + + afPatchHandle = AUDIO_PATCH_HANDLE_NONE; + status = mpClientInterface->createAudioPatch(&patch, &afPatchHandle, 0); + ALOGW_IF(status != NO_ERROR, "setPhoneState() error %d creating TX audio patch", + status); + if (status == NO_ERROR) { + mCallTxPatch = new AudioPatch((audio_patch_handle_t)nextUniqueId(), + &patch, mUidCached); + mCallTxPatch->mAfPatchHandle = afPatchHandle; + mCallTxPatch->mUid = mUidCached; + } + } +} + +void AudioPolicyManager::setPhoneState(audio_mode_t state) +{ + ALOGV("setPhoneState() state %d", state); + if (state < 0 || state >= AUDIO_MODE_CNT) { + ALOGW("setPhoneState() invalid state %d", state); + return; + } + + if (state == mPhoneState ) { + ALOGW("setPhoneState() setting same state %d", state); + return; + } + + // if leaving call state, handle special case of active streams + // pertaining to sonification strategy see handleIncallSonification() + if (isInCall()) { + ALOGV("setPhoneState() in call state management: new state is %d", state); + for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) { + if (stream == AUDIO_STREAM_PATCH) { + continue; + } + handleIncallSonification((audio_stream_type_t)stream, false, true); + } + + // force reevaluating accessibility routing when call starts + mpClientInterface->invalidateStream(AUDIO_STREAM_ACCESSIBILITY); + } + + // store previous phone state for management of sonification strategy below + int oldState = mPhoneState; + mPhoneState = state; + bool force = false; + + // are we entering or starting a call + if (!isStateInCall(oldState) && isStateInCall(state)) { + ALOGV(" Entering call in setPhoneState()"); + // force routing command to audio hardware when starting a call + // even if no device change is needed + force = true; + for (int j = 0; j < ApmGains::DEVICE_CATEGORY_CNT; j++) { + mStreams[AUDIO_STREAM_DTMF].mVolumeCurve[j] = + ApmGains::sVolumeProfiles[AUDIO_STREAM_VOICE_CALL][j]; + } + } else if (isStateInCall(oldState) && !isStateInCall(state)) { + ALOGV(" Exiting call in setPhoneState()"); + // force routing command to audio hardware when exiting a call + // even if no device change is needed + force = true; + for (int j = 0; j < ApmGains::DEVICE_CATEGORY_CNT; j++) { + mStreams[AUDIO_STREAM_DTMF].mVolumeCurve[j] = + ApmGains::sVolumeProfiles[AUDIO_STREAM_DTMF][j]; + } + } else if (isStateInCall(state) && (state != oldState)) { + ALOGV(" Switching between telephony and VoIP in setPhoneState()"); + // force routing command to audio hardware when switching between telephony and VoIP + // even if no device change is needed + force = true; + } + + // check for device and output changes triggered by new phone state + checkA2dpSuspend(); + checkOutputForAllStrategies(); + updateDevicesAndOutputs(); + + sp hwOutputDesc = mOutputs.valueFor(mPrimaryOutput); + + int delayMs = 0; + if (isStateInCall(state)) { + nsecs_t sysTime = systemTime(); + for (size_t i = 0; i < mOutputs.size(); i++) { + sp desc = mOutputs.valueAt(i); + // mute media and sonification strategies and delay device switch by the largest + // latency of any output where either strategy is active. + // This avoid sending the ring tone or music tail into the earpiece or headset. + if ((desc->isStrategyActive(STRATEGY_MEDIA, + SONIFICATION_HEADSET_MUSIC_DELAY, + sysTime) || + desc->isStrategyActive(STRATEGY_SONIFICATION, + SONIFICATION_HEADSET_MUSIC_DELAY, + sysTime)) && + (delayMs < (int)desc->mLatency*2)) { + delayMs = desc->mLatency*2; + } + setStrategyMute(STRATEGY_MEDIA, true, mOutputs.keyAt(i)); + setStrategyMute(STRATEGY_MEDIA, false, mOutputs.keyAt(i), MUTE_TIME_MS, + getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/)); + setStrategyMute(STRATEGY_SONIFICATION, true, mOutputs.keyAt(i)); + setStrategyMute(STRATEGY_SONIFICATION, false, mOutputs.keyAt(i), MUTE_TIME_MS, + getDeviceForStrategy(STRATEGY_SONIFICATION, true /*fromCache*/)); + } + } + + // Note that despite the fact that getNewOutputDevice() is called on the primary output, + // the device returned is not necessarily reachable via this output + audio_devices_t rxDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/); + // force routing command to audio hardware when ending call + // even if no device change is needed + if (isStateInCall(oldState) && rxDevice == AUDIO_DEVICE_NONE) { + rxDevice = hwOutputDesc->device(); + } + + if (state == AUDIO_MODE_IN_CALL) { + updateCallRouting(rxDevice, delayMs); + } else if (oldState == AUDIO_MODE_IN_CALL) { + if (mCallRxPatch != 0) { + mpClientInterface->releaseAudioPatch(mCallRxPatch->mAfPatchHandle, 0); + mCallRxPatch.clear(); + } + if (mCallTxPatch != 0) { + mpClientInterface->releaseAudioPatch(mCallTxPatch->mAfPatchHandle, 0); + mCallTxPatch.clear(); + } + setOutputDevice(mPrimaryOutput, rxDevice, force, 0); + } else { + setOutputDevice(mPrimaryOutput, rxDevice, force, 0); + } + // if entering in call state, handle special case of active streams + // pertaining to sonification strategy see handleIncallSonification() + if (isStateInCall(state)) { + ALOGV("setPhoneState() in call state management: new state is %d", state); + for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) { + if (stream == AUDIO_STREAM_PATCH) { + continue; + } + handleIncallSonification((audio_stream_type_t)stream, true, true); + } + } + + // Flag that ringtone volume must be limited to music volume until we exit MODE_RINGTONE + if (state == AUDIO_MODE_RINGTONE && + isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY)) { + mLimitRingtoneVolume = true; + } else { + mLimitRingtoneVolume = false; + } +} + +void AudioPolicyManager::setForceUse(audio_policy_force_use_t usage, + audio_policy_forced_cfg_t config) +{ + ALOGV("setForceUse() usage %d, config %d, mPhoneState %d", usage, config, mPhoneState); + + bool forceVolumeReeval = false; + switch(usage) { + case AUDIO_POLICY_FORCE_FOR_COMMUNICATION: + if (config != AUDIO_POLICY_FORCE_SPEAKER && config != AUDIO_POLICY_FORCE_BT_SCO && + config != AUDIO_POLICY_FORCE_NONE) { + ALOGW("setForceUse() invalid config %d for FOR_COMMUNICATION", config); + return; + } + forceVolumeReeval = true; + mForceUse[usage] = config; + break; + case AUDIO_POLICY_FORCE_FOR_MEDIA: + if (config != AUDIO_POLICY_FORCE_HEADPHONES && config != AUDIO_POLICY_FORCE_BT_A2DP && + config != AUDIO_POLICY_FORCE_WIRED_ACCESSORY && + config != AUDIO_POLICY_FORCE_ANALOG_DOCK && + config != AUDIO_POLICY_FORCE_DIGITAL_DOCK && config != AUDIO_POLICY_FORCE_NONE && + config != AUDIO_POLICY_FORCE_NO_BT_A2DP && config != AUDIO_POLICY_FORCE_SPEAKER ) { + ALOGW("setForceUse() invalid config %d for FOR_MEDIA", config); + return; + } + mForceUse[usage] = config; + break; + case AUDIO_POLICY_FORCE_FOR_RECORD: + if (config != AUDIO_POLICY_FORCE_BT_SCO && config != AUDIO_POLICY_FORCE_WIRED_ACCESSORY && + config != AUDIO_POLICY_FORCE_NONE) { + ALOGW("setForceUse() invalid config %d for FOR_RECORD", config); + return; + } + mForceUse[usage] = config; + break; + case AUDIO_POLICY_FORCE_FOR_DOCK: + if (config != AUDIO_POLICY_FORCE_NONE && config != AUDIO_POLICY_FORCE_BT_CAR_DOCK && + config != AUDIO_POLICY_FORCE_BT_DESK_DOCK && + config != AUDIO_POLICY_FORCE_WIRED_ACCESSORY && + config != AUDIO_POLICY_FORCE_ANALOG_DOCK && + config != AUDIO_POLICY_FORCE_DIGITAL_DOCK) { + ALOGW("setForceUse() invalid config %d for FOR_DOCK", config); + } + forceVolumeReeval = true; + mForceUse[usage] = config; + break; + case AUDIO_POLICY_FORCE_FOR_SYSTEM: + if (config != AUDIO_POLICY_FORCE_NONE && + config != AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) { + ALOGW("setForceUse() invalid config %d for FOR_SYSTEM", config); + } + forceVolumeReeval = true; + mForceUse[usage] = config; + break; + case AUDIO_POLICY_FORCE_FOR_HDMI_SYSTEM_AUDIO: + if (config != AUDIO_POLICY_FORCE_NONE && + config != AUDIO_POLICY_FORCE_HDMI_SYSTEM_AUDIO_ENFORCED) { + ALOGW("setForceUse() invalid config %d forHDMI_SYSTEM_AUDIO", config); + } + mForceUse[usage] = config; + break; + default: + ALOGW("setForceUse() invalid usage %d", usage); + break; + } + + // check for device and output changes triggered by new force usage + checkA2dpSuspend(); + checkOutputForAllStrategies(); + updateDevicesAndOutputs(); + if (mPhoneState == AUDIO_MODE_IN_CALL) { + audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, true /*fromCache*/); + updateCallRouting(newDevice); + } + for (size_t i = 0; i < mOutputs.size(); i++) { + audio_io_handle_t output = mOutputs.keyAt(i); + audio_devices_t newDevice = getNewOutputDevice(output, true /*fromCache*/); + if ((mPhoneState != AUDIO_MODE_IN_CALL) || (output != mPrimaryOutput)) { + setOutputDevice(output, newDevice, (newDevice != AUDIO_DEVICE_NONE)); + } + if (forceVolumeReeval && (newDevice != AUDIO_DEVICE_NONE)) { + applyStreamVolumes(output, newDevice, 0, true); + } + } + + audio_io_handle_t activeInput = getActiveInput(); + if (activeInput != 0) { + setInputDevice(activeInput, getNewInputDevice(activeInput)); + } + +} + +audio_policy_forced_cfg_t AudioPolicyManager::getForceUse(audio_policy_force_use_t usage) +{ + return mForceUse[usage]; +} + +void AudioPolicyManager::setSystemProperty(const char* property, const char* value) +{ + ALOGV("setSystemProperty() property %s, value %s", property, value); +} + +// Find a direct output profile compatible with the parameters passed, even if the input flags do +// not explicitly request a direct output +sp AudioPolicyManager::getProfileForDirectOutput( + audio_devices_t device, + uint32_t samplingRate, + audio_format_t format, + audio_channel_mask_t channelMask, + audio_output_flags_t flags) +{ + for (size_t i = 0; i < mHwModules.size(); i++) { + if (mHwModules[i]->mHandle == 0) { + continue; + } + for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++) { + sp profile = mHwModules[i]->mOutputProfiles[j]; + bool found = profile->isCompatibleProfile(device, String8(""), samplingRate, + NULL /*updatedSamplingRate*/, format, channelMask, + flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD ? + AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD : AUDIO_OUTPUT_FLAG_DIRECT); + if (found && (mAvailableOutputDevices.types() & profile->mSupportedDevices.types())) { + return profile; + } + } + } + return 0; +} + +audio_io_handle_t AudioPolicyManager::getOutput(audio_stream_type_t stream, + uint32_t samplingRate, + audio_format_t format, + audio_channel_mask_t channelMask, + audio_output_flags_t flags, + const audio_offload_info_t *offloadInfo) +{ + routing_strategy strategy = getStrategy(stream); + audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/); + ALOGV("getOutput() device %d, stream %d, samplingRate %d, format %x, channelMask %x, flags %x", + device, stream, samplingRate, format, channelMask, flags); + + return getOutputForDevice(device, AUDIO_SESSION_ALLOCATE, + stream, samplingRate,format, channelMask, + flags, offloadInfo); +} + +status_t AudioPolicyManager::getOutputForAttr(const audio_attributes_t *attr, + audio_io_handle_t *output, + audio_session_t session, + audio_stream_type_t *stream, + uint32_t samplingRate, + audio_format_t format, + audio_channel_mask_t channelMask, + audio_output_flags_t flags, + const audio_offload_info_t *offloadInfo) +{ + audio_attributes_t attributes; + if (attr != NULL) { + if (!isValidAttributes(attr)) { + ALOGE("getOutputForAttr() invalid attributes: usage=%d content=%d flags=0x%x tags=[%s]", + attr->usage, attr->content_type, attr->flags, + attr->tags); + return BAD_VALUE; + } + attributes = *attr; + } else { + if (*stream < AUDIO_STREAM_MIN || *stream >= AUDIO_STREAM_PUBLIC_CNT) { + ALOGE("getOutputForAttr(): invalid stream type"); + return BAD_VALUE; + } + stream_type_to_audio_attributes(*stream, &attributes); + } + + for (size_t i = 0; i < mPolicyMixes.size(); i++) { + sp desc; + if (mPolicyMixes[i]->mMix.mMixType == MIX_TYPE_PLAYERS) { + for (size_t j = 0; j < mPolicyMixes[i]->mMix.mCriteria.size(); j++) { + if ((RULE_MATCH_ATTRIBUTE_USAGE == mPolicyMixes[i]->mMix.mCriteria[j].mRule && + mPolicyMixes[i]->mMix.mCriteria[j].mAttr.mUsage == attributes.usage) || + (RULE_EXCLUDE_ATTRIBUTE_USAGE == mPolicyMixes[i]->mMix.mCriteria[j].mRule && + mPolicyMixes[i]->mMix.mCriteria[j].mAttr.mUsage != attributes.usage)) { + desc = mPolicyMixes[i]->mOutput; + break; + } + if (strncmp(attributes.tags, "addr=", strlen("addr=")) == 0 && + strncmp(attributes.tags + strlen("addr="), + mPolicyMixes[i]->mMix.mRegistrationId.string(), + AUDIO_ATTRIBUTES_TAGS_MAX_SIZE - strlen("addr=") - 1) == 0) { + desc = mPolicyMixes[i]->mOutput; + break; + } + } + } else if (mPolicyMixes[i]->mMix.mMixType == MIX_TYPE_RECORDERS) { + if (attributes.usage == AUDIO_USAGE_VIRTUAL_SOURCE && + strncmp(attributes.tags, "addr=", strlen("addr=")) == 0 && + strncmp(attributes.tags + strlen("addr="), + mPolicyMixes[i]->mMix.mRegistrationId.string(), + AUDIO_ATTRIBUTES_TAGS_MAX_SIZE - strlen("addr=") - 1) == 0) { + desc = mPolicyMixes[i]->mOutput; + } + } + if (desc != 0) { + if (!audio_is_linear_pcm(format)) { + return BAD_VALUE; + } + desc->mPolicyMix = &mPolicyMixes[i]->mMix; + *stream = streamTypefromAttributesInt(&attributes); + *output = desc->mIoHandle; + ALOGV("getOutputForAttr() returns output %d", *output); + return NO_ERROR; + } + } + if (attributes.usage == AUDIO_USAGE_VIRTUAL_SOURCE) { + ALOGW("getOutputForAttr() no policy mix found for usage AUDIO_USAGE_VIRTUAL_SOURCE"); + return BAD_VALUE; + } + + ALOGV("getOutputForAttr() usage=%d, content=%d, tag=%s flags=%08x", + attributes.usage, attributes.content_type, attributes.tags, attributes.flags); + + routing_strategy strategy = (routing_strategy) getStrategyForAttr(&attributes); + audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/); + + if ((attributes.flags & AUDIO_FLAG_HW_AV_SYNC) != 0) { + flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_HW_AV_SYNC); + } + + ALOGV("getOutputForAttr() device 0x%x, samplingRate %d, format %x, channelMask %x, flags %x", + device, samplingRate, format, channelMask, flags); + + *stream = streamTypefromAttributesInt(&attributes); + *output = getOutputForDevice(device, session, *stream, + samplingRate, format, channelMask, + flags, offloadInfo); + if (*output == AUDIO_IO_HANDLE_NONE) { + return INVALID_OPERATION; + } + return NO_ERROR; +} + +audio_io_handle_t AudioPolicyManager::getOutputForDevice( + audio_devices_t device, + audio_session_t session __unused, + audio_stream_type_t stream, + uint32_t samplingRate, + audio_format_t format, + audio_channel_mask_t channelMask, + audio_output_flags_t flags, + const audio_offload_info_t *offloadInfo) +{ + audio_io_handle_t output = AUDIO_IO_HANDLE_NONE; + uint32_t latency = 0; + status_t status; + +#ifdef AUDIO_POLICY_TEST + if (mCurOutput != 0) { + ALOGV("getOutput() test output mCurOutput %d, samplingRate %d, format %d, channelMask %x, mDirectOutput %d", + mCurOutput, mTestSamplingRate, mTestFormat, mTestChannels, mDirectOutput); + + if (mTestOutputs[mCurOutput] == 0) { + ALOGV("getOutput() opening test output"); + sp outputDesc = new AudioOutputDescriptor(NULL); + outputDesc->mDevice = mTestDevice; + outputDesc->mLatency = mTestLatencyMs; + outputDesc->mFlags = + (audio_output_flags_t)(mDirectOutput ? AUDIO_OUTPUT_FLAG_DIRECT : 0); + outputDesc->mRefCount[stream] = 0; + audio_config_t config = AUDIO_CONFIG_INITIALIZER; + config.sample_rate = mTestSamplingRate; + config.channel_mask = mTestChannels; + config.format = mTestFormat; + if (offloadInfo != NULL) { + config.offload_info = *offloadInfo; + } + status = mpClientInterface->openOutput(0, + &mTestOutputs[mCurOutput], + &config, + &outputDesc->mDevice, + String8(""), + &outputDesc->mLatency, + outputDesc->mFlags); + if (status == NO_ERROR) { + outputDesc->mSamplingRate = config.sample_rate; + outputDesc->mFormat = config.format; + outputDesc->mChannelMask = config.channel_mask; + AudioParameter outputCmd = AudioParameter(); + outputCmd.addInt(String8("set_id"),mCurOutput); + mpClientInterface->setParameters(mTestOutputs[mCurOutput],outputCmd.toString()); + addOutput(mTestOutputs[mCurOutput], outputDesc); + } + } + return mTestOutputs[mCurOutput]; + } +#endif //AUDIO_POLICY_TEST + + // open a direct output if required by specified parameters + //force direct flag if offload flag is set: offloading implies a direct output stream + // and all common behaviors are driven by checking only the direct flag + // this should normally be set appropriately in the policy configuration file + if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) { + flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT); + } + if ((flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) != 0) { + flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT); + } + // only allow deep buffering for music stream type + if (stream != AUDIO_STREAM_MUSIC) { + flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER); + } + + sp profile; + + // skip direct output selection if the request can obviously be attached to a mixed output + // and not explicitly requested + if (((flags & AUDIO_OUTPUT_FLAG_DIRECT) == 0) && + audio_is_linear_pcm(format) && samplingRate <= MAX_MIXER_SAMPLING_RATE && + audio_channel_count_from_out_mask(channelMask) <= 2) { + goto non_direct_output; + } + + // Do not allow offloading if one non offloadable effect is enabled. This prevents from + // creating an offloaded track and tearing it down immediately after start when audioflinger + // detects there is an active non offloadable effect. + // FIXME: We should check the audio session here but we do not have it in this context. + // This may prevent offloading in rare situations where effects are left active by apps + // in the background. + + if (((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) == 0) || + !isNonOffloadableEffectEnabled()) { + profile = getProfileForDirectOutput(device, + samplingRate, + format, + channelMask, + (audio_output_flags_t)flags); + } + + if (profile != 0) { + sp outputDesc = NULL; + + for (size_t i = 0; i < mOutputs.size(); i++) { + sp desc = mOutputs.valueAt(i); + if (!desc->isDuplicated() && (profile == desc->mProfile)) { + outputDesc = desc; + // reuse direct output if currently open and configured with same parameters + if ((samplingRate == outputDesc->mSamplingRate) && + (format == outputDesc->mFormat) && + (channelMask == outputDesc->mChannelMask)) { + outputDesc->mDirectOpenCount++; + ALOGV("getOutput() reusing direct output %d", mOutputs.keyAt(i)); + return mOutputs.keyAt(i); + } + } + } + // close direct output if currently open and configured with different parameters + if (outputDesc != NULL) { + closeOutput(outputDesc->mIoHandle); + } + outputDesc = new AudioOutputDescriptor(profile); + outputDesc->mDevice = device; + outputDesc->mLatency = 0; + outputDesc->mFlags =(audio_output_flags_t) (outputDesc->mFlags | flags); + audio_config_t config = AUDIO_CONFIG_INITIALIZER; + config.sample_rate = samplingRate; + config.channel_mask = channelMask; + config.format = format; + if (offloadInfo != NULL) { + config.offload_info = *offloadInfo; + } + status = mpClientInterface->openOutput(profile->mModule->mHandle, + &output, + &config, + &outputDesc->mDevice, + String8(""), + &outputDesc->mLatency, + outputDesc->mFlags); + + // only accept an output with the requested parameters + if (status != NO_ERROR || + (samplingRate != 0 && samplingRate != config.sample_rate) || + (format != AUDIO_FORMAT_DEFAULT && format != config.format) || + (channelMask != 0 && channelMask != config.channel_mask)) { + ALOGV("getOutput() failed opening direct output: output %d samplingRate %d %d," + "format %d %d, channelMask %04x %04x", output, samplingRate, + outputDesc->mSamplingRate, format, outputDesc->mFormat, channelMask, + outputDesc->mChannelMask); + if (output != AUDIO_IO_HANDLE_NONE) { + mpClientInterface->closeOutput(output); + } + // fall back to mixer output if possible when the direct output could not be open + if (audio_is_linear_pcm(format) && samplingRate <= MAX_MIXER_SAMPLING_RATE) { + goto non_direct_output; + } + // fall back to mixer output if possible when the direct output could not be open + if (audio_is_linear_pcm(format) && samplingRate <= MAX_MIXER_SAMPLING_RATE) { + goto non_direct_output; + } + return AUDIO_IO_HANDLE_NONE; + } + outputDesc->mSamplingRate = config.sample_rate; + outputDesc->mChannelMask = config.channel_mask; + outputDesc->mFormat = config.format; + outputDesc->mRefCount[stream] = 0; + outputDesc->mStopTime[stream] = 0; + outputDesc->mDirectOpenCount = 1; + + audio_io_handle_t srcOutput = getOutputForEffect(); + addOutput(output, outputDesc); + audio_io_handle_t dstOutput = getOutputForEffect(); + if (dstOutput == output) { + mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, srcOutput, dstOutput); + } + mPreviousOutputs = mOutputs; + ALOGV("getOutput() returns new direct output %d", output); + mpClientInterface->onAudioPortListUpdate(); + return output; + } + +non_direct_output: + + // ignoring channel mask due to downmix capability in mixer + + // open a non direct output + + // for non direct outputs, only PCM is supported + if (audio_is_linear_pcm(format)) { + // get which output is suitable for the specified stream. The actual + // routing change will happen when startOutput() will be called + SortedVector outputs = getOutputsForDevice(device, mOutputs); + + // at this stage we should ignore the DIRECT flag as no direct output could be found earlier + flags = (audio_output_flags_t)(flags & ~AUDIO_OUTPUT_FLAG_DIRECT); + output = selectOutput(outputs, flags, format); + } + ALOGW_IF((output == 0), "getOutput() could not find output for stream %d, samplingRate %d," + "format %d, channels %x, flags %x", stream, samplingRate, format, channelMask, flags); + + ALOGV("getOutput() returns output %d", output); + + return output; +} + +audio_io_handle_t AudioPolicyManager::selectOutput(const SortedVector& outputs, + audio_output_flags_t flags, + audio_format_t format) +{ + // select one output among several that provide a path to a particular device or set of + // devices (the list was previously build by getOutputsForDevice()). + // The priority is as follows: + // 1: the output with the highest number of requested policy flags + // 2: the primary output + // 3: the first output in the list + + if (outputs.size() == 0) { + return 0; + } + if (outputs.size() == 1) { + return outputs[0]; + } + + int maxCommonFlags = 0; + audio_io_handle_t outputFlags = 0; + audio_io_handle_t outputPrimary = 0; + + for (size_t i = 0; i < outputs.size(); i++) { + sp outputDesc = mOutputs.valueFor(outputs[i]); + if (!outputDesc->isDuplicated()) { + // if a valid format is specified, skip output if not compatible + if (format != AUDIO_FORMAT_INVALID) { + if (outputDesc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) { + if (format != outputDesc->mFormat) { + continue; + } + } else if (!audio_is_linear_pcm(format)) { + continue; + } + } + + int commonFlags = popcount(outputDesc->mProfile->mFlags & flags); + if (commonFlags > maxCommonFlags) { + outputFlags = outputs[i]; + maxCommonFlags = commonFlags; + ALOGV("selectOutput() commonFlags for output %d, %04x", outputs[i], commonFlags); + } + if (outputDesc->mProfile->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY) { + outputPrimary = outputs[i]; + } + } + } + + if (outputFlags != 0) { + return outputFlags; + } + if (outputPrimary != 0) { + return outputPrimary; + } + + return outputs[0]; +} + +status_t AudioPolicyManager::startOutput(audio_io_handle_t output, + audio_stream_type_t stream, + audio_session_t session) +{ + ALOGV("startOutput() output %d, stream %d, session %d", output, stream, session); + ssize_t index = mOutputs.indexOfKey(output); + if (index < 0) { + ALOGW("startOutput() unknown output %d", output); + return BAD_VALUE; + } + + // cannot start playback of STREAM_TTS if any other output is being used + uint32_t beaconMuteLatency = 0; + if (stream == AUDIO_STREAM_TTS) { + ALOGV("\t found BEACON stream"); + if (isAnyOutputActive(AUDIO_STREAM_TTS /*streamToIgnore*/)) { + return INVALID_OPERATION; + } else { + beaconMuteLatency = handleEventForBeacon(STARTING_BEACON); + } + } else { + // some playback other than beacon starts + beaconMuteLatency = handleEventForBeacon(STARTING_OUTPUT); + } + + sp outputDesc = mOutputs.valueAt(index); + + // increment usage count for this stream on the requested output: + // NOTE that the usage count is the same for duplicated output and hardware output which is + // necessary for a correct control of hardware output routing by startOutput() and stopOutput() + outputDesc->changeRefCount(stream, 1); + + if (outputDesc->mRefCount[stream] == 1) { + // starting an output being rerouted? + audio_devices_t newDevice; + if (outputDesc->mPolicyMix != NULL) { + newDevice = AUDIO_DEVICE_OUT_REMOTE_SUBMIX; + } else { + newDevice = getNewOutputDevice(output, false /*fromCache*/); + } + routing_strategy strategy = getStrategy(stream); + bool shouldWait = (strategy == STRATEGY_SONIFICATION) || + (strategy == STRATEGY_SONIFICATION_RESPECTFUL) || + (beaconMuteLatency > 0); + uint32_t waitMs = beaconMuteLatency; + bool force = false; + for (size_t i = 0; i < mOutputs.size(); i++) { + sp desc = mOutputs.valueAt(i); + if (desc != outputDesc) { + // force a device change if any other output is managed by the same hw + // module and has a current device selection that differs from selected device. + // In this case, the audio HAL must receive the new device selection so that it can + // change the device currently selected by the other active output. + if (outputDesc->sharesHwModuleWith(desc) && + desc->device() != newDevice) { + force = true; + } + // wait for audio on other active outputs to be presented when starting + // a notification so that audio focus effect can propagate, or that a mute/unmute + // event occurred for beacon + uint32_t latency = desc->latency(); + if (shouldWait && desc->isActive(latency * 2) && (waitMs < latency)) { + waitMs = latency; + } + } + } + uint32_t muteWaitMs = setOutputDevice(output, newDevice, force); + + // handle special case for sonification while in call + if (isInCall()) { + handleIncallSonification(stream, true, false); + } + + // apply volume rules for current stream and device if necessary + checkAndSetVolume(stream, + mStreams[stream].getVolumeIndex(newDevice), + output, + newDevice); + + // update the outputs if starting an output with a stream that can affect notification + // routing + handleNotificationRoutingForStream(stream); + + // Automatically enable the remote submix input when output is started on a re routing mix + // of type MIX_TYPE_RECORDERS + if (audio_is_remote_submix_device(newDevice) && outputDesc->mPolicyMix != NULL && + outputDesc->mPolicyMix->mMixType == MIX_TYPE_RECORDERS) { + setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX, + AUDIO_POLICY_DEVICE_STATE_AVAILABLE, + outputDesc->mPolicyMix->mRegistrationId, + "remote-submix"); + } + + // force reevaluating accessibility routing when ringtone or alarm starts + if (strategy == STRATEGY_SONIFICATION) { + mpClientInterface->invalidateStream(AUDIO_STREAM_ACCESSIBILITY); + } + + if (waitMs > muteWaitMs) { + usleep((waitMs - muteWaitMs) * 2 * 1000); + } + } + return NO_ERROR; +} + + +status_t AudioPolicyManager::stopOutput(audio_io_handle_t output, + audio_stream_type_t stream, + audio_session_t session) +{ + ALOGV("stopOutput() output %d, stream %d, session %d", output, stream, session); + ssize_t index = mOutputs.indexOfKey(output); + if (index < 0) { + ALOGW("stopOutput() unknown output %d", output); + return BAD_VALUE; + } + + sp outputDesc = mOutputs.valueAt(index); + + // always handle stream stop, check which stream type is stopping + handleEventForBeacon(stream == AUDIO_STREAM_TTS ? STOPPING_BEACON : STOPPING_OUTPUT); + + // handle special case for sonification while in call + if (isInCall()) { + handleIncallSonification(stream, false, false); + } + + if (outputDesc->mRefCount[stream] > 0) { + // decrement usage count of this stream on the output + outputDesc->changeRefCount(stream, -1); + // store time at which the stream was stopped - see isStreamActive() + if (outputDesc->mRefCount[stream] == 0) { + // Automatically disable the remote submix input when output is stopped on a + // re routing mix of type MIX_TYPE_RECORDERS + if (audio_is_remote_submix_device(outputDesc->mDevice) && + outputDesc->mPolicyMix != NULL && + outputDesc->mPolicyMix->mMixType == MIX_TYPE_RECORDERS) { + setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX, + AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE, + outputDesc->mPolicyMix->mRegistrationId, + "remote-submix"); + } + + outputDesc->mStopTime[stream] = systemTime(); + audio_devices_t newDevice = getNewOutputDevice(output, false /*fromCache*/); + // delay the device switch by twice the latency because stopOutput() is executed when + // the track stop() command is received and at that time the audio track buffer can + // still contain data that needs to be drained. The latency only covers the audio HAL + // and kernel buffers. Also the latency does not always include additional delay in the + // audio path (audio DSP, CODEC ...) + setOutputDevice(output, newDevice, false, outputDesc->mLatency*2); + + // force restoring the device selection on other active outputs if it differs from the + // one being selected for this output + for (size_t i = 0; i < mOutputs.size(); i++) { + audio_io_handle_t curOutput = mOutputs.keyAt(i); + sp desc = mOutputs.valueAt(i); + if (curOutput != output && + desc->isActive() && + outputDesc->sharesHwModuleWith(desc) && + (newDevice != desc->device())) { + setOutputDevice(curOutput, + getNewOutputDevice(curOutput, false /*fromCache*/), + true, + outputDesc->mLatency*2); + } + } + // update the outputs if stopping one with a stream that can affect notification routing + handleNotificationRoutingForStream(stream); + } + return NO_ERROR; + } else { + ALOGW("stopOutput() refcount is already 0 for output %d", output); + return INVALID_OPERATION; + } +} + +void AudioPolicyManager::releaseOutput(audio_io_handle_t output, + audio_stream_type_t stream __unused, + audio_session_t session __unused) +{ + ALOGV("releaseOutput() %d", output); + ssize_t index = mOutputs.indexOfKey(output); + if (index < 0) { + ALOGW("releaseOutput() releasing unknown output %d", output); + return; + } + +#ifdef AUDIO_POLICY_TEST + int testIndex = testOutputIndex(output); + if (testIndex != 0) { + sp outputDesc = mOutputs.valueAt(index); + if (outputDesc->isActive()) { + mpClientInterface->closeOutput(output); + mOutputs.removeItem(output); + mTestOutputs[testIndex] = 0; + } + return; + } +#endif //AUDIO_POLICY_TEST + + sp desc = mOutputs.valueAt(index); + if (desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) { + if (desc->mDirectOpenCount <= 0) { + ALOGW("releaseOutput() invalid open count %d for output %d", + desc->mDirectOpenCount, output); + return; + } + if (--desc->mDirectOpenCount == 0) { + closeOutput(output); + // If effects where present on the output, audioflinger moved them to the primary + // output by default: move them back to the appropriate output. + audio_io_handle_t dstOutput = getOutputForEffect(); + if (dstOutput != mPrimaryOutput) { + mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, mPrimaryOutput, dstOutput); + } + mpClientInterface->onAudioPortListUpdate(); + } + } +} + + +status_t AudioPolicyManager::getInputForAttr(const audio_attributes_t *attr, + audio_io_handle_t *input, + audio_session_t session, + uint32_t samplingRate, + audio_format_t format, + audio_channel_mask_t channelMask, + audio_input_flags_t flags, + input_type_t *inputType) +{ + ALOGV("getInputForAttr() source %d, samplingRate %d, format %d, channelMask %x," + "session %d, flags %#x", + attr->source, samplingRate, format, channelMask, session, flags); + + *input = AUDIO_IO_HANDLE_NONE; + *inputType = API_INPUT_INVALID; + audio_devices_t device; + // handle legacy remote submix case where the address was not always specified + String8 address = String8(""); + bool isSoundTrigger = false; + audio_source_t inputSource = attr->source; + audio_source_t halInputSource; + AudioMix *policyMix = NULL; + + if (inputSource == AUDIO_SOURCE_DEFAULT) { + inputSource = AUDIO_SOURCE_MIC; + } + halInputSource = inputSource; + + if (inputSource == AUDIO_SOURCE_REMOTE_SUBMIX && + strncmp(attr->tags, "addr=", strlen("addr=")) == 0) { + device = AUDIO_DEVICE_IN_REMOTE_SUBMIX; + address = String8(attr->tags + strlen("addr=")); + ssize_t index = mPolicyMixes.indexOfKey(address); + if (index < 0) { + ALOGW("getInputForAttr() no policy for address %s", address.string()); + return BAD_VALUE; + } + if (mPolicyMixes[index]->mMix.mMixType != MIX_TYPE_PLAYERS) { + ALOGW("getInputForAttr() bad policy mix type for address %s", address.string()); + return BAD_VALUE; + } + policyMix = &mPolicyMixes[index]->mMix; + *inputType = API_INPUT_MIX_EXT_POLICY_REROUTE; + } else { + device = getDeviceAndMixForInputSource(inputSource, &policyMix); + if (device == AUDIO_DEVICE_NONE) { + ALOGW("getInputForAttr() could not find device for source %d", inputSource); + return BAD_VALUE; + } + if (policyMix != NULL) { + address = policyMix->mRegistrationId; + if (policyMix->mMixType == MIX_TYPE_RECORDERS) { + // there is an external policy, but this input is attached to a mix of recorders, + // meaning it receives audio injected into the framework, so the recorder doesn't + // know about it and is therefore considered "legacy" + *inputType = API_INPUT_LEGACY; + } else { + // recording a mix of players defined by an external policy, we're rerouting for + // an external policy + *inputType = API_INPUT_MIX_EXT_POLICY_REROUTE; + } + } else if (audio_is_remote_submix_device(device)) { + address = String8("0"); + *inputType = API_INPUT_MIX_CAPTURE; + } else { + *inputType = API_INPUT_LEGACY; + } + // adapt channel selection to input source + switch (inputSource) { + case AUDIO_SOURCE_VOICE_UPLINK: + channelMask = AUDIO_CHANNEL_IN_VOICE_UPLINK; + break; + case AUDIO_SOURCE_VOICE_DOWNLINK: + channelMask = AUDIO_CHANNEL_IN_VOICE_DNLINK; + break; + case AUDIO_SOURCE_VOICE_CALL: + channelMask = AUDIO_CHANNEL_IN_VOICE_UPLINK | AUDIO_CHANNEL_IN_VOICE_DNLINK; + break; + default: + break; + } + if (inputSource == AUDIO_SOURCE_HOTWORD) { + ssize_t index = mSoundTriggerSessions.indexOfKey(session); + if (index >= 0) { + *input = mSoundTriggerSessions.valueFor(session); + isSoundTrigger = true; + flags = (audio_input_flags_t)(flags | AUDIO_INPUT_FLAG_HW_HOTWORD); + ALOGV("SoundTrigger capture on session %d input %d", session, *input); + } else { + halInputSource = AUDIO_SOURCE_VOICE_RECOGNITION; + } + } + } + + sp profile = getInputProfile(device, address, + samplingRate, format, channelMask, + flags); + if (profile == 0) { + //retry without flags + audio_input_flags_t log_flags = flags; + flags = AUDIO_INPUT_FLAG_NONE; + profile = getInputProfile(device, address, + samplingRate, format, channelMask, + flags); + if (profile == 0) { + ALOGW("getInputForAttr() could not find profile for device 0x%X, samplingRate %u," + "format %#x, channelMask 0x%X, flags %#x", + device, samplingRate, format, channelMask, log_flags); + return BAD_VALUE; + } + } + + if (profile->mModule->mHandle == 0) { + ALOGE("getInputForAttr(): HW module %s not opened", profile->mModule->mName); + return NO_INIT; + } + + audio_config_t config = AUDIO_CONFIG_INITIALIZER; + config.sample_rate = samplingRate; + config.channel_mask = channelMask; + config.format = format; + + status_t status = mpClientInterface->openInput(profile->mModule->mHandle, + input, + &config, + &device, + address, + halInputSource, + flags); + + // only accept input with the exact requested set of parameters + if (status != NO_ERROR || *input == AUDIO_IO_HANDLE_NONE || + (samplingRate != config.sample_rate) || + (format != config.format) || + (channelMask != config.channel_mask)) { + ALOGW("getInputForAttr() failed opening input: samplingRate %d, format %d, channelMask %x", + samplingRate, format, channelMask); + if (*input != AUDIO_IO_HANDLE_NONE) { + mpClientInterface->closeInput(*input); + } + return BAD_VALUE; + } + + sp inputDesc = new AudioInputDescriptor(profile); + inputDesc->mInputSource = inputSource; + inputDesc->mRefCount = 0; + inputDesc->mOpenRefCount = 1; + inputDesc->mSamplingRate = samplingRate; + inputDesc->mFormat = format; + inputDesc->mChannelMask = channelMask; + inputDesc->mDevice = device; + inputDesc->mSessions.add(session); + inputDesc->mIsSoundTrigger = isSoundTrigger; + inputDesc->mPolicyMix = policyMix; + + ALOGV("getInputForAttr() returns input type = %d", inputType); + + addInput(*input, inputDesc); + mpClientInterface->onAudioPortListUpdate(); + return NO_ERROR; +} + +status_t AudioPolicyManager::startInput(audio_io_handle_t input, + audio_session_t session) +{ + ALOGV("startInput() input %d", input); + ssize_t index = mInputs.indexOfKey(input); + if (index < 0) { + ALOGW("startInput() unknown input %d", input); + return BAD_VALUE; + } + sp inputDesc = mInputs.valueAt(index); + + index = inputDesc->mSessions.indexOf(session); + if (index < 0) { + ALOGW("startInput() unknown session %d on input %d", session, input); + return BAD_VALUE; + } + + // virtual input devices are compatible with other input devices + if (!isVirtualInputDevice(inputDesc->mDevice)) { + + // for a non-virtual input device, check if there is another (non-virtual) active input + audio_io_handle_t activeInput = getActiveInput(); + if (activeInput != 0 && activeInput != input) { + + // If the already active input uses AUDIO_SOURCE_HOTWORD then it is closed, + // otherwise the active input continues and the new input cannot be started. + sp activeDesc = mInputs.valueFor(activeInput); + if (activeDesc->mInputSource == AUDIO_SOURCE_HOTWORD) { + ALOGW("startInput(%d) preempting low-priority input %d", input, activeInput); + stopInput(activeInput, activeDesc->mSessions.itemAt(0)); + releaseInput(activeInput, activeDesc->mSessions.itemAt(0)); + } else { + ALOGE("startInput(%d) failed: other input %d already started", input, activeInput); + return INVALID_OPERATION; + } + } + } + + if (inputDesc->mRefCount == 0) { + if (activeInputsCount() == 0) { + SoundTrigger::setCaptureState(true); + } + setInputDevice(input, getNewInputDevice(input), true /* force */); + + // automatically enable the remote submix output when input is started if not + // used by a policy mix of type MIX_TYPE_RECORDERS + // For remote submix (a virtual device), we open only one input per capture request. + if (audio_is_remote_submix_device(inputDesc->mDevice)) { + String8 address = String8(""); + if (inputDesc->mPolicyMix == NULL) { + address = String8("0"); + } else if (inputDesc->mPolicyMix->mMixType == MIX_TYPE_PLAYERS) { + address = inputDesc->mPolicyMix->mRegistrationId; + } + if (address != "") { + setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, + AUDIO_POLICY_DEVICE_STATE_AVAILABLE, + address, "remote-submix"); + } + } + } + + ALOGV("AudioPolicyManager::startInput() input source = %d", inputDesc->mInputSource); + + inputDesc->mRefCount++; + return NO_ERROR; +} + +status_t AudioPolicyManager::stopInput(audio_io_handle_t input, + audio_session_t session) +{ + ALOGV("stopInput() input %d", input); + ssize_t index = mInputs.indexOfKey(input); + if (index < 0) { + ALOGW("stopInput() unknown input %d", input); + return BAD_VALUE; + } + sp inputDesc = mInputs.valueAt(index); + + index = inputDesc->mSessions.indexOf(session); + if (index < 0) { + ALOGW("stopInput() unknown session %d on input %d", session, input); + return BAD_VALUE; + } + + if (inputDesc->mRefCount == 0) { + ALOGW("stopInput() input %d already stopped", input); + return INVALID_OPERATION; + } + + inputDesc->mRefCount--; + if (inputDesc->mRefCount == 0) { + + // automatically disable the remote submix output when input is stopped if not + // used by a policy mix of type MIX_TYPE_RECORDERS + if (audio_is_remote_submix_device(inputDesc->mDevice)) { + String8 address = String8(""); + if (inputDesc->mPolicyMix == NULL) { + address = String8("0"); + } else if (inputDesc->mPolicyMix->mMixType == MIX_TYPE_PLAYERS) { + address = inputDesc->mPolicyMix->mRegistrationId; + } + if (address != "") { + setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, + AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE, + address, "remote-submix"); + } + } + + resetInputDevice(input); + + if (activeInputsCount() == 0) { + SoundTrigger::setCaptureState(false); + } + } + return NO_ERROR; +} + +void AudioPolicyManager::releaseInput(audio_io_handle_t input, + audio_session_t session) +{ + ALOGV("releaseInput() %d", input); + ssize_t index = mInputs.indexOfKey(input); + if (index < 0) { + ALOGW("releaseInput() releasing unknown input %d", input); + return; + } + sp inputDesc = mInputs.valueAt(index); + ALOG_ASSERT(inputDesc != 0); + + index = inputDesc->mSessions.indexOf(session); + if (index < 0) { + ALOGW("releaseInput() unknown session %d on input %d", session, input); + return; + } + inputDesc->mSessions.remove(session); + if (inputDesc->mOpenRefCount == 0) { + ALOGW("releaseInput() invalid open ref count %d", inputDesc->mOpenRefCount); + return; + } + inputDesc->mOpenRefCount--; + if (inputDesc->mOpenRefCount > 0) { + ALOGV("releaseInput() exit > 0"); + return; + } + + closeInput(input); + mpClientInterface->onAudioPortListUpdate(); + ALOGV("releaseInput() exit"); +} + +void AudioPolicyManager::closeAllInputs() { + bool patchRemoved = false; + + for(size_t input_index = 0; input_index < mInputs.size(); input_index++) { + sp inputDesc = mInputs.valueAt(input_index); + ssize_t patch_index = mAudioPatches.indexOfKey(inputDesc->mPatchHandle); + if (patch_index >= 0) { + sp patchDesc = mAudioPatches.valueAt(patch_index); + status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0); + mAudioPatches.removeItemsAt(patch_index); + patchRemoved = true; + } + mpClientInterface->closeInput(mInputs.keyAt(input_index)); + } + mInputs.clear(); + nextAudioPortGeneration(); + + if (patchRemoved) { + mpClientInterface->onAudioPatchListUpdate(); + } +} + +void AudioPolicyManager::initStreamVolume(audio_stream_type_t stream, + int indexMin, + int indexMax) +{ + ALOGV("initStreamVolume() stream %d, min %d, max %d", stream , indexMin, indexMax); + if (indexMin < 0 || indexMin >= indexMax) { + ALOGW("initStreamVolume() invalid index limits for stream %d, min %d, max %d", stream , indexMin, indexMax); + return; + } + mStreams[stream].mIndexMin = indexMin; + mStreams[stream].mIndexMax = indexMax; + //FIXME: AUDIO_STREAM_ACCESSIBILITY volume follows AUDIO_STREAM_MUSIC for now + if (stream == AUDIO_STREAM_MUSIC) { + mStreams[AUDIO_STREAM_ACCESSIBILITY].mIndexMin = indexMin; + mStreams[AUDIO_STREAM_ACCESSIBILITY].mIndexMax = indexMax; + } +} + +status_t AudioPolicyManager::setStreamVolumeIndex(audio_stream_type_t stream, + int index, + audio_devices_t device) +{ + + if ((index < mStreams[stream].mIndexMin) || (index > mStreams[stream].mIndexMax)) { + return BAD_VALUE; + } + if (!audio_is_output_device(device)) { + return BAD_VALUE; + } + + // Force max volume if stream cannot be muted + if (!mStreams[stream].mCanBeMuted) index = mStreams[stream].mIndexMax; + + ALOGV("setStreamVolumeIndex() stream %d, device %04x, index %d", + stream, device, index); + + // if device is AUDIO_DEVICE_OUT_DEFAULT set default value and + // clear all device specific values + if (device == AUDIO_DEVICE_OUT_DEFAULT) { + mStreams[stream].mIndexCur.clear(); + } + mStreams[stream].mIndexCur.add(device, index); + + // update volume on all outputs whose current device is also selected by the same + // strategy as the device specified by the caller + audio_devices_t strategyDevice = getDeviceForStrategy(getStrategy(stream), true /*fromCache*/); + + + //FIXME: AUDIO_STREAM_ACCESSIBILITY volume follows AUDIO_STREAM_MUSIC for now + audio_devices_t accessibilityDevice = AUDIO_DEVICE_NONE; + if (stream == AUDIO_STREAM_MUSIC) { + mStreams[AUDIO_STREAM_ACCESSIBILITY].mIndexCur.add(device, index); + accessibilityDevice = getDeviceForStrategy(STRATEGY_ACCESSIBILITY, true /*fromCache*/); + } + if ((device != AUDIO_DEVICE_OUT_DEFAULT) && + (device & (strategyDevice | accessibilityDevice)) == 0) { + return NO_ERROR; + } + status_t status = NO_ERROR; + for (size_t i = 0; i < mOutputs.size(); i++) { + audio_devices_t curDevice = + ApmGains::getDeviceForVolume(mOutputs.valueAt(i)->device()); + if ((device == AUDIO_DEVICE_OUT_DEFAULT) || ((curDevice & strategyDevice) != 0)) { + status_t volStatus = checkAndSetVolume(stream, index, mOutputs.keyAt(i), curDevice); + if (volStatus != NO_ERROR) { + status = volStatus; + } + } + if ((device == AUDIO_DEVICE_OUT_DEFAULT) || ((curDevice & accessibilityDevice) != 0)) { + status_t volStatus = checkAndSetVolume(AUDIO_STREAM_ACCESSIBILITY, + index, mOutputs.keyAt(i), curDevice); + } + } + return status; +} + +status_t AudioPolicyManager::getStreamVolumeIndex(audio_stream_type_t stream, + int *index, + audio_devices_t device) +{ + if (index == NULL) { + return BAD_VALUE; + } + if (!audio_is_output_device(device)) { + return BAD_VALUE; + } + // if device is AUDIO_DEVICE_OUT_DEFAULT, return volume for device corresponding to + // the strategy the stream belongs to. + if (device == AUDIO_DEVICE_OUT_DEFAULT) { + device = getDeviceForStrategy(getStrategy(stream), true /*fromCache*/); + } + device = ApmGains::getDeviceForVolume(device); + + *index = mStreams[stream].getVolumeIndex(device); + ALOGV("getStreamVolumeIndex() stream %d device %08x index %d", stream, device, *index); + return NO_ERROR; +} + +audio_io_handle_t AudioPolicyManager::selectOutputForEffects( + const SortedVector& outputs) +{ + // select one output among several suitable for global effects. + // The priority is as follows: + // 1: An offloaded output. If the effect ends up not being offloadable, + // AudioFlinger will invalidate the track and the offloaded output + // will be closed causing the effect to be moved to a PCM output. + // 2: A deep buffer output + // 3: the first output in the list + + if (outputs.size() == 0) { + return 0; + } + + audio_io_handle_t outputOffloaded = 0; + audio_io_handle_t outputDeepBuffer = 0; + + for (size_t i = 0; i < outputs.size(); i++) { + sp desc = mOutputs.valueFor(outputs[i]); + ALOGV("selectOutputForEffects outputs[%zu] flags %x", i, desc->mFlags); + if ((desc->mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) { + outputOffloaded = outputs[i]; + } + if ((desc->mFlags & AUDIO_OUTPUT_FLAG_DEEP_BUFFER) != 0) { + outputDeepBuffer = outputs[i]; + } + } + + ALOGV("selectOutputForEffects outputOffloaded %d outputDeepBuffer %d", + outputOffloaded, outputDeepBuffer); + if (outputOffloaded != 0) { + return outputOffloaded; + } + if (outputDeepBuffer != 0) { + return outputDeepBuffer; + } + + return outputs[0]; +} + +audio_io_handle_t AudioPolicyManager::getOutputForEffect(const effect_descriptor_t *desc) +{ + // apply simple rule where global effects are attached to the same output as MUSIC streams + + routing_strategy strategy = getStrategy(AUDIO_STREAM_MUSIC); + audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/); + SortedVector dstOutputs = getOutputsForDevice(device, mOutputs); + + audio_io_handle_t output = selectOutputForEffects(dstOutputs); + ALOGV("getOutputForEffect() got output %d for fx %s flags %x", + output, (desc == NULL) ? "unspecified" : desc->name, (desc == NULL) ? 0 : desc->flags); + + return output; +} + +status_t AudioPolicyManager::registerEffect(const effect_descriptor_t *desc, + audio_io_handle_t io, + uint32_t strategy, + int session, + int id) +{ + ssize_t index = mOutputs.indexOfKey(io); + if (index < 0) { + index = mInputs.indexOfKey(io); + if (index < 0) { + ALOGW("registerEffect() unknown io %d", io); + return INVALID_OPERATION; + } + } + + if (mTotalEffectsMemory + desc->memoryUsage > getMaxEffectsMemory()) { + ALOGW("registerEffect() memory limit exceeded for Fx %s, Memory %d KB", + desc->name, desc->memoryUsage); + return INVALID_OPERATION; + } + mTotalEffectsMemory += desc->memoryUsage; + ALOGV("registerEffect() effect %s, io %d, strategy %d session %d id %d", + desc->name, io, strategy, session, id); + ALOGV("registerEffect() memory %d, total memory %d", desc->memoryUsage, mTotalEffectsMemory); + + sp effectDesc = new EffectDescriptor(); + memcpy (&effectDesc->mDesc, desc, sizeof(effect_descriptor_t)); + effectDesc->mIo = io; + effectDesc->mStrategy = (routing_strategy)strategy; + effectDesc->mSession = session; + effectDesc->mEnabled = false; + + mEffects.add(id, effectDesc); + + return NO_ERROR; +} + +status_t AudioPolicyManager::unregisterEffect(int id) +{ + ssize_t index = mEffects.indexOfKey(id); + if (index < 0) { + ALOGW("unregisterEffect() unknown effect ID %d", id); + return INVALID_OPERATION; + } + + sp effectDesc = mEffects.valueAt(index); + + setEffectEnabled(effectDesc, false); + + if (mTotalEffectsMemory < effectDesc->mDesc.memoryUsage) { + ALOGW("unregisterEffect() memory %d too big for total %d", + effectDesc->mDesc.memoryUsage, mTotalEffectsMemory); + effectDesc->mDesc.memoryUsage = mTotalEffectsMemory; + } + mTotalEffectsMemory -= effectDesc->mDesc.memoryUsage; + ALOGV("unregisterEffect() effect %s, ID %d, memory %d total memory %d", + effectDesc->mDesc.name, id, effectDesc->mDesc.memoryUsage, mTotalEffectsMemory); + + mEffects.removeItem(id); + + return NO_ERROR; +} + +status_t AudioPolicyManager::setEffectEnabled(int id, bool enabled) +{ + ssize_t index = mEffects.indexOfKey(id); + if (index < 0) { + ALOGW("unregisterEffect() unknown effect ID %d", id); + return INVALID_OPERATION; + } + + return setEffectEnabled(mEffects.valueAt(index), enabled); +} + +status_t AudioPolicyManager::setEffectEnabled(const sp& effectDesc, bool enabled) +{ + if (enabled == effectDesc->mEnabled) { + ALOGV("setEffectEnabled(%s) effect already %s", + enabled?"true":"false", enabled?"enabled":"disabled"); + return INVALID_OPERATION; + } + + if (enabled) { + if (mTotalEffectsCpuLoad + effectDesc->mDesc.cpuLoad > getMaxEffectsCpuLoad()) { + ALOGW("setEffectEnabled(true) CPU Load limit exceeded for Fx %s, CPU %f MIPS", + effectDesc->mDesc.name, (float)effectDesc->mDesc.cpuLoad/10); + return INVALID_OPERATION; + } + mTotalEffectsCpuLoad += effectDesc->mDesc.cpuLoad; + ALOGV("setEffectEnabled(true) total CPU %d", mTotalEffectsCpuLoad); + } else { + if (mTotalEffectsCpuLoad < effectDesc->mDesc.cpuLoad) { + ALOGW("setEffectEnabled(false) CPU load %d too high for total %d", + effectDesc->mDesc.cpuLoad, mTotalEffectsCpuLoad); + effectDesc->mDesc.cpuLoad = mTotalEffectsCpuLoad; + } + mTotalEffectsCpuLoad -= effectDesc->mDesc.cpuLoad; + ALOGV("setEffectEnabled(false) total CPU %d", mTotalEffectsCpuLoad); + } + effectDesc->mEnabled = enabled; + return NO_ERROR; +} + +bool AudioPolicyManager::isNonOffloadableEffectEnabled() +{ + for (size_t i = 0; i < mEffects.size(); i++) { + sp effectDesc = mEffects.valueAt(i); + if (effectDesc->mEnabled && (effectDesc->mStrategy == STRATEGY_MEDIA) && + ((effectDesc->mDesc.flags & EFFECT_FLAG_OFFLOAD_SUPPORTED) == 0)) { + ALOGV("isNonOffloadableEffectEnabled() non offloadable effect %s enabled on session %d", + effectDesc->mDesc.name, effectDesc->mSession); + return true; + } + } + return false; +} + +bool AudioPolicyManager::isStreamActive(audio_stream_type_t stream, uint32_t inPastMs) const +{ + nsecs_t sysTime = systemTime(); + for (size_t i = 0; i < mOutputs.size(); i++) { + const sp outputDesc = mOutputs.valueAt(i); + if (outputDesc->isStreamActive(stream, inPastMs, sysTime)) { + return true; + } + } + return false; +} + +bool AudioPolicyManager::isStreamActiveRemotely(audio_stream_type_t stream, + uint32_t inPastMs) const +{ + nsecs_t sysTime = systemTime(); + for (size_t i = 0; i < mOutputs.size(); i++) { + const sp outputDesc = mOutputs.valueAt(i); + if (((outputDesc->device() & APM_AUDIO_OUT_DEVICE_REMOTE_ALL) != 0) && + outputDesc->isStreamActive(stream, inPastMs, sysTime)) { + // do not consider re routing (when the output is going to a dynamic policy) + // as "remote playback" + if (outputDesc->mPolicyMix == NULL) { + return true; + } + } + } + return false; +} + +bool AudioPolicyManager::isSourceActive(audio_source_t source) const +{ + for (size_t i = 0; i < mInputs.size(); i++) { + const sp inputDescriptor = mInputs.valueAt(i); + if (inputDescriptor->mRefCount == 0) { + continue; + } + if (inputDescriptor->mInputSource == (int)source) { + return true; + } + // AUDIO_SOURCE_HOTWORD is equivalent to AUDIO_SOURCE_VOICE_RECOGNITION only if it + // corresponds to an active capture triggered by a hardware hotword recognition + if ((source == AUDIO_SOURCE_VOICE_RECOGNITION) && + (inputDescriptor->mInputSource == AUDIO_SOURCE_HOTWORD)) { + // FIXME: we should not assume that the first session is the active one and keep + // activity count per session. Same in startInput(). + ssize_t index = mSoundTriggerSessions.indexOfKey(inputDescriptor->mSessions.itemAt(0)); + if (index >= 0) { + return true; + } + } + } + return false; +} + +// Register a list of custom mixes with their attributes and format. +// When a mix is registered, corresponding input and output profiles are +// added to the remote submix hw module. The profile contains only the +// parameters (sampling rate, format...) specified by the mix. +// The corresponding input remote submix device is also connected. +// +// When a remote submix device is connected, the address is checked to select the +// appropriate profile and the corresponding input or output stream is opened. +// +// When capture starts, getInputForAttr() will: +// - 1 look for a mix matching the address passed in attribtutes tags if any +// - 2 if none found, getDeviceForInputSource() will: +// - 2.1 look for a mix matching the attributes source +// - 2.2 if none found, default to device selection by policy rules +// At this time, the corresponding output remote submix device is also connected +// and active playback use cases can be transferred to this mix if needed when reconnecting +// after AudioTracks are invalidated +// +// When playback starts, getOutputForAttr() will: +// - 1 look for a mix matching the address passed in attribtutes tags if any +// - 2 if none found, look for a mix matching the attributes usage +// - 3 if none found, default to device and output selection by policy rules. + +status_t AudioPolicyManager::registerPolicyMixes(Vector mixes) +{ + sp module; + for (size_t i = 0; i < mHwModules.size(); i++) { + if (strcmp(AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX, mHwModules[i]->mName) == 0 && + mHwModules[i]->mHandle != 0) { + module = mHwModules[i]; + break; + } + } + + if (module == 0) { + return INVALID_OPERATION; + } + + ALOGV("registerPolicyMixes() num mixes %d", mixes.size()); + + for (size_t i = 0; i < mixes.size(); i++) { + String8 address = mixes[i].mRegistrationId; + ssize_t index = mPolicyMixes.indexOfKey(address); + if (index >= 0) { + ALOGE("registerPolicyMixes(): mix for address %s already registered", address.string()); + continue; + } + audio_config_t outputConfig = mixes[i].mFormat; + audio_config_t inputConfig = mixes[i].mFormat; + // NOTE: audio flinger mixer does not support mono output: configure remote submix HAL in + // stereo and let audio flinger do the channel conversion if needed. + outputConfig.channel_mask = AUDIO_CHANNEL_OUT_STEREO; + inputConfig.channel_mask = AUDIO_CHANNEL_IN_STEREO; + module->addOutputProfile(address, &outputConfig, + AUDIO_DEVICE_OUT_REMOTE_SUBMIX, address); + module->addInputProfile(address, &inputConfig, + AUDIO_DEVICE_IN_REMOTE_SUBMIX, address); + sp policyMix = new AudioPolicyMix(); + policyMix->mMix = mixes[i]; + mPolicyMixes.add(address, policyMix); + if (mixes[i].mMixType == MIX_TYPE_PLAYERS) { + setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX, + AUDIO_POLICY_DEVICE_STATE_AVAILABLE, + address.string(), "remote-submix"); + } else { + setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, + AUDIO_POLICY_DEVICE_STATE_AVAILABLE, + address.string(), "remote-submix"); + } + } + return NO_ERROR; +} + +status_t AudioPolicyManager::unregisterPolicyMixes(Vector mixes) +{ + sp module; + for (size_t i = 0; i < mHwModules.size(); i++) { + if (strcmp(AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX, mHwModules[i]->mName) == 0 && + mHwModules[i]->mHandle != 0) { + module = mHwModules[i]; + break; + } + } + + if (module == 0) { + return INVALID_OPERATION; + } + + ALOGV("unregisterPolicyMixes() num mixes %d", mixes.size()); + + for (size_t i = 0; i < mixes.size(); i++) { + String8 address = mixes[i].mRegistrationId; + ssize_t index = mPolicyMixes.indexOfKey(address); + if (index < 0) { + ALOGE("unregisterPolicyMixes(): mix for address %s not registered", address.string()); + continue; + } + + mPolicyMixes.removeItemsAt(index); + + if (getDeviceConnectionState(AUDIO_DEVICE_IN_REMOTE_SUBMIX, address.string()) == + AUDIO_POLICY_DEVICE_STATE_AVAILABLE) + { + setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX, + AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE, + address.string(), "remote-submix"); + } + + if (getDeviceConnectionState(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, address.string()) == + AUDIO_POLICY_DEVICE_STATE_AVAILABLE) + { + setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, + AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE, + address.string(), "remote-submix"); + } + module->removeOutputProfile(address); + module->removeInputProfile(address); + } + return NO_ERROR; +} + + +status_t AudioPolicyManager::dump(int fd) +{ + const size_t SIZE = 256; + char buffer[SIZE]; + String8 result; + + snprintf(buffer, SIZE, "\nAudioPolicyManager Dump: %p\n", this); + result.append(buffer); + + snprintf(buffer, SIZE, " Primary Output: %d\n", mPrimaryOutput); + result.append(buffer); + snprintf(buffer, SIZE, " Phone state: %d\n", mPhoneState); + result.append(buffer); + snprintf(buffer, SIZE, " Force use for communications %d\n", + mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION]); + result.append(buffer); + snprintf(buffer, SIZE, " Force use for media %d\n", mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA]); + result.append(buffer); + snprintf(buffer, SIZE, " Force use for record %d\n", mForceUse[AUDIO_POLICY_FORCE_FOR_RECORD]); + result.append(buffer); + snprintf(buffer, SIZE, " Force use for dock %d\n", mForceUse[AUDIO_POLICY_FORCE_FOR_DOCK]); + result.append(buffer); + snprintf(buffer, SIZE, " Force use for system %d\n", mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM]); + result.append(buffer); + snprintf(buffer, SIZE, " Force use for hdmi system audio %d\n", + mForceUse[AUDIO_POLICY_FORCE_FOR_HDMI_SYSTEM_AUDIO]); + result.append(buffer); + + snprintf(buffer, SIZE, " Available output devices:\n"); + result.append(buffer); + write(fd, result.string(), result.size()); + for (size_t i = 0; i < mAvailableOutputDevices.size(); i++) { + mAvailableOutputDevices[i]->dump(fd, 2, i); + } + snprintf(buffer, SIZE, "\n Available input devices:\n"); + write(fd, buffer, strlen(buffer)); + for (size_t i = 0; i < mAvailableInputDevices.size(); i++) { + mAvailableInputDevices[i]->dump(fd, 2, i); + } + + snprintf(buffer, SIZE, "\nHW Modules dump:\n"); + write(fd, buffer, strlen(buffer)); + for (size_t i = 0; i < mHwModules.size(); i++) { + snprintf(buffer, SIZE, "- HW Module %zu:\n", i + 1); + write(fd, buffer, strlen(buffer)); + mHwModules[i]->dump(fd); + } + + snprintf(buffer, SIZE, "\nOutputs dump:\n"); + write(fd, buffer, strlen(buffer)); + for (size_t i = 0; i < mOutputs.size(); i++) { + snprintf(buffer, SIZE, "- Output %d dump:\n", mOutputs.keyAt(i)); + write(fd, buffer, strlen(buffer)); + mOutputs.valueAt(i)->dump(fd); + } + + snprintf(buffer, SIZE, "\nInputs dump:\n"); + write(fd, buffer, strlen(buffer)); + for (size_t i = 0; i < mInputs.size(); i++) { + snprintf(buffer, SIZE, "- Input %d dump:\n", mInputs.keyAt(i)); + write(fd, buffer, strlen(buffer)); + mInputs.valueAt(i)->dump(fd); + } + + snprintf(buffer, SIZE, "\nStreams dump:\n"); + write(fd, buffer, strlen(buffer)); + snprintf(buffer, SIZE, + " Stream Can be muted Index Min Index Max Index Cur [device : index]...\n"); + write(fd, buffer, strlen(buffer)); + for (size_t i = 0; i < AUDIO_STREAM_CNT; i++) { + snprintf(buffer, SIZE, " %02zu ", i); + write(fd, buffer, strlen(buffer)); + mStreams[i].dump(fd); + } + + snprintf(buffer, SIZE, "\nTotal Effects CPU: %f MIPS, Total Effects memory: %d KB\n", + (float)mTotalEffectsCpuLoad/10, mTotalEffectsMemory); + write(fd, buffer, strlen(buffer)); + + snprintf(buffer, SIZE, "Registered effects:\n"); + write(fd, buffer, strlen(buffer)); + for (size_t i = 0; i < mEffects.size(); i++) { + snprintf(buffer, SIZE, "- Effect %d dump:\n", mEffects.keyAt(i)); + write(fd, buffer, strlen(buffer)); + mEffects.valueAt(i)->dump(fd); + } + + snprintf(buffer, SIZE, "\nAudio Patches:\n"); + write(fd, buffer, strlen(buffer)); + for (size_t i = 0; i < mAudioPatches.size(); i++) { + mAudioPatches[i]->dump(fd, 2, i); + } + + return NO_ERROR; +} + +// This function checks for the parameters which can be offloaded. +// This can be enhanced depending on the capability of the DSP and policy +// of the system. +bool AudioPolicyManager::isOffloadSupported(const audio_offload_info_t& offloadInfo) +{ + ALOGV("isOffloadSupported: SR=%u, CM=0x%x, Format=0x%x, StreamType=%d," + " BitRate=%u, duration=%" PRId64 " us, has_video=%d", + offloadInfo.sample_rate, offloadInfo.channel_mask, + offloadInfo.format, + offloadInfo.stream_type, offloadInfo.bit_rate, offloadInfo.duration_us, + offloadInfo.has_video); + + // Check if offload has been disabled + char propValue[PROPERTY_VALUE_MAX]; + if (property_get("audio.offload.disable", propValue, "0")) { + if (atoi(propValue) != 0) { + ALOGV("offload disabled by audio.offload.disable=%s", propValue ); + return false; + } + } + + // Check if stream type is music, then only allow offload as of now. + if (offloadInfo.stream_type != AUDIO_STREAM_MUSIC) + { + ALOGV("isOffloadSupported: stream_type != MUSIC, returning false"); + return false; + } + + //TODO: enable audio offloading with video when ready + if (offloadInfo.has_video) + { + ALOGV("isOffloadSupported: has_video == true, returning false"); + return false; + } + + //If duration is less than minimum value defined in property, return false + if (property_get("audio.offload.min.duration.secs", propValue, NULL)) { + if (offloadInfo.duration_us < (atoi(propValue) * 1000000 )) { + ALOGV("Offload denied by duration < audio.offload.min.duration.secs(=%s)", propValue); + return false; + } + } else if (offloadInfo.duration_us < OFFLOAD_DEFAULT_MIN_DURATION_SECS * 1000000) { + ALOGV("Offload denied by duration < default min(=%u)", OFFLOAD_DEFAULT_MIN_DURATION_SECS); + return false; + } + + // Do not allow offloading if one non offloadable effect is enabled. This prevents from + // creating an offloaded track and tearing it down immediately after start when audioflinger + // detects there is an active non offloadable effect. + // FIXME: We should check the audio session here but we do not have it in this context. + // This may prevent offloading in rare situations where effects are left active by apps + // in the background. + if (isNonOffloadableEffectEnabled()) { + return false; + } + + // See if there is a profile to support this. + // AUDIO_DEVICE_NONE + sp profile = getProfileForDirectOutput(AUDIO_DEVICE_NONE /*ignore device */, + offloadInfo.sample_rate, + offloadInfo.format, + offloadInfo.channel_mask, + AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD); + ALOGV("isOffloadSupported() profile %sfound", profile != 0 ? "" : "NOT "); + return (profile != 0); +} + +status_t AudioPolicyManager::listAudioPorts(audio_port_role_t role, + audio_port_type_t type, + unsigned int *num_ports, + struct audio_port *ports, + unsigned int *generation) +{ + if (num_ports == NULL || (*num_ports != 0 && ports == NULL) || + generation == NULL) { + return BAD_VALUE; + } + ALOGV("listAudioPorts() role %d type %d num_ports %d ports %p", role, type, *num_ports, ports); + if (ports == NULL) { + *num_ports = 0; + } + + size_t portsWritten = 0; + size_t portsMax = *num_ports; + *num_ports = 0; + if (type == AUDIO_PORT_TYPE_NONE || type == AUDIO_PORT_TYPE_DEVICE) { + if (role == AUDIO_PORT_ROLE_SINK || role == AUDIO_PORT_ROLE_NONE) { + for (size_t i = 0; + i < mAvailableOutputDevices.size() && portsWritten < portsMax; i++) { + mAvailableOutputDevices[i]->toAudioPort(&ports[portsWritten++]); + } + *num_ports += mAvailableOutputDevices.size(); + } + if (role == AUDIO_PORT_ROLE_SOURCE || role == AUDIO_PORT_ROLE_NONE) { + for (size_t i = 0; + i < mAvailableInputDevices.size() && portsWritten < portsMax; i++) { + mAvailableInputDevices[i]->toAudioPort(&ports[portsWritten++]); + } + *num_ports += mAvailableInputDevices.size(); + } + } + if (type == AUDIO_PORT_TYPE_NONE || type == AUDIO_PORT_TYPE_MIX) { + if (role == AUDIO_PORT_ROLE_SINK || role == AUDIO_PORT_ROLE_NONE) { + for (size_t i = 0; i < mInputs.size() && portsWritten < portsMax; i++) { + mInputs[i]->toAudioPort(&ports[portsWritten++]); + } + *num_ports += mInputs.size(); + } + if (role == AUDIO_PORT_ROLE_SOURCE || role == AUDIO_PORT_ROLE_NONE) { + size_t numOutputs = 0; + for (size_t i = 0; i < mOutputs.size(); i++) { + if (!mOutputs[i]->isDuplicated()) { + numOutputs++; + if (portsWritten < portsMax) { + mOutputs[i]->toAudioPort(&ports[portsWritten++]); + } + } + } + *num_ports += numOutputs; + } + } + *generation = curAudioPortGeneration(); + ALOGV("listAudioPorts() got %zu ports needed %d", portsWritten, *num_ports); + return NO_ERROR; +} + +status_t AudioPolicyManager::getAudioPort(struct audio_port *port __unused) +{ + return NO_ERROR; +} + +sp AudioPolicyManager::getOutputFromId( + audio_port_handle_t id) const +{ + sp outputDesc = NULL; + for (size_t i = 0; i < mOutputs.size(); i++) { + outputDesc = mOutputs.valueAt(i); + if (outputDesc->mId == id) { + break; + } + } + return outputDesc; +} + +sp AudioPolicyManager::getInputFromId( + audio_port_handle_t id) const +{ + sp inputDesc = NULL; + for (size_t i = 0; i < mInputs.size(); i++) { + inputDesc = mInputs.valueAt(i); + if (inputDesc->mId == id) { + break; + } + } + return inputDesc; +} + +sp AudioPolicyManager::getModuleForDevice( + audio_devices_t device) const +{ + sp module; + + for (size_t i = 0; i < mHwModules.size(); i++) { + if (mHwModules[i]->mHandle == 0) { + continue; + } + if (audio_is_output_device(device)) { + for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++) + { + if (mHwModules[i]->mOutputProfiles[j]->mSupportedDevices.types() & device) { + return mHwModules[i]; + } + } + } else { + for (size_t j = 0; j < mHwModules[i]->mInputProfiles.size(); j++) { + if (mHwModules[i]->mInputProfiles[j]->mSupportedDevices.types() & + device & ~AUDIO_DEVICE_BIT_IN) { + return mHwModules[i]; + } + } + } + } + return module; +} + +sp AudioPolicyManager::getModuleFromName(const char *name) const +{ + sp module; + + for (size_t i = 0; i < mHwModules.size(); i++) + { + if (strcmp(mHwModules[i]->mName, name) == 0) { + return mHwModules[i]; + } + } + return module; +} + +audio_devices_t AudioPolicyManager::availablePrimaryOutputDevices() +{ + sp outputDesc = mOutputs.valueFor(mPrimaryOutput); + audio_devices_t devices = outputDesc->mProfile->mSupportedDevices.types(); + return devices & mAvailableOutputDevices.types(); +} + +audio_devices_t AudioPolicyManager::availablePrimaryInputDevices() +{ + audio_module_handle_t primaryHandle = + mOutputs.valueFor(mPrimaryOutput)->mProfile->mModule->mHandle; + audio_devices_t devices = AUDIO_DEVICE_NONE; + for (size_t i = 0; i < mAvailableInputDevices.size(); i++) { + if (mAvailableInputDevices[i]->mModule->mHandle == primaryHandle) { + devices |= mAvailableInputDevices[i]->mDeviceType; + } + } + return devices; +} + +status_t AudioPolicyManager::createAudioPatch(const struct audio_patch *patch, + audio_patch_handle_t *handle, + uid_t uid) +{ + ALOGV("createAudioPatch()"); + + if (handle == NULL || patch == NULL) { + return BAD_VALUE; + } + ALOGV("createAudioPatch() num sources %d num sinks %d", patch->num_sources, patch->num_sinks); + + if (patch->num_sources == 0 || patch->num_sources > AUDIO_PATCH_PORTS_MAX || + patch->num_sinks == 0 || patch->num_sinks > AUDIO_PATCH_PORTS_MAX) { + return BAD_VALUE; + } + // only one source per audio patch supported for now + if (patch->num_sources > 1) { + return INVALID_OPERATION; + } + + if (patch->sources[0].role != AUDIO_PORT_ROLE_SOURCE) { + return INVALID_OPERATION; + } + for (size_t i = 0; i < patch->num_sinks; i++) { + if (patch->sinks[i].role != AUDIO_PORT_ROLE_SINK) { + return INVALID_OPERATION; + } + } + + sp patchDesc; + ssize_t index = mAudioPatches.indexOfKey(*handle); + + ALOGV("createAudioPatch source id %d role %d type %d", patch->sources[0].id, + patch->sources[0].role, + patch->sources[0].type); +#if LOG_NDEBUG == 0 + for (size_t i = 0; i < patch->num_sinks; i++) { + ALOGV("createAudioPatch sink %d: id %d role %d type %d", i, patch->sinks[i].id, + patch->sinks[i].role, + patch->sinks[i].type); + } +#endif + + if (index >= 0) { + patchDesc = mAudioPatches.valueAt(index); + ALOGV("createAudioPatch() mUidCached %d patchDesc->mUid %d uid %d", + mUidCached, patchDesc->mUid, uid); + if (patchDesc->mUid != mUidCached && uid != patchDesc->mUid) { + return INVALID_OPERATION; + } + } else { + *handle = 0; + } + + if (patch->sources[0].type == AUDIO_PORT_TYPE_MIX) { + sp outputDesc = getOutputFromId(patch->sources[0].id); + if (outputDesc == NULL) { + ALOGV("createAudioPatch() output not found for id %d", patch->sources[0].id); + return BAD_VALUE; + } + ALOG_ASSERT(!outputDesc->isDuplicated(),"duplicated output %d in source in ports", + outputDesc->mIoHandle); + if (patchDesc != 0) { + if (patchDesc->mPatch.sources[0].id != patch->sources[0].id) { + ALOGV("createAudioPatch() source id differs for patch current id %d new id %d", + patchDesc->mPatch.sources[0].id, patch->sources[0].id); + return BAD_VALUE; + } + } + DeviceVector devices; + for (size_t i = 0; i < patch->num_sinks; i++) { + // Only support mix to devices connection + // TODO add support for mix to mix connection + if (patch->sinks[i].type != AUDIO_PORT_TYPE_DEVICE) { + ALOGV("createAudioPatch() source mix but sink is not a device"); + return INVALID_OPERATION; + } + sp devDesc = + mAvailableOutputDevices.getDeviceFromId(patch->sinks[i].id); + if (devDesc == 0) { + ALOGV("createAudioPatch() out device not found for id %d", patch->sinks[i].id); + return BAD_VALUE; + } + + if (!outputDesc->mProfile->isCompatibleProfile(devDesc->mDeviceType, + devDesc->mAddress, + patch->sources[0].sample_rate, + NULL, // updatedSamplingRate + patch->sources[0].format, + patch->sources[0].channel_mask, + AUDIO_OUTPUT_FLAG_NONE /*FIXME*/)) { + ALOGV("createAudioPatch() profile not supported for device %08x", + devDesc->mDeviceType); + return INVALID_OPERATION; + } + devices.add(devDesc); + } + if (devices.size() == 0) { + return INVALID_OPERATION; + } + + // TODO: reconfigure output format and channels here + ALOGV("createAudioPatch() setting device %08x on output %d", + devices.types(), outputDesc->mIoHandle); + setOutputDevice(outputDesc->mIoHandle, devices.types(), true, 0, handle); + index = mAudioPatches.indexOfKey(*handle); + if (index >= 0) { + if (patchDesc != 0 && patchDesc != mAudioPatches.valueAt(index)) { + ALOGW("createAudioPatch() setOutputDevice() did not reuse the patch provided"); + } + patchDesc = mAudioPatches.valueAt(index); + patchDesc->mUid = uid; + ALOGV("createAudioPatch() success"); + } else { + ALOGW("createAudioPatch() setOutputDevice() failed to create a patch"); + return INVALID_OPERATION; + } + } else if (patch->sources[0].type == AUDIO_PORT_TYPE_DEVICE) { + if (patch->sinks[0].type == AUDIO_PORT_TYPE_MIX) { + // input device to input mix connection + // only one sink supported when connecting an input device to a mix + if (patch->num_sinks > 1) { + return INVALID_OPERATION; + } + sp inputDesc = getInputFromId(patch->sinks[0].id); + if (inputDesc == NULL) { + return BAD_VALUE; + } + if (patchDesc != 0) { + if (patchDesc->mPatch.sinks[0].id != patch->sinks[0].id) { + return BAD_VALUE; + } + } + sp devDesc = + mAvailableInputDevices.getDeviceFromId(patch->sources[0].id); + if (devDesc == 0) { + return BAD_VALUE; + } + + if (!inputDesc->mProfile->isCompatibleProfile(devDesc->mDeviceType, + devDesc->mAddress, + patch->sinks[0].sample_rate, + NULL, /*updatedSampleRate*/ + patch->sinks[0].format, + patch->sinks[0].channel_mask, + // FIXME for the parameter type, + // and the NONE + (audio_output_flags_t) + AUDIO_INPUT_FLAG_NONE)) { + return INVALID_OPERATION; + } + // TODO: reconfigure output format and channels here + ALOGV("createAudioPatch() setting device %08x on output %d", + devDesc->mDeviceType, inputDesc->mIoHandle); + setInputDevice(inputDesc->mIoHandle, devDesc->mDeviceType, true, handle); + index = mAudioPatches.indexOfKey(*handle); + if (index >= 0) { + if (patchDesc != 0 && patchDesc != mAudioPatches.valueAt(index)) { + ALOGW("createAudioPatch() setInputDevice() did not reuse the patch provided"); + } + patchDesc = mAudioPatches.valueAt(index); + patchDesc->mUid = uid; + ALOGV("createAudioPatch() success"); + } else { + ALOGW("createAudioPatch() setInputDevice() failed to create a patch"); + return INVALID_OPERATION; + } + } else if (patch->sinks[0].type == AUDIO_PORT_TYPE_DEVICE) { + // device to device connection + if (patchDesc != 0) { + if (patchDesc->mPatch.sources[0].id != patch->sources[0].id) { + return BAD_VALUE; + } + } + sp srcDeviceDesc = + mAvailableInputDevices.getDeviceFromId(patch->sources[0].id); + if (srcDeviceDesc == 0) { + return BAD_VALUE; + } + + //update source and sink with our own data as the data passed in the patch may + // be incomplete. + struct audio_patch newPatch = *patch; + srcDeviceDesc->toAudioPortConfig(&newPatch.sources[0], &patch->sources[0]); + + for (size_t i = 0; i < patch->num_sinks; i++) { + if (patch->sinks[i].type != AUDIO_PORT_TYPE_DEVICE) { + ALOGV("createAudioPatch() source device but one sink is not a device"); + return INVALID_OPERATION; + } + + sp sinkDeviceDesc = + mAvailableOutputDevices.getDeviceFromId(patch->sinks[i].id); + if (sinkDeviceDesc == 0) { + return BAD_VALUE; + } + sinkDeviceDesc->toAudioPortConfig(&newPatch.sinks[i], &patch->sinks[i]); + + if (srcDeviceDesc->mModule != sinkDeviceDesc->mModule) { + // only one sink supported when connected devices across HW modules + if (patch->num_sinks > 1) { + return INVALID_OPERATION; + } + SortedVector outputs = + getOutputsForDevice(sinkDeviceDesc->mDeviceType, + mOutputs); + // if the sink device is reachable via an opened output stream, request to go via + // this output stream by adding a second source to the patch description + audio_io_handle_t output = selectOutput(outputs, + AUDIO_OUTPUT_FLAG_NONE, + AUDIO_FORMAT_INVALID); + if (output != AUDIO_IO_HANDLE_NONE) { + sp outputDesc = mOutputs.valueFor(output); + if (outputDesc->isDuplicated()) { + return INVALID_OPERATION; + } + outputDesc->toAudioPortConfig(&newPatch.sources[1], &patch->sources[0]); + newPatch.num_sources = 2; + } + } + } + // TODO: check from routing capabilities in config file and other conflicting patches + + audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE; + if (index >= 0) { + afPatchHandle = patchDesc->mAfPatchHandle; + } + + status_t status = mpClientInterface->createAudioPatch(&newPatch, + &afPatchHandle, + 0); + ALOGV("createAudioPatch() patch panel returned %d patchHandle %d", + status, afPatchHandle); + if (status == NO_ERROR) { + if (index < 0) { + patchDesc = new AudioPatch((audio_patch_handle_t)nextUniqueId(), + &newPatch, uid); + addAudioPatch(patchDesc->mHandle, patchDesc); + } else { + patchDesc->mPatch = newPatch; + } + patchDesc->mAfPatchHandle = afPatchHandle; + *handle = patchDesc->mHandle; + nextAudioPortGeneration(); + mpClientInterface->onAudioPatchListUpdate(); + } else { + ALOGW("createAudioPatch() patch panel could not connect device patch, error %d", + status); + return INVALID_OPERATION; + } + } else { + return BAD_VALUE; + } + } else { + return BAD_VALUE; + } + return NO_ERROR; +} + +status_t AudioPolicyManager::releaseAudioPatch(audio_patch_handle_t handle, + uid_t uid) +{ + ALOGV("releaseAudioPatch() patch %d", handle); + + ssize_t index = mAudioPatches.indexOfKey(handle); + + if (index < 0) { + return BAD_VALUE; + } + sp patchDesc = mAudioPatches.valueAt(index); + ALOGV("releaseAudioPatch() mUidCached %d patchDesc->mUid %d uid %d", + mUidCached, patchDesc->mUid, uid); + if (patchDesc->mUid != mUidCached && uid != patchDesc->mUid) { + return INVALID_OPERATION; + } + + struct audio_patch *patch = &patchDesc->mPatch; + patchDesc->mUid = mUidCached; + if (patch->sources[0].type == AUDIO_PORT_TYPE_MIX) { + sp outputDesc = getOutputFromId(patch->sources[0].id); + if (outputDesc == NULL) { + ALOGV("releaseAudioPatch() output not found for id %d", patch->sources[0].id); + return BAD_VALUE; + } + + setOutputDevice(outputDesc->mIoHandle, + getNewOutputDevice(outputDesc->mIoHandle, true /*fromCache*/), + true, + 0, + NULL); + } else if (patch->sources[0].type == AUDIO_PORT_TYPE_DEVICE) { + if (patch->sinks[0].type == AUDIO_PORT_TYPE_MIX) { + sp inputDesc = getInputFromId(patch->sinks[0].id); + if (inputDesc == NULL) { + ALOGV("releaseAudioPatch() input not found for id %d", patch->sinks[0].id); + return BAD_VALUE; + } + setInputDevice(inputDesc->mIoHandle, + getNewInputDevice(inputDesc->mIoHandle), + true, + NULL); + } else if (patch->sinks[0].type == AUDIO_PORT_TYPE_DEVICE) { + audio_patch_handle_t afPatchHandle = patchDesc->mAfPatchHandle; + status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0); + ALOGV("releaseAudioPatch() patch panel returned %d patchHandle %d", + status, patchDesc->mAfPatchHandle); + removeAudioPatch(patchDesc->mHandle); + nextAudioPortGeneration(); + mpClientInterface->onAudioPatchListUpdate(); + } else { + return BAD_VALUE; + } + } else { + return BAD_VALUE; + } + return NO_ERROR; +} + +status_t AudioPolicyManager::listAudioPatches(unsigned int *num_patches, + struct audio_patch *patches, + unsigned int *generation) +{ + if (num_patches == NULL || (*num_patches != 0 && patches == NULL) || + generation == NULL) { + return BAD_VALUE; + } + ALOGV("listAudioPatches() num_patches %d patches %p available patches %zu", + *num_patches, patches, mAudioPatches.size()); + if (patches == NULL) { + *num_patches = 0; + } + + size_t patchesWritten = 0; + size_t patchesMax = *num_patches; + for (size_t i = 0; + i < mAudioPatches.size() && patchesWritten < patchesMax; i++) { + patches[patchesWritten] = mAudioPatches[i]->mPatch; + patches[patchesWritten++].id = mAudioPatches[i]->mHandle; + ALOGV("listAudioPatches() patch %zu num_sources %d num_sinks %d", + i, mAudioPatches[i]->mPatch.num_sources, mAudioPatches[i]->mPatch.num_sinks); + } + *num_patches = mAudioPatches.size(); + + *generation = curAudioPortGeneration(); + ALOGV("listAudioPatches() got %zu patches needed %d", patchesWritten, *num_patches); + return NO_ERROR; +} + +status_t AudioPolicyManager::setAudioPortConfig(const struct audio_port_config *config) +{ + ALOGV("setAudioPortConfig()"); + + if (config == NULL) { + return BAD_VALUE; + } + ALOGV("setAudioPortConfig() on port handle %d", config->id); + // Only support gain configuration for now + if (config->config_mask != AUDIO_PORT_CONFIG_GAIN) { + return INVALID_OPERATION; + } + + sp audioPortConfig; + if (config->type == AUDIO_PORT_TYPE_MIX) { + if (config->role == AUDIO_PORT_ROLE_SOURCE) { + sp outputDesc = getOutputFromId(config->id); + if (outputDesc == NULL) { + return BAD_VALUE; + } + ALOG_ASSERT(!outputDesc->isDuplicated(), + "setAudioPortConfig() called on duplicated output %d", + outputDesc->mIoHandle); + audioPortConfig = outputDesc; + } else if (config->role == AUDIO_PORT_ROLE_SINK) { + sp inputDesc = getInputFromId(config->id); + if (inputDesc == NULL) { + return BAD_VALUE; + } + audioPortConfig = inputDesc; + } else { + return BAD_VALUE; + } + } else if (config->type == AUDIO_PORT_TYPE_DEVICE) { + sp deviceDesc; + if (config->role == AUDIO_PORT_ROLE_SOURCE) { + deviceDesc = mAvailableInputDevices.getDeviceFromId(config->id); + } else if (config->role == AUDIO_PORT_ROLE_SINK) { + deviceDesc = mAvailableOutputDevices.getDeviceFromId(config->id); + } else { + return BAD_VALUE; + } + if (deviceDesc == NULL) { + return BAD_VALUE; + } + audioPortConfig = deviceDesc; + } else { + return BAD_VALUE; + } + + struct audio_port_config backupConfig; + status_t status = audioPortConfig->applyAudioPortConfig(config, &backupConfig); + if (status == NO_ERROR) { + struct audio_port_config newConfig; + audioPortConfig->toAudioPortConfig(&newConfig, config); + status = mpClientInterface->setAudioPortConfig(&newConfig, 0); + } + if (status != NO_ERROR) { + audioPortConfig->applyAudioPortConfig(&backupConfig); + } + + return status; +} + +void AudioPolicyManager::clearAudioPatches(uid_t uid) +{ + for (ssize_t i = (ssize_t)mAudioPatches.size() - 1; i >= 0; i--) { + sp patchDesc = mAudioPatches.valueAt(i); + if (patchDesc->mUid == uid) { + releaseAudioPatch(mAudioPatches.keyAt(i), uid); + } + } +} + +status_t AudioPolicyManager::acquireSoundTriggerSession(audio_session_t *session, + audio_io_handle_t *ioHandle, + audio_devices_t *device) +{ + *session = (audio_session_t)mpClientInterface->newAudioUniqueId(); + *ioHandle = (audio_io_handle_t)mpClientInterface->newAudioUniqueId(); + *device = getDeviceAndMixForInputSource(AUDIO_SOURCE_HOTWORD); + + mSoundTriggerSessions.add(*session, *ioHandle); + + return NO_ERROR; +} + +status_t AudioPolicyManager::releaseSoundTriggerSession(audio_session_t session) +{ + ssize_t index = mSoundTriggerSessions.indexOfKey(session); + if (index < 0) { + ALOGW("acquireSoundTriggerSession() session %d not registered", session); + return BAD_VALUE; + } + + mSoundTriggerSessions.removeItem(session); + return NO_ERROR; +} + +status_t AudioPolicyManager::addAudioPatch(audio_patch_handle_t handle, + const sp& patch) +{ + ssize_t index = mAudioPatches.indexOfKey(handle); + + if (index >= 0) { + ALOGW("addAudioPatch() patch %d already in", handle); + return ALREADY_EXISTS; + } + mAudioPatches.add(handle, patch); + ALOGV("addAudioPatch() handle %d af handle %d num_sources %d num_sinks %d source handle %d" + "sink handle %d", + handle, patch->mAfPatchHandle, patch->mPatch.num_sources, patch->mPatch.num_sinks, + patch->mPatch.sources[0].id, patch->mPatch.sinks[0].id); + return NO_ERROR; +} + +status_t AudioPolicyManager::removeAudioPatch(audio_patch_handle_t handle) +{ + ssize_t index = mAudioPatches.indexOfKey(handle); + + if (index < 0) { + ALOGW("removeAudioPatch() patch %d not in", handle); + return ALREADY_EXISTS; + } + ALOGV("removeAudioPatch() handle %d af handle %d", handle, + mAudioPatches.valueAt(index)->mAfPatchHandle); + mAudioPatches.removeItemsAt(index); + return NO_ERROR; +} + +// ---------------------------------------------------------------------------- +// AudioPolicyManager +// ---------------------------------------------------------------------------- + +uint32_t AudioPolicyManager::nextUniqueId() +{ + return android_atomic_inc(&mNextUniqueId); +} + +uint32_t AudioPolicyManager::nextAudioPortGeneration() +{ + return android_atomic_inc(&mAudioPortGeneration); +} + +int32_t volatile AudioPolicyManager::mNextUniqueId = 1; + +AudioPolicyManager::AudioPolicyManager(AudioPolicyClientInterface *clientInterface) + : +#ifdef AUDIO_POLICY_TEST + Thread(false), +#endif //AUDIO_POLICY_TEST + mPrimaryOutput((audio_io_handle_t)0), + mPhoneState(AUDIO_MODE_NORMAL), + mLimitRingtoneVolume(false), mLastVoiceVolume(-1.0f), + mTotalEffectsCpuLoad(0), mTotalEffectsMemory(0), + mA2dpSuspended(false), + mSpeakerDrcEnabled(false), + mAudioPortGeneration(1), + mBeaconMuteRefCount(0), + mBeaconPlayingRefCount(0), + mBeaconMuted(false) +{ + mUidCached = getuid(); + mpClientInterface = clientInterface; + + for (int i = 0; i < AUDIO_POLICY_FORCE_USE_CNT; i++) { + mForceUse[i] = AUDIO_POLICY_FORCE_NONE; + } + + mDefaultOutputDevice = new DeviceDescriptor(String8("Speaker"), AUDIO_DEVICE_OUT_SPEAKER); + if (loadAudioPolicyConfig(AUDIO_POLICY_VENDOR_CONFIG_FILE) != NO_ERROR) { + if (loadAudioPolicyConfig(AUDIO_POLICY_CONFIG_FILE) != NO_ERROR) { + ALOGE("could not load audio policy configuration file, setting defaults"); + defaultAudioPolicyConfig(); + } + } + // mAvailableOutputDevices and mAvailableInputDevices now contain all attached devices + + // must be done after reading the policy + initializeVolumeCurves(); + + // open all output streams needed to access attached devices + audio_devices_t outputDeviceTypes = mAvailableOutputDevices.types(); + audio_devices_t inputDeviceTypes = mAvailableInputDevices.types() & ~AUDIO_DEVICE_BIT_IN; + for (size_t i = 0; i < mHwModules.size(); i++) { + mHwModules[i]->mHandle = mpClientInterface->loadHwModule(mHwModules[i]->mName); + if (mHwModules[i]->mHandle == 0) { + ALOGW("could not open HW module %s", mHwModules[i]->mName); + continue; + } + // open all output streams needed to access attached devices + // except for direct output streams that are only opened when they are actually + // required by an app. + // This also validates mAvailableOutputDevices list + for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++) + { + const sp outProfile = mHwModules[i]->mOutputProfiles[j]; + + if (outProfile->mSupportedDevices.isEmpty()) { + ALOGW("Output profile contains no device on module %s", mHwModules[i]->mName); + continue; + } + + if ((outProfile->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) { + continue; + } + audio_devices_t profileType = outProfile->mSupportedDevices.types(); + if ((profileType & mDefaultOutputDevice->mDeviceType) != AUDIO_DEVICE_NONE) { + profileType = mDefaultOutputDevice->mDeviceType; + } else { + // chose first device present in mSupportedDevices also part of + // outputDeviceTypes + for (size_t k = 0; k < outProfile->mSupportedDevices.size(); k++) { + profileType = outProfile->mSupportedDevices[k]->mDeviceType; + if ((profileType & outputDeviceTypes) != 0) { + break; + } + } + } + if ((profileType & outputDeviceTypes) == 0) { + continue; + } + sp outputDesc = new AudioOutputDescriptor(outProfile); + + outputDesc->mDevice = profileType; + audio_config_t config = AUDIO_CONFIG_INITIALIZER; + config.sample_rate = outputDesc->mSamplingRate; + config.channel_mask = outputDesc->mChannelMask; + config.format = outputDesc->mFormat; + audio_io_handle_t output = AUDIO_IO_HANDLE_NONE; + status_t status = mpClientInterface->openOutput(outProfile->mModule->mHandle, + &output, + &config, + &outputDesc->mDevice, + String8(""), + &outputDesc->mLatency, + outputDesc->mFlags); + + if (status != NO_ERROR) { + ALOGW("Cannot open output stream for device %08x on hw module %s", + outputDesc->mDevice, + mHwModules[i]->mName); + } else { + outputDesc->mSamplingRate = config.sample_rate; + outputDesc->mChannelMask = config.channel_mask; + outputDesc->mFormat = config.format; + + for (size_t k = 0; k < outProfile->mSupportedDevices.size(); k++) { + audio_devices_t type = outProfile->mSupportedDevices[k]->mDeviceType; + ssize_t index = + mAvailableOutputDevices.indexOf(outProfile->mSupportedDevices[k]); + // give a valid ID to an attached device once confirmed it is reachable + if (index >= 0 && !mAvailableOutputDevices[index]->isAttached()) { + mAvailableOutputDevices[index]->attach(mHwModules[i]); + } + } + if (mPrimaryOutput == 0 && + outProfile->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY) { + mPrimaryOutput = output; + } + addOutput(output, outputDesc); + setOutputDevice(output, + outputDesc->mDevice, + true); + } + } + // open input streams needed to access attached devices to validate + // mAvailableInputDevices list + for (size_t j = 0; j < mHwModules[i]->mInputProfiles.size(); j++) + { + const sp inProfile = mHwModules[i]->mInputProfiles[j]; + + if (inProfile->mSupportedDevices.isEmpty()) { + ALOGW("Input profile contains no device on module %s", mHwModules[i]->mName); + continue; + } + // chose first device present in mSupportedDevices also part of + // inputDeviceTypes + audio_devices_t profileType = AUDIO_DEVICE_NONE; + for (size_t k = 0; k < inProfile->mSupportedDevices.size(); k++) { + profileType = inProfile->mSupportedDevices[k]->mDeviceType; + if (profileType & inputDeviceTypes) { + break; + } + } + if ((profileType & inputDeviceTypes) == 0) { + continue; + } + sp inputDesc = new AudioInputDescriptor(inProfile); + + inputDesc->mInputSource = AUDIO_SOURCE_MIC; + inputDesc->mDevice = profileType; + + // find the address + DeviceVector inputDevices = mAvailableInputDevices.getDevicesFromType(profileType); + // the inputs vector must be of size 1, but we don't want to crash here + String8 address = inputDevices.size() > 0 ? inputDevices.itemAt(0)->mAddress + : String8(""); + ALOGV(" for input device 0x%x using address %s", profileType, address.string()); + ALOGE_IF(inputDevices.size() == 0, "Input device list is empty!"); + + audio_config_t config = AUDIO_CONFIG_INITIALIZER; + config.sample_rate = inputDesc->mSamplingRate; + config.channel_mask = inputDesc->mChannelMask; + config.format = inputDesc->mFormat; + audio_io_handle_t input = AUDIO_IO_HANDLE_NONE; + status_t status = mpClientInterface->openInput(inProfile->mModule->mHandle, + &input, + &config, + &inputDesc->mDevice, + address, + AUDIO_SOURCE_MIC, + AUDIO_INPUT_FLAG_NONE); + + if (status == NO_ERROR) { + for (size_t k = 0; k < inProfile->mSupportedDevices.size(); k++) { + audio_devices_t type = inProfile->mSupportedDevices[k]->mDeviceType; + ssize_t index = + mAvailableInputDevices.indexOf(inProfile->mSupportedDevices[k]); + // give a valid ID to an attached device once confirmed it is reachable + if (index >= 0 && !mAvailableInputDevices[index]->isAttached()) { + mAvailableInputDevices[index]->attach(mHwModules[i]); + } + } + mpClientInterface->closeInput(input); + } else { + ALOGW("Cannot open input stream for device %08x on hw module %s", + inputDesc->mDevice, + mHwModules[i]->mName); + } + } + } + // make sure all attached devices have been allocated a unique ID + for (size_t i = 0; i < mAvailableOutputDevices.size();) { + if (!mAvailableOutputDevices[i]->isAttached()) { + ALOGW("Input device %08x unreachable", mAvailableOutputDevices[i]->mDeviceType); + mAvailableOutputDevices.remove(mAvailableOutputDevices[i]); + continue; + } + i++; + } + for (size_t i = 0; i < mAvailableInputDevices.size();) { + if (!mAvailableInputDevices[i]->isAttached()) { + ALOGW("Input device %08x unreachable", mAvailableInputDevices[i]->mDeviceType); + mAvailableInputDevices.remove(mAvailableInputDevices[i]); + continue; + } + i++; + } + // make sure default device is reachable + if (mAvailableOutputDevices.indexOf(mDefaultOutputDevice) < 0) { + ALOGE("Default device %08x is unreachable", mDefaultOutputDevice->mDeviceType); + } + + ALOGE_IF((mPrimaryOutput == 0), "Failed to open primary output"); + + updateDevicesAndOutputs(); + +#ifdef AUDIO_POLICY_TEST + if (mPrimaryOutput != 0) { + AudioParameter outputCmd = AudioParameter(); + outputCmd.addInt(String8("set_id"), 0); + mpClientInterface->setParameters(mPrimaryOutput, outputCmd.toString()); + + mTestDevice = AUDIO_DEVICE_OUT_SPEAKER; + mTestSamplingRate = 44100; + mTestFormat = AUDIO_FORMAT_PCM_16_BIT; + mTestChannels = AUDIO_CHANNEL_OUT_STEREO; + mTestLatencyMs = 0; + mCurOutput = 0; + mDirectOutput = false; + for (int i = 0; i < NUM_TEST_OUTPUTS; i++) { + mTestOutputs[i] = 0; + } + + const size_t SIZE = 256; + char buffer[SIZE]; + snprintf(buffer, SIZE, "AudioPolicyManagerTest"); + run(buffer, ANDROID_PRIORITY_AUDIO); + } +#endif //AUDIO_POLICY_TEST +} + +AudioPolicyManager::~AudioPolicyManager() +{ +#ifdef AUDIO_POLICY_TEST + exit(); +#endif //AUDIO_POLICY_TEST + for (size_t i = 0; i < mOutputs.size(); i++) { + mpClientInterface->closeOutput(mOutputs.keyAt(i)); + } + for (size_t i = 0; i < mInputs.size(); i++) { + mpClientInterface->closeInput(mInputs.keyAt(i)); + } + mAvailableOutputDevices.clear(); + mAvailableInputDevices.clear(); + mOutputs.clear(); + mInputs.clear(); + mHwModules.clear(); +} + +status_t AudioPolicyManager::initCheck() +{ + return (mPrimaryOutput == 0) ? NO_INIT : NO_ERROR; +} + +#ifdef AUDIO_POLICY_TEST +bool AudioPolicyManager::threadLoop() +{ + ALOGV("entering threadLoop()"); + while (!exitPending()) + { + String8 command; + int valueInt; + String8 value; + + Mutex::Autolock _l(mLock); + mWaitWorkCV.waitRelative(mLock, milliseconds(50)); + + command = mpClientInterface->getParameters(0, String8("test_cmd_policy")); + AudioParameter param = AudioParameter(command); + + if (param.getInt(String8("test_cmd_policy"), valueInt) == NO_ERROR && + valueInt != 0) { + ALOGV("Test command %s received", command.string()); + String8 target; + if (param.get(String8("target"), target) != NO_ERROR) { + target = "Manager"; + } + if (param.getInt(String8("test_cmd_policy_output"), valueInt) == NO_ERROR) { + param.remove(String8("test_cmd_policy_output")); + mCurOutput = valueInt; + } + if (param.get(String8("test_cmd_policy_direct"), value) == NO_ERROR) { + param.remove(String8("test_cmd_policy_direct")); + if (value == "false") { + mDirectOutput = false; + } else if (value == "true") { + mDirectOutput = true; + } + } + if (param.getInt(String8("test_cmd_policy_input"), valueInt) == NO_ERROR) { + param.remove(String8("test_cmd_policy_input")); + mTestInput = valueInt; + } + + if (param.get(String8("test_cmd_policy_format"), value) == NO_ERROR) { + param.remove(String8("test_cmd_policy_format")); + int format = AUDIO_FORMAT_INVALID; + if (value == "PCM 16 bits") { + format = AUDIO_FORMAT_PCM_16_BIT; + } else if (value == "PCM 8 bits") { + format = AUDIO_FORMAT_PCM_8_BIT; + } else if (value == "Compressed MP3") { + format = AUDIO_FORMAT_MP3; + } + if (format != AUDIO_FORMAT_INVALID) { + if (target == "Manager") { + mTestFormat = format; + } else if (mTestOutputs[mCurOutput] != 0) { + AudioParameter outputParam = AudioParameter(); + outputParam.addInt(String8("format"), format); + mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString()); + } + } + } + if (param.get(String8("test_cmd_policy_channels"), value) == NO_ERROR) { + param.remove(String8("test_cmd_policy_channels")); + int channels = 0; + + if (value == "Channels Stereo") { + channels = AUDIO_CHANNEL_OUT_STEREO; + } else if (value == "Channels Mono") { + channels = AUDIO_CHANNEL_OUT_MONO; + } + if (channels != 0) { + if (target == "Manager") { + mTestChannels = channels; + } else if (mTestOutputs[mCurOutput] != 0) { + AudioParameter outputParam = AudioParameter(); + outputParam.addInt(String8("channels"), channels); + mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString()); + } + } + } + if (param.getInt(String8("test_cmd_policy_sampleRate"), valueInt) == NO_ERROR) { + param.remove(String8("test_cmd_policy_sampleRate")); + if (valueInt >= 0 && valueInt <= 96000) { + int samplingRate = valueInt; + if (target == "Manager") { + mTestSamplingRate = samplingRate; + } else if (mTestOutputs[mCurOutput] != 0) { + AudioParameter outputParam = AudioParameter(); + outputParam.addInt(String8("sampling_rate"), samplingRate); + mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString()); + } + } + } + + if (param.get(String8("test_cmd_policy_reopen"), value) == NO_ERROR) { + param.remove(String8("test_cmd_policy_reopen")); + + sp outputDesc = mOutputs.valueFor(mPrimaryOutput); + mpClientInterface->closeOutput(mPrimaryOutput); + + audio_module_handle_t moduleHandle = outputDesc->mModule->mHandle; + + mOutputs.removeItem(mPrimaryOutput); + + sp outputDesc = new AudioOutputDescriptor(NULL); + outputDesc->mDevice = AUDIO_DEVICE_OUT_SPEAKER; + audio_config_t config = AUDIO_CONFIG_INITIALIZER; + config.sample_rate = outputDesc->mSamplingRate; + config.channel_mask = outputDesc->mChannelMask; + config.format = outputDesc->mFormat; + status_t status = mpClientInterface->openOutput(moduleHandle, + &mPrimaryOutput, + &config, + &outputDesc->mDevice, + String8(""), + &outputDesc->mLatency, + outputDesc->mFlags); + if (status != NO_ERROR) { + ALOGE("Failed to reopen hardware output stream, " + "samplingRate: %d, format %d, channels %d", + outputDesc->mSamplingRate, outputDesc->mFormat, outputDesc->mChannelMask); + } else { + outputDesc->mSamplingRate = config.sample_rate; + outputDesc->mChannelMask = config.channel_mask; + outputDesc->mFormat = config.format; + AudioParameter outputCmd = AudioParameter(); + outputCmd.addInt(String8("set_id"), 0); + mpClientInterface->setParameters(mPrimaryOutput, outputCmd.toString()); + addOutput(mPrimaryOutput, outputDesc); + } + } + + + mpClientInterface->setParameters(0, String8("test_cmd_policy=")); + } + } + return false; +} + +void AudioPolicyManager::exit() +{ + { + AutoMutex _l(mLock); + requestExit(); + mWaitWorkCV.signal(); + } + requestExitAndWait(); +} + +int AudioPolicyManager::testOutputIndex(audio_io_handle_t output) +{ + for (int i = 0; i < NUM_TEST_OUTPUTS; i++) { + if (output == mTestOutputs[i]) return i; + } + return 0; +} +#endif //AUDIO_POLICY_TEST + +// --- + +void AudioPolicyManager::addOutput(audio_io_handle_t output, sp outputDesc) +{ + outputDesc->mIoHandle = output; + outputDesc->mId = nextUniqueId(); + mOutputs.add(output, outputDesc); + nextAudioPortGeneration(); +} + +void AudioPolicyManager::addInput(audio_io_handle_t input, sp inputDesc) +{ + inputDesc->mIoHandle = input; + inputDesc->mId = nextUniqueId(); + mInputs.add(input, inputDesc); + nextAudioPortGeneration(); +} + +void AudioPolicyManager::findIoHandlesByAddress(sp desc /*in*/, + const audio_devices_t device /*in*/, + const String8 address /*in*/, + SortedVector& outputs /*out*/) { + sp devDesc = + desc->mProfile->mSupportedDevices.getDevice(device, address); + if (devDesc != 0) { + ALOGV("findIoHandlesByAddress(): adding opened output %d on same address %s", + desc->mIoHandle, address.string()); + outputs.add(desc->mIoHandle); + } +} + +status_t AudioPolicyManager::checkOutputsForDevice(const sp devDesc, + audio_policy_dev_state_t state, + SortedVector& outputs, + const String8 address) +{ + audio_devices_t device = devDesc->mDeviceType; + sp desc; + // erase all current sample rates, formats and channel masks + devDesc->clearCapabilities(); + + if (state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) { + // first list already open outputs that can be routed to this device + for (size_t i = 0; i < mOutputs.size(); i++) { + desc = mOutputs.valueAt(i); + if (!desc->isDuplicated() && (desc->mProfile->mSupportedDevices.types() & device)) { + if (!deviceDistinguishesOnAddress(device)) { + ALOGV("checkOutputsForDevice(): adding opened output %d", mOutputs.keyAt(i)); + outputs.add(mOutputs.keyAt(i)); + } else { + ALOGV(" checking address match due to device 0x%x", device); + findIoHandlesByAddress(desc, device, address, outputs); + } + } + } + // then look for output profiles that can be routed to this device + SortedVector< sp > profiles; + for (size_t i = 0; i < mHwModules.size(); i++) + { + if (mHwModules[i]->mHandle == 0) { + continue; + } + for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++) + { + sp profile = mHwModules[i]->mOutputProfiles[j]; + if (profile->mSupportedDevices.types() & device) { + if (!deviceDistinguishesOnAddress(device) || + address == profile->mSupportedDevices[0]->mAddress) { + profiles.add(profile); + ALOGV("checkOutputsForDevice(): adding profile %zu from module %zu", j, i); + } + } + } + } + + ALOGV(" found %d profiles, %d outputs", profiles.size(), outputs.size()); + + if (profiles.isEmpty() && outputs.isEmpty()) { + ALOGW("checkOutputsForDevice(): No output available for device %04x", device); + return BAD_VALUE; + } + + // open outputs for matching profiles if needed. Direct outputs are also opened to + // query for dynamic parameters and will be closed later by setDeviceConnectionState() + for (ssize_t profile_index = 0; profile_index < (ssize_t)profiles.size(); profile_index++) { + sp profile = profiles[profile_index]; + + // nothing to do if one output is already opened for this profile + size_t j; + for (j = 0; j < outputs.size(); j++) { + desc = mOutputs.valueFor(outputs.itemAt(j)); + if (!desc->isDuplicated() && desc->mProfile == profile) { + // matching profile: save the sample rates, format and channel masks supported + // by the profile in our device descriptor + devDesc->importAudioPort(profile); + break; + } + } + if (j != outputs.size()) { + continue; + } + + ALOGV("opening output for device %08x with params %s profile %p", + device, address.string(), profile.get()); + desc = new AudioOutputDescriptor(profile); + desc->mDevice = device; + audio_config_t config = AUDIO_CONFIG_INITIALIZER; + config.sample_rate = desc->mSamplingRate; + config.channel_mask = desc->mChannelMask; + config.format = desc->mFormat; + config.offload_info.sample_rate = desc->mSamplingRate; + config.offload_info.channel_mask = desc->mChannelMask; + config.offload_info.format = desc->mFormat; + audio_io_handle_t output = AUDIO_IO_HANDLE_NONE; + status_t status = mpClientInterface->openOutput(profile->mModule->mHandle, + &output, + &config, + &desc->mDevice, + address, + &desc->mLatency, + desc->mFlags); + if (status == NO_ERROR) { + desc->mSamplingRate = config.sample_rate; + desc->mChannelMask = config.channel_mask; + desc->mFormat = config.format; + + // Here is where the out_set_parameters() for card & device gets called + if (!address.isEmpty()) { + char *param = audio_device_address_to_parameter(device, address); + mpClientInterface->setParameters(output, String8(param)); + free(param); + } + + // Here is where we step through and resolve any "dynamic" fields + String8 reply; + char *value; + if (profile->mSamplingRates[0] == 0) { + reply = mpClientInterface->getParameters(output, + String8(AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES)); + ALOGV("checkOutputsForDevice() supported sampling rates %s", + reply.string()); + value = strpbrk((char *)reply.string(), "="); + if (value != NULL) { + profile->loadSamplingRates(value + 1); + } + } + if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) { + reply = mpClientInterface->getParameters(output, + String8(AUDIO_PARAMETER_STREAM_SUP_FORMATS)); + ALOGV("checkOutputsForDevice() supported formats %s", + reply.string()); + value = strpbrk((char *)reply.string(), "="); + if (value != NULL) { + profile->loadFormats(value + 1); + } + } + if (profile->mChannelMasks[0] == 0) { + reply = mpClientInterface->getParameters(output, + String8(AUDIO_PARAMETER_STREAM_SUP_CHANNELS)); + ALOGV("checkOutputsForDevice() supported channel masks %s", + reply.string()); + value = strpbrk((char *)reply.string(), "="); + if (value != NULL) { + profile->loadOutChannels(value + 1); + } + } + if (((profile->mSamplingRates[0] == 0) && + (profile->mSamplingRates.size() < 2)) || + ((profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) && + (profile->mFormats.size() < 2)) || + ((profile->mChannelMasks[0] == 0) && + (profile->mChannelMasks.size() < 2))) { + ALOGW("checkOutputsForDevice() missing param"); + mpClientInterface->closeOutput(output); + output = AUDIO_IO_HANDLE_NONE; + } else if (profile->mSamplingRates[0] == 0 || profile->mFormats[0] == 0 || + profile->mChannelMasks[0] == 0) { + mpClientInterface->closeOutput(output); + config.sample_rate = profile->pickSamplingRate(); + config.channel_mask = profile->pickChannelMask(); + config.format = profile->pickFormat(); + config.offload_info.sample_rate = config.sample_rate; + config.offload_info.channel_mask = config.channel_mask; + config.offload_info.format = config.format; + status = mpClientInterface->openOutput(profile->mModule->mHandle, + &output, + &config, + &desc->mDevice, + address, + &desc->mLatency, + desc->mFlags); + if (status == NO_ERROR) { + desc->mSamplingRate = config.sample_rate; + desc->mChannelMask = config.channel_mask; + desc->mFormat = config.format; + } else { + output = AUDIO_IO_HANDLE_NONE; + } + } + + if (output != AUDIO_IO_HANDLE_NONE) { + addOutput(output, desc); + if (deviceDistinguishesOnAddress(device) && address != "0") { + ssize_t index = mPolicyMixes.indexOfKey(address); + if (index >= 0) { + mPolicyMixes[index]->mOutput = desc; + desc->mPolicyMix = &mPolicyMixes[index]->mMix; + } else { + ALOGE("checkOutputsForDevice() cannot find policy for address %s", + address.string()); + } + } else if ((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) == 0) { + // no duplicated output for direct outputs and + // outputs used by dynamic policy mixes + audio_io_handle_t duplicatedOutput = AUDIO_IO_HANDLE_NONE; + + // set initial stream volume for device + applyStreamVolumes(output, device, 0, true); + + //TODO: configure audio effect output stage here + + // open a duplicating output thread for the new output and the primary output + duplicatedOutput = mpClientInterface->openDuplicateOutput(output, + mPrimaryOutput); + if (duplicatedOutput != AUDIO_IO_HANDLE_NONE) { + // add duplicated output descriptor + sp dupOutputDesc = + new AudioOutputDescriptor(NULL); + dupOutputDesc->mOutput1 = mOutputs.valueFor(mPrimaryOutput); + dupOutputDesc->mOutput2 = mOutputs.valueFor(output); + dupOutputDesc->mSamplingRate = desc->mSamplingRate; + dupOutputDesc->mFormat = desc->mFormat; + dupOutputDesc->mChannelMask = desc->mChannelMask; + dupOutputDesc->mLatency = desc->mLatency; + addOutput(duplicatedOutput, dupOutputDesc); + applyStreamVolumes(duplicatedOutput, device, 0, true); + } else { + ALOGW("checkOutputsForDevice() could not open dup output for %d and %d", + mPrimaryOutput, output); + mpClientInterface->closeOutput(output); + mOutputs.removeItem(output); + nextAudioPortGeneration(); + output = AUDIO_IO_HANDLE_NONE; + } + } + } + } else { + output = AUDIO_IO_HANDLE_NONE; + } + if (output == AUDIO_IO_HANDLE_NONE) { + ALOGW("checkOutputsForDevice() could not open output for device %x", device); + profiles.removeAt(profile_index); + profile_index--; + } else { + outputs.add(output); + devDesc->importAudioPort(profile); + + if (deviceDistinguishesOnAddress(device)) { + ALOGV("checkOutputsForDevice(): setOutputDevice(dev=0x%x, addr=%s)", + device, address.string()); + setOutputDevice(output, device, true/*force*/, 0/*delay*/, + NULL/*patch handle*/, address.string()); + } + ALOGV("checkOutputsForDevice(): adding output %d", output); + } + } + + if (profiles.isEmpty()) { + ALOGW("checkOutputsForDevice(): No output available for device %04x", device); + return BAD_VALUE; + } + } else { // Disconnect + // check if one opened output is not needed any more after disconnecting one device + for (size_t i = 0; i < mOutputs.size(); i++) { + desc = mOutputs.valueAt(i); + if (!desc->isDuplicated()) { + // exact match on device + if (deviceDistinguishesOnAddress(device) && + (desc->mProfile->mSupportedDevices.types() == device)) { + findIoHandlesByAddress(desc, device, address, outputs); + } else if (!(desc->mProfile->mSupportedDevices.types() + & mAvailableOutputDevices.types())) { + ALOGV("checkOutputsForDevice(): disconnecting adding output %d", + mOutputs.keyAt(i)); + outputs.add(mOutputs.keyAt(i)); + } + } + } + // Clear any profiles associated with the disconnected device. + for (size_t i = 0; i < mHwModules.size(); i++) + { + if (mHwModules[i]->mHandle == 0) { + continue; + } + for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++) + { + sp profile = mHwModules[i]->mOutputProfiles[j]; + if (profile->mSupportedDevices.types() & device) { + ALOGV("checkOutputsForDevice(): " + "clearing direct output profile %zu on module %zu", j, i); + if (profile->mSamplingRates[0] == 0) { + profile->mSamplingRates.clear(); + profile->mSamplingRates.add(0); + } + if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) { + profile->mFormats.clear(); + profile->mFormats.add(AUDIO_FORMAT_DEFAULT); + } + if (profile->mChannelMasks[0] == 0) { + profile->mChannelMasks.clear(); + profile->mChannelMasks.add(0); + } + } + } + } + } + return NO_ERROR; +} + +status_t AudioPolicyManager::checkInputsForDevice(audio_devices_t device, + audio_policy_dev_state_t state, + SortedVector& inputs, + const String8 address) +{ + sp desc; + if (state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) { + // first list already open inputs that can be routed to this device + for (size_t input_index = 0; input_index < mInputs.size(); input_index++) { + desc = mInputs.valueAt(input_index); + if (desc->mProfile->mSupportedDevices.types() & (device & ~AUDIO_DEVICE_BIT_IN)) { + ALOGV("checkInputsForDevice(): adding opened input %d", mInputs.keyAt(input_index)); + inputs.add(mInputs.keyAt(input_index)); + } + } + + // then look for input profiles that can be routed to this device + SortedVector< sp > profiles; + for (size_t module_idx = 0; module_idx < mHwModules.size(); module_idx++) + { + if (mHwModules[module_idx]->mHandle == 0) { + continue; + } + for (size_t profile_index = 0; + profile_index < mHwModules[module_idx]->mInputProfiles.size(); + profile_index++) + { + sp profile = mHwModules[module_idx]->mInputProfiles[profile_index]; + + if (profile->mSupportedDevices.types() & (device & ~AUDIO_DEVICE_BIT_IN)) { + if (!deviceDistinguishesOnAddress(device) || + address == profile->mSupportedDevices[0]->mAddress) { + profiles.add(profile); + ALOGV("checkInputsForDevice(): adding profile %zu from module %zu", + profile_index, module_idx); + } + } + } + } + + if (profiles.isEmpty() && inputs.isEmpty()) { + ALOGW("checkInputsForDevice(): No input available for device 0x%X", device); + return BAD_VALUE; + } + + // open inputs for matching profiles if needed. Direct inputs are also opened to + // query for dynamic parameters and will be closed later by setDeviceConnectionState() + for (ssize_t profile_index = 0; profile_index < (ssize_t)profiles.size(); profile_index++) { + + sp profile = profiles[profile_index]; + // nothing to do if one input is already opened for this profile + size_t input_index; + for (input_index = 0; input_index < mInputs.size(); input_index++) { + desc = mInputs.valueAt(input_index); + if (desc->mProfile == profile) { + break; + } + } + if (input_index != mInputs.size()) { + continue; + } + + ALOGV("opening input for device 0x%X with params %s", device, address.string()); + desc = new AudioInputDescriptor(profile); + desc->mDevice = device; + audio_config_t config = AUDIO_CONFIG_INITIALIZER; + config.sample_rate = desc->mSamplingRate; + config.channel_mask = desc->mChannelMask; + config.format = desc->mFormat; + audio_io_handle_t input = AUDIO_IO_HANDLE_NONE; + status_t status = mpClientInterface->openInput(profile->mModule->mHandle, + &input, + &config, + &desc->mDevice, + address, + AUDIO_SOURCE_MIC, + AUDIO_INPUT_FLAG_NONE /*FIXME*/); + + if (status == NO_ERROR) { + desc->mSamplingRate = config.sample_rate; + desc->mChannelMask = config.channel_mask; + desc->mFormat = config.format; + + if (!address.isEmpty()) { + char *param = audio_device_address_to_parameter(device, address); + mpClientInterface->setParameters(input, String8(param)); + free(param); + } + + // Here is where we step through and resolve any "dynamic" fields + String8 reply; + char *value; + if (profile->mSamplingRates[0] == 0) { + reply = mpClientInterface->getParameters(input, + String8(AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES)); + ALOGV("checkInputsForDevice() direct input sup sampling rates %s", + reply.string()); + value = strpbrk((char *)reply.string(), "="); + if (value != NULL) { + profile->loadSamplingRates(value + 1); + } + } + if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) { + reply = mpClientInterface->getParameters(input, + String8(AUDIO_PARAMETER_STREAM_SUP_FORMATS)); + ALOGV("checkInputsForDevice() direct input sup formats %s", reply.string()); + value = strpbrk((char *)reply.string(), "="); + if (value != NULL) { + profile->loadFormats(value + 1); + } + } + if (profile->mChannelMasks[0] == 0) { + reply = mpClientInterface->getParameters(input, + String8(AUDIO_PARAMETER_STREAM_SUP_CHANNELS)); + ALOGV("checkInputsForDevice() direct input sup channel masks %s", + reply.string()); + value = strpbrk((char *)reply.string(), "="); + if (value != NULL) { + profile->loadInChannels(value + 1); + } + } + if (((profile->mSamplingRates[0] == 0) && (profile->mSamplingRates.size() < 2)) || + ((profile->mFormats[0] == 0) && (profile->mFormats.size() < 2)) || + ((profile->mChannelMasks[0] == 0) && (profile->mChannelMasks.size() < 2))) { + ALOGW("checkInputsForDevice() direct input missing param"); + mpClientInterface->closeInput(input); + input = AUDIO_IO_HANDLE_NONE; + } + + if (input != 0) { + addInput(input, desc); + } + } // endif input != 0 + + if (input == AUDIO_IO_HANDLE_NONE) { + ALOGW("checkInputsForDevice() could not open input for device 0x%X", device); + profiles.removeAt(profile_index); + profile_index--; + } else { + inputs.add(input); + ALOGV("checkInputsForDevice(): adding input %d", input); + } + } // end scan profiles + + if (profiles.isEmpty()) { + ALOGW("checkInputsForDevice(): No input available for device 0x%X", device); + return BAD_VALUE; + } + } else { + // Disconnect + // check if one opened input is not needed any more after disconnecting one device + for (size_t input_index = 0; input_index < mInputs.size(); input_index++) { + desc = mInputs.valueAt(input_index); + if (!(desc->mProfile->mSupportedDevices.types() & mAvailableInputDevices.types() & + ~AUDIO_DEVICE_BIT_IN)) { + ALOGV("checkInputsForDevice(): disconnecting adding input %d", + mInputs.keyAt(input_index)); + inputs.add(mInputs.keyAt(input_index)); + } + } + // Clear any profiles associated with the disconnected device. + for (size_t module_index = 0; module_index < mHwModules.size(); module_index++) { + if (mHwModules[module_index]->mHandle == 0) { + continue; + } + for (size_t profile_index = 0; + profile_index < mHwModules[module_index]->mInputProfiles.size(); + profile_index++) { + sp profile = mHwModules[module_index]->mInputProfiles[profile_index]; + if (profile->mSupportedDevices.types() & device & ~AUDIO_DEVICE_BIT_IN) { + ALOGV("checkInputsForDevice(): clearing direct input profile %zu on module %zu", + profile_index, module_index); + if (profile->mSamplingRates[0] == 0) { + profile->mSamplingRates.clear(); + profile->mSamplingRates.add(0); + } + if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) { + profile->mFormats.clear(); + profile->mFormats.add(AUDIO_FORMAT_DEFAULT); + } + if (profile->mChannelMasks[0] == 0) { + profile->mChannelMasks.clear(); + profile->mChannelMasks.add(0); + } + } + } + } + } // end disconnect + + return NO_ERROR; +} + + +void AudioPolicyManager::closeOutput(audio_io_handle_t output) +{ + ALOGV("closeOutput(%d)", output); + + sp outputDesc = mOutputs.valueFor(output); + if (outputDesc == NULL) { + ALOGW("closeOutput() unknown output %d", output); + return; + } + + for (size_t i = 0; i < mPolicyMixes.size(); i++) { + if (mPolicyMixes[i]->mOutput == outputDesc) { + mPolicyMixes[i]->mOutput.clear(); + } + } + + // look for duplicated outputs connected to the output being removed. + for (size_t i = 0; i < mOutputs.size(); i++) { + sp dupOutputDesc = mOutputs.valueAt(i); + if (dupOutputDesc->isDuplicated() && + (dupOutputDesc->mOutput1 == outputDesc || + dupOutputDesc->mOutput2 == outputDesc)) { + sp outputDesc2; + if (dupOutputDesc->mOutput1 == outputDesc) { + outputDesc2 = dupOutputDesc->mOutput2; + } else { + outputDesc2 = dupOutputDesc->mOutput1; + } + // As all active tracks on duplicated output will be deleted, + // and as they were also referenced on the other output, the reference + // count for their stream type must be adjusted accordingly on + // the other output. + for (int j = 0; j < AUDIO_STREAM_CNT; j++) { + int refCount = dupOutputDesc->mRefCount[j]; + outputDesc2->changeRefCount((audio_stream_type_t)j,-refCount); + } + audio_io_handle_t duplicatedOutput = mOutputs.keyAt(i); + ALOGV("closeOutput() closing also duplicated output %d", duplicatedOutput); + + mpClientInterface->closeOutput(duplicatedOutput); + mOutputs.removeItem(duplicatedOutput); + } + } + + nextAudioPortGeneration(); + + ssize_t index = mAudioPatches.indexOfKey(outputDesc->mPatchHandle); + if (index >= 0) { + sp patchDesc = mAudioPatches.valueAt(index); + status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0); + mAudioPatches.removeItemsAt(index); + mpClientInterface->onAudioPatchListUpdate(); + } + + AudioParameter param; + param.add(String8("closing"), String8("true")); + mpClientInterface->setParameters(output, param.toString()); + + mpClientInterface->closeOutput(output); + mOutputs.removeItem(output); + mPreviousOutputs = mOutputs; +} + +void AudioPolicyManager::closeInput(audio_io_handle_t input) +{ + ALOGV("closeInput(%d)", input); + + sp inputDesc = mInputs.valueFor(input); + if (inputDesc == NULL) { + ALOGW("closeInput() unknown input %d", input); + return; + } + + nextAudioPortGeneration(); + + ssize_t index = mAudioPatches.indexOfKey(inputDesc->mPatchHandle); + if (index >= 0) { + sp patchDesc = mAudioPatches.valueAt(index); + status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0); + mAudioPatches.removeItemsAt(index); + mpClientInterface->onAudioPatchListUpdate(); + } + + mpClientInterface->closeInput(input); + mInputs.removeItem(input); +} + +SortedVector AudioPolicyManager::getOutputsForDevice(audio_devices_t device, + DefaultKeyedVector > openOutputs) +{ + SortedVector outputs; + + ALOGVV("getOutputsForDevice() device %04x", device); + for (size_t i = 0; i < openOutputs.size(); i++) { + ALOGVV("output %d isDuplicated=%d device=%04x", + i, openOutputs.valueAt(i)->isDuplicated(), openOutputs.valueAt(i)->supportedDevices()); + if ((device & openOutputs.valueAt(i)->supportedDevices()) == device) { + ALOGVV("getOutputsForDevice() found output %d", openOutputs.keyAt(i)); + outputs.add(openOutputs.keyAt(i)); + } + } + return outputs; +} + +bool AudioPolicyManager::vectorsEqual(SortedVector& outputs1, + SortedVector& outputs2) +{ + if (outputs1.size() != outputs2.size()) { + return false; + } + for (size_t i = 0; i < outputs1.size(); i++) { + if (outputs1[i] != outputs2[i]) { + return false; + } + } + return true; +} + +void AudioPolicyManager::checkOutputForStrategy(routing_strategy strategy) +{ + audio_devices_t oldDevice = getDeviceForStrategy(strategy, true /*fromCache*/); + audio_devices_t newDevice = getDeviceForStrategy(strategy, false /*fromCache*/); + SortedVector srcOutputs = getOutputsForDevice(oldDevice, mPreviousOutputs); + SortedVector dstOutputs = getOutputsForDevice(newDevice, mOutputs); + + // also take into account external policy-related changes: add all outputs which are + // associated with policies in the "before" and "after" output vectors + ALOGVV("checkOutputForStrategy(): policy related outputs"); + for (size_t i = 0 ; i < mPreviousOutputs.size() ; i++) { + const sp desc = mPreviousOutputs.valueAt(i); + if (desc != 0 && desc->mPolicyMix != NULL) { + srcOutputs.add(desc->mIoHandle); + ALOGVV(" previous outputs: adding %d", desc->mIoHandle); + } + } + for (size_t i = 0 ; i < mOutputs.size() ; i++) { + const sp desc = mOutputs.valueAt(i); + if (desc != 0 && desc->mPolicyMix != NULL) { + dstOutputs.add(desc->mIoHandle); + ALOGVV(" new outputs: adding %d", desc->mIoHandle); + } + } + + if (!vectorsEqual(srcOutputs,dstOutputs)) { + ALOGV("checkOutputForStrategy() strategy %d, moving from output %d to output %d", + strategy, srcOutputs[0], dstOutputs[0]); + // mute strategy while moving tracks from one output to another + for (size_t i = 0; i < srcOutputs.size(); i++) { + sp desc = mOutputs.valueFor(srcOutputs[i]); + if (desc->isStrategyActive(strategy)) { + setStrategyMute(strategy, true, srcOutputs[i]); + setStrategyMute(strategy, false, srcOutputs[i], MUTE_TIME_MS, newDevice); + } + } + + // Move effects associated to this strategy from previous output to new output + if (strategy == STRATEGY_MEDIA) { + audio_io_handle_t fxOutput = selectOutputForEffects(dstOutputs); + SortedVector moved; + for (size_t i = 0; i < mEffects.size(); i++) { + sp effectDesc = mEffects.valueAt(i); + if (effectDesc->mSession == AUDIO_SESSION_OUTPUT_MIX && + effectDesc->mIo != fxOutput) { + if (moved.indexOf(effectDesc->mIo) < 0) { + ALOGV("checkOutputForStrategy() moving effect %d to output %d", + mEffects.keyAt(i), fxOutput); + mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, effectDesc->mIo, + fxOutput); + moved.add(effectDesc->mIo); + } + effectDesc->mIo = fxOutput; + } + } + } + // Move tracks associated to this strategy from previous output to new output + for (int i = 0; i < AUDIO_STREAM_CNT; i++) { + if (i == AUDIO_STREAM_PATCH) { + continue; + } + if (getStrategy((audio_stream_type_t)i) == strategy) { + mpClientInterface->invalidateStream((audio_stream_type_t)i); + } + } + } +} + +void AudioPolicyManager::checkOutputForAllStrategies() +{ + if (mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) + checkOutputForStrategy(STRATEGY_ENFORCED_AUDIBLE); + checkOutputForStrategy(STRATEGY_PHONE); + if (mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] != AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) + checkOutputForStrategy(STRATEGY_ENFORCED_AUDIBLE); + checkOutputForStrategy(STRATEGY_SONIFICATION); + checkOutputForStrategy(STRATEGY_SONIFICATION_RESPECTFUL); + checkOutputForStrategy(STRATEGY_ACCESSIBILITY); + checkOutputForStrategy(STRATEGY_MEDIA); + checkOutputForStrategy(STRATEGY_DTMF); + checkOutputForStrategy(STRATEGY_REROUTING); +} + +audio_io_handle_t AudioPolicyManager::getA2dpOutput() +{ + for (size_t i = 0; i < mOutputs.size(); i++) { + sp outputDesc = mOutputs.valueAt(i); + if (!outputDesc->isDuplicated() && outputDesc->device() & AUDIO_DEVICE_OUT_ALL_A2DP) { + return mOutputs.keyAt(i); + } + } + + return 0; +} + +void AudioPolicyManager::checkA2dpSuspend() +{ + audio_io_handle_t a2dpOutput = getA2dpOutput(); + if (a2dpOutput == 0) { + mA2dpSuspended = false; + return; + } + + bool isScoConnected = + ((mAvailableInputDevices.types() & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET & + ~AUDIO_DEVICE_BIT_IN) != 0) || + ((mAvailableOutputDevices.types() & AUDIO_DEVICE_OUT_ALL_SCO) != 0); + // suspend A2DP output if: + // (NOT already suspended) && + // ((SCO device is connected && + // (forced usage for communication || for record is SCO))) || + // (phone state is ringing || in call) + // + // restore A2DP output if: + // (Already suspended) && + // ((SCO device is NOT connected || + // (forced usage NOT for communication && NOT for record is SCO))) && + // (phone state is NOT ringing && NOT in call) + // + if (mA2dpSuspended) { + if ((!isScoConnected || + ((mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION] != AUDIO_POLICY_FORCE_BT_SCO) && + (mForceUse[AUDIO_POLICY_FORCE_FOR_RECORD] != AUDIO_POLICY_FORCE_BT_SCO))) && + ((mPhoneState != AUDIO_MODE_IN_CALL) && + (mPhoneState != AUDIO_MODE_RINGTONE))) { + + mpClientInterface->restoreOutput(a2dpOutput); + mA2dpSuspended = false; + } + } else { + if ((isScoConnected && + ((mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION] == AUDIO_POLICY_FORCE_BT_SCO) || + (mForceUse[AUDIO_POLICY_FORCE_FOR_RECORD] == AUDIO_POLICY_FORCE_BT_SCO))) || + ((mPhoneState == AUDIO_MODE_IN_CALL) || + (mPhoneState == AUDIO_MODE_RINGTONE))) { + + mpClientInterface->suspendOutput(a2dpOutput); + mA2dpSuspended = true; + } + } +} + +audio_devices_t AudioPolicyManager::getNewOutputDevice(audio_io_handle_t output, bool fromCache) +{ + audio_devices_t device = AUDIO_DEVICE_NONE; + + sp outputDesc = mOutputs.valueFor(output); + + ssize_t index = mAudioPatches.indexOfKey(outputDesc->mPatchHandle); + if (index >= 0) { + sp patchDesc = mAudioPatches.valueAt(index); + if (patchDesc->mUid != mUidCached) { + ALOGV("getNewOutputDevice() device %08x forced by patch %d", + outputDesc->device(), outputDesc->mPatchHandle); + return outputDesc->device(); + } + } + + // check the following by order of priority to request a routing change if necessary: + // 1: the strategy enforced audible is active and enforced on the output: + // use device for strategy enforced audible + // 2: we are in call or the strategy phone is active on the output: + // use device for strategy phone + // 3: the strategy for enforced audible is active but not enforced on the output: + // use the device for strategy enforced audible + // 4: the strategy sonification is active on the output: + // use device for strategy sonification + // 5: the strategy "respectful" sonification is active on the output: + // use device for strategy "respectful" sonification + // 6: the strategy accessibility is active on the output: + // use device for strategy accessibility + // 7: the strategy media is active on the output: + // use device for strategy media + // 8: the strategy DTMF is active on the output: + // use device for strategy DTMF + // 9: the strategy for beacon, a.k.a. "transmitted through speaker" is active on the output: + // use device for strategy t-t-s + if (outputDesc->isStrategyActive(STRATEGY_ENFORCED_AUDIBLE) && + mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) { + device = getDeviceForStrategy(STRATEGY_ENFORCED_AUDIBLE, fromCache); + } else if (isInCall() || + outputDesc->isStrategyActive(STRATEGY_PHONE)) { + device = getDeviceForStrategy(STRATEGY_PHONE, fromCache); + } else if (outputDesc->isStrategyActive(STRATEGY_ENFORCED_AUDIBLE)) { + device = getDeviceForStrategy(STRATEGY_ENFORCED_AUDIBLE, fromCache); + } else if (outputDesc->isStrategyActive(STRATEGY_SONIFICATION)) { + device = getDeviceForStrategy(STRATEGY_SONIFICATION, fromCache); + } else if (outputDesc->isStrategyActive(STRATEGY_SONIFICATION_RESPECTFUL)) { + device = getDeviceForStrategy(STRATEGY_SONIFICATION_RESPECTFUL, fromCache); + } else if (outputDesc->isStrategyActive(STRATEGY_ACCESSIBILITY)) { + device = getDeviceForStrategy(STRATEGY_ACCESSIBILITY, fromCache); + } else if (outputDesc->isStrategyActive(STRATEGY_MEDIA)) { + device = getDeviceForStrategy(STRATEGY_MEDIA, fromCache); + } else if (outputDesc->isStrategyActive(STRATEGY_DTMF)) { + device = getDeviceForStrategy(STRATEGY_DTMF, fromCache); + } else if (outputDesc->isStrategyActive(STRATEGY_TRANSMITTED_THROUGH_SPEAKER)) { + device = getDeviceForStrategy(STRATEGY_TRANSMITTED_THROUGH_SPEAKER, fromCache); + } else if (outputDesc->isStrategyActive(STRATEGY_REROUTING)) { + device = getDeviceForStrategy(STRATEGY_REROUTING, fromCache); + } + + ALOGV("getNewOutputDevice() selected device %x", device); + return device; +} + +audio_devices_t AudioPolicyManager::getNewInputDevice(audio_io_handle_t input) +{ + sp inputDesc = mInputs.valueFor(input); + + ssize_t index = mAudioPatches.indexOfKey(inputDesc->mPatchHandle); + if (index >= 0) { + sp patchDesc = mAudioPatches.valueAt(index); + if (patchDesc->mUid != mUidCached) { + ALOGV("getNewInputDevice() device %08x forced by patch %d", + inputDesc->mDevice, inputDesc->mPatchHandle); + return inputDesc->mDevice; + } + } + + audio_devices_t device = getDeviceAndMixForInputSource(inputDesc->mInputSource); + + ALOGV("getNewInputDevice() selected device %x", device); + return device; +} + +uint32_t AudioPolicyManager::getStrategyForStream(audio_stream_type_t stream) { + return (uint32_t)getStrategy(stream); +} + +audio_devices_t AudioPolicyManager::getDevicesForStream(audio_stream_type_t stream) { + // By checking the range of stream before calling getStrategy, we avoid + // getStrategy's behavior for invalid streams. getStrategy would do a ALOGE + // and then return STRATEGY_MEDIA, but we want to return the empty set. + if (stream < (audio_stream_type_t) 0 || stream >= AUDIO_STREAM_PUBLIC_CNT) { + return AUDIO_DEVICE_NONE; + } + audio_devices_t devices; + routing_strategy strategy = getStrategy(stream); + devices = getDeviceForStrategy(strategy, true /*fromCache*/); + SortedVector outputs = getOutputsForDevice(devices, mOutputs); + for (size_t i = 0; i < outputs.size(); i++) { + sp outputDesc = mOutputs.valueFor(outputs[i]); + if (outputDesc->isStrategyActive(strategy)) { + devices = outputDesc->device(); + break; + } + } + + /*Filter SPEAKER_SAFE out of results, as AudioService doesn't know about it + and doesn't really need to.*/ + if (devices & AUDIO_DEVICE_OUT_SPEAKER_SAFE) { + devices |= AUDIO_DEVICE_OUT_SPEAKER; + devices &= ~AUDIO_DEVICE_OUT_SPEAKER_SAFE; + } + + return devices; +} + +routing_strategy AudioPolicyManager::getStrategy( + audio_stream_type_t stream) { + + ALOG_ASSERT(stream != AUDIO_STREAM_PATCH,"getStrategy() called for AUDIO_STREAM_PATCH"); + + // stream to strategy mapping + switch (stream) { + case AUDIO_STREAM_VOICE_CALL: + case AUDIO_STREAM_BLUETOOTH_SCO: + return STRATEGY_PHONE; + case AUDIO_STREAM_RING: + case AUDIO_STREAM_ALARM: + return STRATEGY_SONIFICATION; + case AUDIO_STREAM_NOTIFICATION: + return STRATEGY_SONIFICATION_RESPECTFUL; + case AUDIO_STREAM_DTMF: + return STRATEGY_DTMF; + default: + ALOGE("unknown stream type %d", stream); + case AUDIO_STREAM_SYSTEM: + // NOTE: SYSTEM stream uses MEDIA strategy because muting music and switching outputs + // while key clicks are played produces a poor result + case AUDIO_STREAM_MUSIC: + return STRATEGY_MEDIA; + case AUDIO_STREAM_ENFORCED_AUDIBLE: + return STRATEGY_ENFORCED_AUDIBLE; + case AUDIO_STREAM_TTS: + return STRATEGY_TRANSMITTED_THROUGH_SPEAKER; + case AUDIO_STREAM_ACCESSIBILITY: + return STRATEGY_ACCESSIBILITY; + case AUDIO_STREAM_REROUTING: + return STRATEGY_REROUTING; + } +} + +uint32_t AudioPolicyManager::getStrategyForAttr(const audio_attributes_t *attr) { + // flags to strategy mapping + if ((attr->flags & AUDIO_FLAG_BEACON) == AUDIO_FLAG_BEACON) { + return (uint32_t) STRATEGY_TRANSMITTED_THROUGH_SPEAKER; + } + if ((attr->flags & AUDIO_FLAG_AUDIBILITY_ENFORCED) == AUDIO_FLAG_AUDIBILITY_ENFORCED) { + return (uint32_t) STRATEGY_ENFORCED_AUDIBLE; + } + + // usage to strategy mapping + switch (attr->usage) { + case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY: + if (isStreamActive(AUDIO_STREAM_RING) || isStreamActive(AUDIO_STREAM_ALARM)) { + return (uint32_t) STRATEGY_SONIFICATION; + } + if (isInCall()) { + return (uint32_t) STRATEGY_PHONE; + } + return (uint32_t) STRATEGY_ACCESSIBILITY; + + case AUDIO_USAGE_MEDIA: + case AUDIO_USAGE_GAME: + case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE: + case AUDIO_USAGE_ASSISTANCE_SONIFICATION: + return (uint32_t) STRATEGY_MEDIA; + + case AUDIO_USAGE_VOICE_COMMUNICATION: + return (uint32_t) STRATEGY_PHONE; + + case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING: + return (uint32_t) STRATEGY_DTMF; + + case AUDIO_USAGE_ALARM: + case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE: + return (uint32_t) STRATEGY_SONIFICATION; + + case AUDIO_USAGE_NOTIFICATION: + case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST: + case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT: + case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED: + case AUDIO_USAGE_NOTIFICATION_EVENT: + return (uint32_t) STRATEGY_SONIFICATION_RESPECTFUL; + + case AUDIO_USAGE_UNKNOWN: + default: + return (uint32_t) STRATEGY_MEDIA; + } +} + +void AudioPolicyManager::handleNotificationRoutingForStream(audio_stream_type_t stream) { + switch(stream) { + case AUDIO_STREAM_MUSIC: + checkOutputForStrategy(STRATEGY_SONIFICATION_RESPECTFUL); + updateDevicesAndOutputs(); + break; + default: + break; + } +} + +bool AudioPolicyManager::isAnyOutputActive(audio_stream_type_t streamToIgnore) { + for (size_t s = 0 ; s < AUDIO_STREAM_CNT ; s++) { + if (s == (size_t) streamToIgnore) { + continue; + } + for (size_t i = 0; i < mOutputs.size(); i++) { + const sp outputDesc = mOutputs.valueAt(i); + if (outputDesc->mRefCount[s] != 0) { + return true; + } + } + } + return false; +} + +uint32_t AudioPolicyManager::handleEventForBeacon(int event) { + switch(event) { + case STARTING_OUTPUT: + mBeaconMuteRefCount++; + break; + case STOPPING_OUTPUT: + if (mBeaconMuteRefCount > 0) { + mBeaconMuteRefCount--; + } + break; + case STARTING_BEACON: + mBeaconPlayingRefCount++; + break; + case STOPPING_BEACON: + if (mBeaconPlayingRefCount > 0) { + mBeaconPlayingRefCount--; + } + break; + } + + if (mBeaconMuteRefCount > 0) { + // any playback causes beacon to be muted + return setBeaconMute(true); + } else { + // no other playback: unmute when beacon starts playing, mute when it stops + return setBeaconMute(mBeaconPlayingRefCount == 0); + } +} + +uint32_t AudioPolicyManager::setBeaconMute(bool mute) { + ALOGV("setBeaconMute(%d) mBeaconMuteRefCount=%d mBeaconPlayingRefCount=%d", + mute, mBeaconMuteRefCount, mBeaconPlayingRefCount); + // keep track of muted state to avoid repeating mute/unmute operations + if (mBeaconMuted != mute) { + // mute/unmute AUDIO_STREAM_TTS on all outputs + ALOGV("\t muting %d", mute); + uint32_t maxLatency = 0; + for (size_t i = 0; i < mOutputs.size(); i++) { + sp desc = mOutputs.valueAt(i); + setStreamMute(AUDIO_STREAM_TTS, mute/*on*/, + desc->mIoHandle, + 0 /*delay*/, AUDIO_DEVICE_NONE); + const uint32_t latency = desc->latency() * 2; + if (latency > maxLatency) { + maxLatency = latency; + } + } + mBeaconMuted = mute; + return maxLatency; + } + return 0; +} + +audio_devices_t AudioPolicyManager::getDeviceForStrategy(routing_strategy strategy, + bool fromCache) +{ + uint32_t device = AUDIO_DEVICE_NONE; + + if (fromCache) { + ALOGVV("getDeviceForStrategy() from cache strategy %d, device %x", + strategy, mDeviceForStrategy[strategy]); + return mDeviceForStrategy[strategy]; + } + audio_devices_t availableOutputDeviceTypes = mAvailableOutputDevices.types(); + switch (strategy) { + + case STRATEGY_TRANSMITTED_THROUGH_SPEAKER: + device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER; + if (!device) { + ALOGE("getDeviceForStrategy() no device found for "\ + "STRATEGY_TRANSMITTED_THROUGH_SPEAKER"); + } + break; + + case STRATEGY_SONIFICATION_RESPECTFUL: + if (isInCall()) { + device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/); + } else if (isStreamActiveRemotely(AUDIO_STREAM_MUSIC, + SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY)) { + // while media is playing on a remote device, use the the sonification behavior. + // Note that we test this usecase before testing if media is playing because + // the isStreamActive() method only informs about the activity of a stream, not + // if it's for local playback. Note also that we use the same delay between both tests + device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/); + //user "safe" speaker if available instead of normal speaker to avoid triggering + //other acoustic safety mechanisms for notification + if (device == AUDIO_DEVICE_OUT_SPEAKER && (availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER_SAFE)) + device = AUDIO_DEVICE_OUT_SPEAKER_SAFE; + } else if (isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY)) { + // while media is playing (or has recently played), use the same device + device = getDeviceForStrategy(STRATEGY_MEDIA, false /*fromCache*/); + } else { + // when media is not playing anymore, fall back on the sonification behavior + device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/); + //user "safe" speaker if available instead of normal speaker to avoid triggering + //other acoustic safety mechanisms for notification + if (device == AUDIO_DEVICE_OUT_SPEAKER && (availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER_SAFE)) + device = AUDIO_DEVICE_OUT_SPEAKER_SAFE; + } + + break; + + case STRATEGY_DTMF: + if (!isInCall()) { + // when off call, DTMF strategy follows the same rules as MEDIA strategy + device = getDeviceForStrategy(STRATEGY_MEDIA, false /*fromCache*/); + break; + } + // when in call, DTMF and PHONE strategies follow the same rules + // FALL THROUGH + + case STRATEGY_PHONE: + // Force use of only devices on primary output if: + // - in call AND + // - cannot route from voice call RX OR + // - audio HAL version is < 3.0 and TX device is on the primary HW module + if (mPhoneState == AUDIO_MODE_IN_CALL) { + audio_devices_t txDevice = + getDeviceAndMixForInputSource(AUDIO_SOURCE_VOICE_COMMUNICATION); + sp hwOutputDesc = mOutputs.valueFor(mPrimaryOutput); + if (((mAvailableInputDevices.types() & + AUDIO_DEVICE_IN_TELEPHONY_RX & ~AUDIO_DEVICE_BIT_IN) == 0) || + (((txDevice & availablePrimaryInputDevices() & ~AUDIO_DEVICE_BIT_IN) != 0) && + (hwOutputDesc->getAudioPort()->mModule->mHalVersion < + AUDIO_DEVICE_API_VERSION_3_0))) { + availableOutputDeviceTypes = availablePrimaryOutputDevices(); + } + } + // for phone strategy, we first consider the forced use and then the available devices by order + // of priority + switch (mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION]) { + case AUDIO_POLICY_FORCE_BT_SCO: + if (!isInCall() || strategy != STRATEGY_DTMF) { + device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT; + if (device) break; + } + device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET; + if (device) break; + device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_SCO; + if (device) break; + // if SCO device is requested but no SCO device is available, fall back to default case + // FALL THROUGH + + default: // FORCE_NONE + // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to A2DP + if (!isInCall() && + (mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] != AUDIO_POLICY_FORCE_NO_BT_A2DP) && + (getA2dpOutput() != 0)) { + device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP; + if (device) break; + device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES; + if (device) break; + } + device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_WIRED_HEADPHONE; + if (device) break; + device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_WIRED_HEADSET; + if (device) break; + device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_DEVICE; + if (device) break; + if (mPhoneState != AUDIO_MODE_IN_CALL) { + device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_ACCESSORY; + if (device) break; + device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET; + if (device) break; + device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_AUX_DIGITAL; + if (device) break; + device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET; + if (device) break; + } + device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_EARPIECE; + if (device) break; + device = mDefaultOutputDevice->mDeviceType; + if (device == AUDIO_DEVICE_NONE) { + ALOGE("getDeviceForStrategy() no device found for STRATEGY_PHONE"); + } + break; + + case AUDIO_POLICY_FORCE_SPEAKER: + // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to + // A2DP speaker when forcing to speaker output + if (!isInCall() && + (mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] != AUDIO_POLICY_FORCE_NO_BT_A2DP) && + (getA2dpOutput() != 0)) { + device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER; + if (device) break; + } + if (!isInCall()) { + device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_ACCESSORY; + if (device) break; + device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_DEVICE; + if (device) break; + device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET; + if (device) break; + device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_AUX_DIGITAL; + if (device) break; + device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET; + if (device) break; + } + device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_LINE; + if (device) break; + device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER; + if (device) break; + device = mDefaultOutputDevice->mDeviceType; + if (device == AUDIO_DEVICE_NONE) { + ALOGE("getDeviceForStrategy() no device found for STRATEGY_PHONE, FORCE_SPEAKER"); + } + break; + } + break; + + case STRATEGY_SONIFICATION: + + // If incall, just select the STRATEGY_PHONE device: The rest of the behavior is handled by + // handleIncallSonification(). + if (isInCall()) { + device = getDeviceForStrategy(STRATEGY_PHONE, false /*fromCache*/); + break; + } + // FALL THROUGH + + case STRATEGY_ENFORCED_AUDIBLE: + // strategy STRATEGY_ENFORCED_AUDIBLE uses same routing policy as STRATEGY_SONIFICATION + // except: + // - when in call where it doesn't default to STRATEGY_PHONE behavior + // - in countries where not enforced in which case it follows STRATEGY_MEDIA + + if ((strategy == STRATEGY_SONIFICATION) || + (mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED)) { + device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER; + if (device == AUDIO_DEVICE_NONE) { + ALOGE("getDeviceForStrategy() speaker device not found for STRATEGY_SONIFICATION"); + } + } + // The second device used for sonification is the same as the device used by media strategy + // FALL THROUGH + + // FIXME: STRATEGY_ACCESSIBILITY and STRATEGY_REROUTING follow STRATEGY_MEDIA for now + case STRATEGY_ACCESSIBILITY: + if (strategy == STRATEGY_ACCESSIBILITY) { + // do not route accessibility prompts to a digital output currently configured with a + // compressed format as they would likely not be mixed and dropped. + for (size_t i = 0; i < mOutputs.size(); i++) { + sp desc = mOutputs.valueAt(i); + audio_devices_t devices = desc->device() & + (AUDIO_DEVICE_OUT_HDMI | AUDIO_DEVICE_OUT_SPDIF | AUDIO_DEVICE_OUT_HDMI_ARC); + if (desc->isActive() && !audio_is_linear_pcm(desc->mFormat) && + devices != AUDIO_DEVICE_NONE) { + availableOutputDeviceTypes = availableOutputDeviceTypes & ~devices; + } + } + } + // FALL THROUGH + + case STRATEGY_REROUTING: + case STRATEGY_MEDIA: { + uint32_t device2 = AUDIO_DEVICE_NONE; + if (strategy != STRATEGY_SONIFICATION) { + // no sonification on remote submix (e.g. WFD) + if (mAvailableOutputDevices.getDevice(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, String8("0")) != 0) { + device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_REMOTE_SUBMIX; + } + } + if ((device2 == AUDIO_DEVICE_NONE) && + (mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] != AUDIO_POLICY_FORCE_NO_BT_A2DP) && + (getA2dpOutput() != 0)) { + device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP; + if (device2 == AUDIO_DEVICE_NONE) { + device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES; + } + if (device2 == AUDIO_DEVICE_NONE) { + device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER; + } + } + if ((device2 == AUDIO_DEVICE_NONE) && + (mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] == AUDIO_POLICY_FORCE_SPEAKER)) { + device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER; + } + if (device2 == AUDIO_DEVICE_NONE) { + device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_WIRED_HEADPHONE; + } + if ((device2 == AUDIO_DEVICE_NONE)) { + device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_LINE; + } + if (device2 == AUDIO_DEVICE_NONE) { + device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_WIRED_HEADSET; + } + if (device2 == AUDIO_DEVICE_NONE) { + device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_ACCESSORY; + } + if (device2 == AUDIO_DEVICE_NONE) { + device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_DEVICE; + } + if (device2 == AUDIO_DEVICE_NONE) { + device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET; + } + if ((device2 == AUDIO_DEVICE_NONE) && (strategy != STRATEGY_SONIFICATION)) { + // no sonification on aux digital (e.g. HDMI) + device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_AUX_DIGITAL; + } + if ((device2 == AUDIO_DEVICE_NONE) && + (mForceUse[AUDIO_POLICY_FORCE_FOR_DOCK] == AUDIO_POLICY_FORCE_ANALOG_DOCK)) { + device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET; + } + if (device2 == AUDIO_DEVICE_NONE) { + device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER; + } + int device3 = AUDIO_DEVICE_NONE; + if (strategy == STRATEGY_MEDIA) { + // ARC, SPDIF and AUX_LINE can co-exist with others. + device3 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_HDMI_ARC; + device3 |= (availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPDIF); + device3 |= (availableOutputDeviceTypes & AUDIO_DEVICE_OUT_AUX_LINE); + } + + device2 |= device3; + // device is DEVICE_OUT_SPEAKER if we come from case STRATEGY_SONIFICATION or + // STRATEGY_ENFORCED_AUDIBLE, AUDIO_DEVICE_NONE otherwise + device |= device2; + + // If hdmi system audio mode is on, remove speaker out of output list. + if ((strategy == STRATEGY_MEDIA) && + (mForceUse[AUDIO_POLICY_FORCE_FOR_HDMI_SYSTEM_AUDIO] == + AUDIO_POLICY_FORCE_HDMI_SYSTEM_AUDIO_ENFORCED)) { + device &= ~AUDIO_DEVICE_OUT_SPEAKER; + } + + if (device) break; + device = mDefaultOutputDevice->mDeviceType; + if (device == AUDIO_DEVICE_NONE) { + ALOGE("getDeviceForStrategy() no device found for STRATEGY_MEDIA"); + } + } break; + + default: + ALOGW("getDeviceForStrategy() unknown strategy: %d", strategy); + break; + } + + ALOGVV("getDeviceForStrategy() strategy %d, device %x", strategy, device); + return device; +} + +void AudioPolicyManager::updateDevicesAndOutputs() +{ + for (int i = 0; i < NUM_STRATEGIES; i++) { + mDeviceForStrategy[i] = getDeviceForStrategy((routing_strategy)i, false /*fromCache*/); + } + mPreviousOutputs = mOutputs; +} + +uint32_t AudioPolicyManager::checkDeviceMuteStrategies(sp outputDesc, + audio_devices_t prevDevice, + uint32_t delayMs) +{ + // mute/unmute strategies using an incompatible device combination + // if muting, wait for the audio in pcm buffer to be drained before proceeding + // if unmuting, unmute only after the specified delay + if (outputDesc->isDuplicated()) { + return 0; + } + + uint32_t muteWaitMs = 0; + audio_devices_t device = outputDesc->device(); + bool shouldMute = outputDesc->isActive() && (popcount(device) >= 2); + + for (size_t i = 0; i < NUM_STRATEGIES; i++) { + audio_devices_t curDevice = getDeviceForStrategy((routing_strategy)i, false /*fromCache*/); + curDevice = curDevice & outputDesc->mProfile->mSupportedDevices.types(); + bool mute = shouldMute && (curDevice & device) && (curDevice != device); + bool doMute = false; + + if (mute && !outputDesc->mStrategyMutedByDevice[i]) { + doMute = true; + outputDesc->mStrategyMutedByDevice[i] = true; + } else if (!mute && outputDesc->mStrategyMutedByDevice[i]){ + doMute = true; + outputDesc->mStrategyMutedByDevice[i] = false; + } + if (doMute) { + for (size_t j = 0; j < mOutputs.size(); j++) { + sp desc = mOutputs.valueAt(j); + // skip output if it does not share any device with current output + if ((desc->supportedDevices() & outputDesc->supportedDevices()) + == AUDIO_DEVICE_NONE) { + continue; + } + audio_io_handle_t curOutput = mOutputs.keyAt(j); + ALOGVV("checkDeviceMuteStrategies() %s strategy %d (curDevice %04x) on output %d", + mute ? "muting" : "unmuting", i, curDevice, curOutput); + setStrategyMute((routing_strategy)i, mute, curOutput, mute ? 0 : delayMs); + if (desc->isStrategyActive((routing_strategy)i)) { + if (mute) { + // FIXME: should not need to double latency if volume could be applied + // immediately by the audioflinger mixer. We must account for the delay + // between now and the next time the audioflinger thread for this output + // will process a buffer (which corresponds to one buffer size, + // usually 1/2 or 1/4 of the latency). + if (muteWaitMs < desc->latency() * 2) { + muteWaitMs = desc->latency() * 2; + } + } + } + } + } + } + + // temporary mute output if device selection changes to avoid volume bursts due to + // different per device volumes + if (outputDesc->isActive() && (device != prevDevice)) { + if (muteWaitMs < outputDesc->latency() * 2) { + muteWaitMs = outputDesc->latency() * 2; + } + for (size_t i = 0; i < NUM_STRATEGIES; i++) { + if (outputDesc->isStrategyActive((routing_strategy)i)) { + setStrategyMute((routing_strategy)i, true, outputDesc->mIoHandle); + // do tempMute unmute after twice the mute wait time + setStrategyMute((routing_strategy)i, false, outputDesc->mIoHandle, + muteWaitMs *2, device); + } + } + } + + // wait for the PCM output buffers to empty before proceeding with the rest of the command + if (muteWaitMs > delayMs) { + muteWaitMs -= delayMs; + usleep(muteWaitMs * 1000); + return muteWaitMs; + } + return 0; +} + +uint32_t AudioPolicyManager::setOutputDevice(audio_io_handle_t output, + audio_devices_t device, + bool force, + int delayMs, + audio_patch_handle_t *patchHandle, + const char* address) +{ + ALOGV("setOutputDevice() output %d device %04x delayMs %d", output, device, delayMs); + sp outputDesc = mOutputs.valueFor(output); + AudioParameter param; + uint32_t muteWaitMs; + + if (outputDesc->isDuplicated()) { + muteWaitMs = setOutputDevice(outputDesc->mOutput1->mIoHandle, device, force, delayMs); + muteWaitMs += setOutputDevice(outputDesc->mOutput2->mIoHandle, device, force, delayMs); + return muteWaitMs; + } + // no need to proceed if new device is not AUDIO_DEVICE_NONE and not supported by current + // output profile + if ((device != AUDIO_DEVICE_NONE) && + ((device & outputDesc->mProfile->mSupportedDevices.types()) == 0)) { + return 0; + } + + // filter devices according to output selected + device = (audio_devices_t)(device & outputDesc->mProfile->mSupportedDevices.types()); + + audio_devices_t prevDevice = outputDesc->mDevice; + + ALOGV("setOutputDevice() prevDevice %04x", prevDevice); + + if (device != AUDIO_DEVICE_NONE) { + outputDesc->mDevice = device; + } + muteWaitMs = checkDeviceMuteStrategies(outputDesc, prevDevice, delayMs); + + // Do not change the routing if: + // the requested device is AUDIO_DEVICE_NONE + // OR the requested device is the same as current device + // AND force is not specified + // AND the output is connected by a valid audio patch. + // Doing this check here allows the caller to call setOutputDevice() without conditions + if ((device == AUDIO_DEVICE_NONE || device == prevDevice) && !force && + outputDesc->mPatchHandle != 0) { + ALOGV("setOutputDevice() setting same device %04x or null device for output %d", + device, output); + return muteWaitMs; + } + + ALOGV("setOutputDevice() changing device"); + + // do the routing + if (device == AUDIO_DEVICE_NONE) { + resetOutputDevice(output, delayMs, NULL); + } else { + DeviceVector deviceList = (address == NULL) ? + mAvailableOutputDevices.getDevicesFromType(device) + : mAvailableOutputDevices.getDevicesFromTypeAddr(device, String8(address)); + if (!deviceList.isEmpty()) { + struct audio_patch patch; + outputDesc->toAudioPortConfig(&patch.sources[0]); + patch.num_sources = 1; + patch.num_sinks = 0; + for (size_t i = 0; i < deviceList.size() && i < AUDIO_PATCH_PORTS_MAX; i++) { + deviceList.itemAt(i)->toAudioPortConfig(&patch.sinks[i]); + patch.num_sinks++; + } + ssize_t index; + if (patchHandle && *patchHandle != AUDIO_PATCH_HANDLE_NONE) { + index = mAudioPatches.indexOfKey(*patchHandle); + } else { + index = mAudioPatches.indexOfKey(outputDesc->mPatchHandle); + } + sp< AudioPatch> patchDesc; + audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE; + if (index >= 0) { + patchDesc = mAudioPatches.valueAt(index); + afPatchHandle = patchDesc->mAfPatchHandle; + } + + status_t status = mpClientInterface->createAudioPatch(&patch, + &afPatchHandle, + delayMs); + ALOGV("setOutputDevice() createAudioPatch returned %d patchHandle %d" + "num_sources %d num_sinks %d", + status, afPatchHandle, patch.num_sources, patch.num_sinks); + if (status == NO_ERROR) { + if (index < 0) { + patchDesc = new AudioPatch((audio_patch_handle_t)nextUniqueId(), + &patch, mUidCached); + addAudioPatch(patchDesc->mHandle, patchDesc); + } else { + patchDesc->mPatch = patch; + } + patchDesc->mAfPatchHandle = afPatchHandle; + patchDesc->mUid = mUidCached; + if (patchHandle) { + *patchHandle = patchDesc->mHandle; + } + outputDesc->mPatchHandle = patchDesc->mHandle; + nextAudioPortGeneration(); + mpClientInterface->onAudioPatchListUpdate(); + } + } + + // inform all input as well + for (size_t i = 0; i < mInputs.size(); i++) { + const sp inputDescriptor = mInputs.valueAt(i); + if (!isVirtualInputDevice(inputDescriptor->mDevice)) { + AudioParameter inputCmd = AudioParameter(); + ALOGV("%s: inform input %d of device:%d", __func__, + inputDescriptor->mIoHandle, device); + inputCmd.addInt(String8(AudioParameter::keyRouting),device); + mpClientInterface->setParameters(inputDescriptor->mIoHandle, + inputCmd.toString(), + delayMs); + } + } + } + + // update stream volumes according to new device + applyStreamVolumes(output, device, delayMs); + + return muteWaitMs; +} + +status_t AudioPolicyManager::resetOutputDevice(audio_io_handle_t output, + int delayMs, + audio_patch_handle_t *patchHandle) +{ + sp outputDesc = mOutputs.valueFor(output); + ssize_t index; + if (patchHandle) { + index = mAudioPatches.indexOfKey(*patchHandle); + } else { + index = mAudioPatches.indexOfKey(outputDesc->mPatchHandle); + } + if (index < 0) { + return INVALID_OPERATION; + } + sp< AudioPatch> patchDesc = mAudioPatches.valueAt(index); + status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, delayMs); + ALOGV("resetOutputDevice() releaseAudioPatch returned %d", status); + outputDesc->mPatchHandle = 0; + removeAudioPatch(patchDesc->mHandle); + nextAudioPortGeneration(); + mpClientInterface->onAudioPatchListUpdate(); + return status; +} + +status_t AudioPolicyManager::setInputDevice(audio_io_handle_t input, + audio_devices_t device, + bool force, + audio_patch_handle_t *patchHandle) +{ + status_t status = NO_ERROR; + + sp inputDesc = mInputs.valueFor(input); + if ((device != AUDIO_DEVICE_NONE) && ((device != inputDesc->mDevice) || force)) { + inputDesc->mDevice = device; + + DeviceVector deviceList = mAvailableInputDevices.getDevicesFromType(device); + if (!deviceList.isEmpty()) { + struct audio_patch patch; + inputDesc->toAudioPortConfig(&patch.sinks[0]); + // AUDIO_SOURCE_HOTWORD is for internal use only: + // handled as AUDIO_SOURCE_VOICE_RECOGNITION by the audio HAL + if (patch.sinks[0].ext.mix.usecase.source == AUDIO_SOURCE_HOTWORD && + !inputDesc->mIsSoundTrigger) { + patch.sinks[0].ext.mix.usecase.source = AUDIO_SOURCE_VOICE_RECOGNITION; + } + patch.num_sinks = 1; + //only one input device for now + deviceList.itemAt(0)->toAudioPortConfig(&patch.sources[0]); + patch.num_sources = 1; + ssize_t index; + if (patchHandle && *patchHandle != AUDIO_PATCH_HANDLE_NONE) { + index = mAudioPatches.indexOfKey(*patchHandle); + } else { + index = mAudioPatches.indexOfKey(inputDesc->mPatchHandle); + } + sp< AudioPatch> patchDesc; + audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE; + if (index >= 0) { + patchDesc = mAudioPatches.valueAt(index); + afPatchHandle = patchDesc->mAfPatchHandle; + } + + status_t status = mpClientInterface->createAudioPatch(&patch, + &afPatchHandle, + 0); + ALOGV("setInputDevice() createAudioPatch returned %d patchHandle %d", + status, afPatchHandle); + if (status == NO_ERROR) { + if (index < 0) { + patchDesc = new AudioPatch((audio_patch_handle_t)nextUniqueId(), + &patch, mUidCached); + addAudioPatch(patchDesc->mHandle, patchDesc); + } else { + patchDesc->mPatch = patch; + } + patchDesc->mAfPatchHandle = afPatchHandle; + patchDesc->mUid = mUidCached; + if (patchHandle) { + *patchHandle = patchDesc->mHandle; + } + inputDesc->mPatchHandle = patchDesc->mHandle; + nextAudioPortGeneration(); + mpClientInterface->onAudioPatchListUpdate(); + } + } + } + return status; +} + +status_t AudioPolicyManager::resetInputDevice(audio_io_handle_t input, + audio_patch_handle_t *patchHandle) +{ + sp inputDesc = mInputs.valueFor(input); + ssize_t index; + if (patchHandle) { + index = mAudioPatches.indexOfKey(*patchHandle); + } else { + index = mAudioPatches.indexOfKey(inputDesc->mPatchHandle); + } + if (index < 0) { + return INVALID_OPERATION; + } + sp< AudioPatch> patchDesc = mAudioPatches.valueAt(index); + status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0); + ALOGV("resetInputDevice() releaseAudioPatch returned %d", status); + inputDesc->mPatchHandle = 0; + removeAudioPatch(patchDesc->mHandle); + nextAudioPortGeneration(); + mpClientInterface->onAudioPatchListUpdate(); + return status; +} + +sp AudioPolicyManager::getInputProfile(audio_devices_t device, + String8 address, + uint32_t& samplingRate, + audio_format_t format, + audio_channel_mask_t channelMask, + audio_input_flags_t flags) +{ + // Choose an input profile based on the requested capture parameters: select the first available + // profile supporting all requested parameters. + + for (size_t i = 0; i < mHwModules.size(); i++) + { + if (mHwModules[i]->mHandle == 0) { + continue; + } + for (size_t j = 0; j < mHwModules[i]->mInputProfiles.size(); j++) + { + sp profile = mHwModules[i]->mInputProfiles[j]; + // profile->log(); + if (profile->isCompatibleProfile(device, address, samplingRate, + &samplingRate /*updatedSamplingRate*/, + format, channelMask, (audio_output_flags_t) flags)) { + + return profile; + } + } + } + return NULL; +} + + +audio_devices_t AudioPolicyManager::getDeviceAndMixForInputSource(audio_source_t inputSource, + AudioMix **policyMix) +{ + audio_devices_t availableDeviceTypes = mAvailableInputDevices.types() & + ~AUDIO_DEVICE_BIT_IN; + + for (size_t i = 0; i < mPolicyMixes.size(); i++) { + if (mPolicyMixes[i]->mMix.mMixType != MIX_TYPE_RECORDERS) { + continue; + } + for (size_t j = 0; j < mPolicyMixes[i]->mMix.mCriteria.size(); j++) { + if ((RULE_MATCH_ATTRIBUTE_CAPTURE_PRESET == mPolicyMixes[i]->mMix.mCriteria[j].mRule && + mPolicyMixes[i]->mMix.mCriteria[j].mAttr.mSource == inputSource) || + (RULE_EXCLUDE_ATTRIBUTE_CAPTURE_PRESET == mPolicyMixes[i]->mMix.mCriteria[j].mRule && + mPolicyMixes[i]->mMix.mCriteria[j].mAttr.mSource != inputSource)) { + if (availableDeviceTypes & AUDIO_DEVICE_IN_REMOTE_SUBMIX) { + if (policyMix != NULL) { + *policyMix = &mPolicyMixes[i]->mMix; + } + return AUDIO_DEVICE_IN_REMOTE_SUBMIX; + } + break; + } + } + } + + return getDeviceForInputSource(inputSource); +} + +audio_devices_t AudioPolicyManager::getDeviceForInputSource(audio_source_t inputSource) +{ + uint32_t device = AUDIO_DEVICE_NONE; + audio_devices_t availableDeviceTypes = mAvailableInputDevices.types() & + ~AUDIO_DEVICE_BIT_IN; + + switch (inputSource) { + case AUDIO_SOURCE_VOICE_UPLINK: + if (availableDeviceTypes & AUDIO_DEVICE_IN_VOICE_CALL) { + device = AUDIO_DEVICE_IN_VOICE_CALL; + break; + } + break; + + case AUDIO_SOURCE_DEFAULT: + case AUDIO_SOURCE_MIC: + if (availableDeviceTypes & AUDIO_DEVICE_IN_BLUETOOTH_A2DP) { + device = AUDIO_DEVICE_IN_BLUETOOTH_A2DP; + } else if ((mForceUse[AUDIO_POLICY_FORCE_FOR_RECORD] == AUDIO_POLICY_FORCE_BT_SCO) && + (availableDeviceTypes & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET)) { + device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET; + } else if (availableDeviceTypes & AUDIO_DEVICE_IN_WIRED_HEADSET) { + device = AUDIO_DEVICE_IN_WIRED_HEADSET; + } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_DEVICE) { + device = AUDIO_DEVICE_IN_USB_DEVICE; + } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) { + device = AUDIO_DEVICE_IN_BUILTIN_MIC; + } + break; + + case AUDIO_SOURCE_VOICE_COMMUNICATION: + // Allow only use of devices on primary input if in call and HAL does not support routing + // to voice call path. + if ((mPhoneState == AUDIO_MODE_IN_CALL) && + (mAvailableOutputDevices.types() & AUDIO_DEVICE_OUT_TELEPHONY_TX) == 0) { + availableDeviceTypes = availablePrimaryInputDevices() & ~AUDIO_DEVICE_BIT_IN; + } + + switch (mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION]) { + case AUDIO_POLICY_FORCE_BT_SCO: + // if SCO device is requested but no SCO device is available, fall back to default case + if (availableDeviceTypes & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) { + device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET; + break; + } + // FALL THROUGH + + default: // FORCE_NONE + if (availableDeviceTypes & AUDIO_DEVICE_IN_WIRED_HEADSET) { + device = AUDIO_DEVICE_IN_WIRED_HEADSET; + } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_DEVICE) { + device = AUDIO_DEVICE_IN_USB_DEVICE; + } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) { + device = AUDIO_DEVICE_IN_BUILTIN_MIC; + } + break; + + case AUDIO_POLICY_FORCE_SPEAKER: + if (availableDeviceTypes & AUDIO_DEVICE_IN_BACK_MIC) { + device = AUDIO_DEVICE_IN_BACK_MIC; + } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) { + device = AUDIO_DEVICE_IN_BUILTIN_MIC; + } + break; + } + break; + + case AUDIO_SOURCE_VOICE_RECOGNITION: + case AUDIO_SOURCE_HOTWORD: + if (mForceUse[AUDIO_POLICY_FORCE_FOR_RECORD] == AUDIO_POLICY_FORCE_BT_SCO && + availableDeviceTypes & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) { + device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET; + } else if (availableDeviceTypes & AUDIO_DEVICE_IN_WIRED_HEADSET) { + device = AUDIO_DEVICE_IN_WIRED_HEADSET; + } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_DEVICE) { + device = AUDIO_DEVICE_IN_USB_DEVICE; + } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) { + device = AUDIO_DEVICE_IN_BUILTIN_MIC; + } + break; + case AUDIO_SOURCE_CAMCORDER: + if (availableDeviceTypes & AUDIO_DEVICE_IN_BACK_MIC) { + device = AUDIO_DEVICE_IN_BACK_MIC; + } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) { + device = AUDIO_DEVICE_IN_BUILTIN_MIC; + } + break; + case AUDIO_SOURCE_VOICE_DOWNLINK: + case AUDIO_SOURCE_VOICE_CALL: + if (availableDeviceTypes & AUDIO_DEVICE_IN_VOICE_CALL) { + device = AUDIO_DEVICE_IN_VOICE_CALL; + } + break; + case AUDIO_SOURCE_REMOTE_SUBMIX: + if (availableDeviceTypes & AUDIO_DEVICE_IN_REMOTE_SUBMIX) { + device = AUDIO_DEVICE_IN_REMOTE_SUBMIX; + } + break; + case AUDIO_SOURCE_FM_TUNER: + if (availableDeviceTypes & AUDIO_DEVICE_IN_FM_TUNER) { + device = AUDIO_DEVICE_IN_FM_TUNER; + } + break; + default: + ALOGW("getDeviceForInputSource() invalid input source %d", inputSource); + break; + } + ALOGV("getDeviceForInputSource()input source %d, device %08x", inputSource, device); + return device; +} + +bool AudioPolicyManager::isVirtualInputDevice(audio_devices_t device) +{ + if ((device & AUDIO_DEVICE_BIT_IN) != 0) { + device &= ~AUDIO_DEVICE_BIT_IN; + if ((popcount(device) == 1) && ((device & ~APM_AUDIO_IN_DEVICE_VIRTUAL_ALL) == 0)) + return true; + } + return false; +} + +bool AudioPolicyManager::deviceDistinguishesOnAddress(audio_devices_t device) { + return ((device & APM_AUDIO_DEVICE_MATCH_ADDRESS_ALL & ~AUDIO_DEVICE_BIT_IN) != 0); +} + +audio_io_handle_t AudioPolicyManager::getActiveInput(bool ignoreVirtualInputs) +{ + for (size_t i = 0; i < mInputs.size(); i++) { + const sp input_descriptor = mInputs.valueAt(i); + if ((input_descriptor->mRefCount > 0) + && (!ignoreVirtualInputs || !isVirtualInputDevice(input_descriptor->mDevice))) { + return mInputs.keyAt(i); + } + } + return 0; +} + +uint32_t AudioPolicyManager::activeInputsCount() const +{ + uint32_t count = 0; + for (size_t i = 0; i < mInputs.size(); i++) { + const sp desc = mInputs.valueAt(i); + if (desc->mRefCount > 0) { + count++; + } + } + return count; +} + + +void AudioPolicyManager::initializeVolumeCurves() +{ + for (int i = 0; i < AUDIO_STREAM_CNT; i++) { + for (int j = 0; j < ApmGains::DEVICE_CATEGORY_CNT; j++) { + mStreams[i].mVolumeCurve[j] = + ApmGains::sVolumeProfiles[i][j]; + } + } + + // Check availability of DRC on speaker path: if available, override some of the speaker curves + if (mSpeakerDrcEnabled) { + mStreams[AUDIO_STREAM_SYSTEM].mVolumeCurve[ApmGains::DEVICE_CATEGORY_SPEAKER] = + ApmGains::sDefaultSystemVolumeCurveDrc; + mStreams[AUDIO_STREAM_RING].mVolumeCurve[ApmGains::DEVICE_CATEGORY_SPEAKER] = + ApmGains::sSpeakerSonificationVolumeCurveDrc; + mStreams[AUDIO_STREAM_ALARM].mVolumeCurve[ApmGains::DEVICE_CATEGORY_SPEAKER] = + ApmGains::sSpeakerSonificationVolumeCurveDrc; + mStreams[AUDIO_STREAM_NOTIFICATION].mVolumeCurve[ApmGains::DEVICE_CATEGORY_SPEAKER] = + ApmGains::sSpeakerSonificationVolumeCurveDrc; + mStreams[AUDIO_STREAM_MUSIC].mVolumeCurve[ApmGains::DEVICE_CATEGORY_SPEAKER] = + ApmGains::sSpeakerMediaVolumeCurveDrc; + mStreams[AUDIO_STREAM_ACCESSIBILITY].mVolumeCurve[ApmGains::DEVICE_CATEGORY_SPEAKER] = + ApmGains::sSpeakerMediaVolumeCurveDrc; + } +} + +float AudioPolicyManager::computeVolume(audio_stream_type_t stream, + int index, + audio_io_handle_t output, + audio_devices_t device) +{ + float volume = 1.0; + sp outputDesc = mOutputs.valueFor(output); + StreamDescriptor &streamDesc = mStreams[stream]; + + if (device == AUDIO_DEVICE_NONE) { + device = outputDesc->device(); + } + + volume = ApmGains::volIndexToAmpl(device, streamDesc, index); + + // if a headset is connected, apply the following rules to ring tones and notifications + // to avoid sound level bursts in user's ears: + // - always attenuate ring tones and notifications volume by 6dB + // - if music is playing, always limit the volume to current music volume, + // with a minimum threshold at -36dB so that notification is always perceived. + const routing_strategy stream_strategy = getStrategy(stream); + if ((device & (AUDIO_DEVICE_OUT_BLUETOOTH_A2DP | + AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES | + AUDIO_DEVICE_OUT_WIRED_HEADSET | + AUDIO_DEVICE_OUT_WIRED_HEADPHONE)) && + ((stream_strategy == STRATEGY_SONIFICATION) + || (stream_strategy == STRATEGY_SONIFICATION_RESPECTFUL) + || (stream == AUDIO_STREAM_SYSTEM) + || ((stream_strategy == STRATEGY_ENFORCED_AUDIBLE) && + (mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] == AUDIO_POLICY_FORCE_NONE))) && + streamDesc.mCanBeMuted) { + volume *= SONIFICATION_HEADSET_VOLUME_FACTOR; + // when the phone is ringing we must consider that music could have been paused just before + // by the music application and behave as if music was active if the last music track was + // just stopped + if (isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY) || + mLimitRingtoneVolume) { + audio_devices_t musicDevice = getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/); + float musicVol = computeVolume(AUDIO_STREAM_MUSIC, + mStreams[AUDIO_STREAM_MUSIC].getVolumeIndex(musicDevice), + output, + musicDevice); + float minVol = (musicVol > SONIFICATION_HEADSET_VOLUME_MIN) ? + musicVol : SONIFICATION_HEADSET_VOLUME_MIN; + if (volume > minVol) { + volume = minVol; + ALOGV("computeVolume limiting volume to %f musicVol %f", minVol, musicVol); + } + } + } + + return volume; +} + +status_t AudioPolicyManager::checkAndSetVolume(audio_stream_type_t stream, + int index, + audio_io_handle_t output, + audio_devices_t device, + int delayMs, + bool force) +{ + + // do not change actual stream volume if the stream is muted + if (mOutputs.valueFor(output)->mMuteCount[stream] != 0) { + ALOGVV("checkAndSetVolume() stream %d muted count %d", + stream, mOutputs.valueFor(output)->mMuteCount[stream]); + return NO_ERROR; + } + + // do not change in call volume if bluetooth is connected and vice versa + if ((stream == AUDIO_STREAM_VOICE_CALL && + mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION] == AUDIO_POLICY_FORCE_BT_SCO) || + (stream == AUDIO_STREAM_BLUETOOTH_SCO && + mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION] != AUDIO_POLICY_FORCE_BT_SCO)) { + ALOGV("checkAndSetVolume() cannot set stream %d volume with force use = %d for comm", + stream, mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION]); + return INVALID_OPERATION; + } + + float volume = computeVolume(stream, index, output, device); + // unit gain if rerouting to external policy + if (device == AUDIO_DEVICE_OUT_REMOTE_SUBMIX) { + ssize_t index = mOutputs.indexOfKey(output); + if (index >= 0) { + sp outputDesc = mOutputs.valueAt(index); + if (outputDesc->mPolicyMix != NULL) { + ALOGV("max gain when rerouting for output=%d", output); + volume = 1.0f; + } + } + + } + // We actually change the volume if: + // - the float value returned by computeVolume() changed + // - the force flag is set + if (volume != mOutputs.valueFor(output)->mCurVolume[stream] || + force) { + mOutputs.valueFor(output)->mCurVolume[stream] = volume; + ALOGVV("checkAndSetVolume() for output %d stream %d, volume %f, delay %d", output, stream, volume, delayMs); + // Force VOICE_CALL to track BLUETOOTH_SCO stream volume when bluetooth audio is + // enabled + if (stream == AUDIO_STREAM_BLUETOOTH_SCO) { + mpClientInterface->setStreamVolume(AUDIO_STREAM_VOICE_CALL, volume, output, delayMs); + } + mpClientInterface->setStreamVolume(stream, volume, output, delayMs); + } + + if (stream == AUDIO_STREAM_VOICE_CALL || + stream == AUDIO_STREAM_BLUETOOTH_SCO) { + float voiceVolume; + // Force voice volume to max for bluetooth SCO as volume is managed by the headset + if (stream == AUDIO_STREAM_VOICE_CALL) { + voiceVolume = (float)index/(float)mStreams[stream].mIndexMax; + } else { + voiceVolume = 1.0; + } + + if (voiceVolume != mLastVoiceVolume && output == mPrimaryOutput) { + mpClientInterface->setVoiceVolume(voiceVolume, delayMs); + mLastVoiceVolume = voiceVolume; + } + } + + return NO_ERROR; +} + +void AudioPolicyManager::applyStreamVolumes(audio_io_handle_t output, + audio_devices_t device, + int delayMs, + bool force) +{ + ALOGVV("applyStreamVolumes() for output %d and device %x", output, device); + + for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) { + if (stream == AUDIO_STREAM_PATCH) { + continue; + } + checkAndSetVolume((audio_stream_type_t)stream, + mStreams[stream].getVolumeIndex(device), + output, + device, + delayMs, + force); + } +} + +void AudioPolicyManager::setStrategyMute(routing_strategy strategy, + bool on, + audio_io_handle_t output, + int delayMs, + audio_devices_t device) +{ + ALOGVV("setStrategyMute() strategy %d, mute %d, output %d", strategy, on, output); + for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) { + if (stream == AUDIO_STREAM_PATCH) { + continue; + } + if (getStrategy((audio_stream_type_t)stream) == strategy) { + setStreamMute((audio_stream_type_t)stream, on, output, delayMs, device); + } + } +} + +void AudioPolicyManager::setStreamMute(audio_stream_type_t stream, + bool on, + audio_io_handle_t output, + int delayMs, + audio_devices_t device) +{ + StreamDescriptor &streamDesc = mStreams[stream]; + sp outputDesc = mOutputs.valueFor(output); + if (device == AUDIO_DEVICE_NONE) { + device = outputDesc->device(); + } + + ALOGVV("setStreamMute() stream %d, mute %d, output %d, mMuteCount %d device %04x", + stream, on, output, outputDesc->mMuteCount[stream], device); + + if (on) { + if (outputDesc->mMuteCount[stream] == 0) { + if (streamDesc.mCanBeMuted && + ((stream != AUDIO_STREAM_ENFORCED_AUDIBLE) || + (mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] == AUDIO_POLICY_FORCE_NONE))) { + checkAndSetVolume(stream, 0, output, device, delayMs); + } + } + // increment mMuteCount after calling checkAndSetVolume() so that volume change is not ignored + outputDesc->mMuteCount[stream]++; + } else { + if (outputDesc->mMuteCount[stream] == 0) { + ALOGV("setStreamMute() unmuting non muted stream!"); + return; + } + if (--outputDesc->mMuteCount[stream] == 0) { + checkAndSetVolume(stream, + streamDesc.getVolumeIndex(device), + output, + device, + delayMs); + } + } +} + +void AudioPolicyManager::handleIncallSonification(audio_stream_type_t stream, + bool starting, bool stateChange) +{ + // if the stream pertains to sonification strategy and we are in call we must + // mute the stream if it is low visibility. If it is high visibility, we must play a tone + // in the device used for phone strategy and play the tone if the selected device does not + // interfere with the device used for phone strategy + // if stateChange is true, we are called from setPhoneState() and we must mute or unmute as + // many times as there are active tracks on the output + const routing_strategy stream_strategy = getStrategy(stream); + if ((stream_strategy == STRATEGY_SONIFICATION) || + ((stream_strategy == STRATEGY_SONIFICATION_RESPECTFUL))) { + sp outputDesc = mOutputs.valueFor(mPrimaryOutput); + ALOGV("handleIncallSonification() stream %d starting %d device %x stateChange %d", + stream, starting, outputDesc->mDevice, stateChange); + if (outputDesc->mRefCount[stream]) { + int muteCount = 1; + if (stateChange) { + muteCount = outputDesc->mRefCount[stream]; + } + if (audio_is_low_visibility(stream)) { + ALOGV("handleIncallSonification() low visibility, muteCount %d", muteCount); + for (int i = 0; i < muteCount; i++) { + setStreamMute(stream, starting, mPrimaryOutput); + } + } else { + ALOGV("handleIncallSonification() high visibility"); + if (outputDesc->device() & + getDeviceForStrategy(STRATEGY_PHONE, true /*fromCache*/)) { + ALOGV("handleIncallSonification() high visibility muted, muteCount %d", muteCount); + for (int i = 0; i < muteCount; i++) { + setStreamMute(stream, starting, mPrimaryOutput); + } + } + if (starting) { + mpClientInterface->startTone(AUDIO_POLICY_TONE_IN_CALL_NOTIFICATION, + AUDIO_STREAM_VOICE_CALL); + } else { + mpClientInterface->stopTone(); + } + } + } + } +} + +bool AudioPolicyManager::isInCall() +{ + return isStateInCall(mPhoneState); +} + +bool AudioPolicyManager::isStateInCall(int state) { + return ((state == AUDIO_MODE_IN_CALL) || + (state == AUDIO_MODE_IN_COMMUNICATION)); +} + +uint32_t AudioPolicyManager::getMaxEffectsCpuLoad() +{ + return MAX_EFFECTS_CPU_LOAD; +} + +uint32_t AudioPolicyManager::getMaxEffectsMemory() +{ + return MAX_EFFECTS_MEMORY; +} + + +// --- EffectDescriptor class implementation + +status_t AudioPolicyManager::EffectDescriptor::dump(int fd) +{ + const size_t SIZE = 256; + char buffer[SIZE]; + String8 result; + + snprintf(buffer, SIZE, " I/O: %d\n", mIo); + result.append(buffer); + snprintf(buffer, SIZE, " Strategy: %d\n", mStrategy); + result.append(buffer); + snprintf(buffer, SIZE, " Session: %d\n", mSession); + result.append(buffer); + snprintf(buffer, SIZE, " Name: %s\n", mDesc.name); + result.append(buffer); + snprintf(buffer, SIZE, " %s\n", mEnabled ? "Enabled" : "Disabled"); + result.append(buffer); + write(fd, result.string(), result.size()); + + return NO_ERROR; +} + + +// --- audio_policy.conf file parsing +// TODO candidate to be moved to ConfigParsingUtils +void AudioPolicyManager::loadHwModule(cnode *root) +{ + status_t status = NAME_NOT_FOUND; + cnode *node; + sp module = new HwModule(root->name); + + node = config_find(root, DEVICES_TAG); + if (node != NULL) { + node = node->first_child; + while (node) { + ALOGV("loadHwModule() loading device %s", node->name); + status_t tmpStatus = module->loadDevice(node); + if (status == NAME_NOT_FOUND || status == NO_ERROR) { + status = tmpStatus; + } + node = node->next; + } + } + node = config_find(root, OUTPUTS_TAG); + if (node != NULL) { + node = node->first_child; + while (node) { + ALOGV("loadHwModule() loading output %s", node->name); + status_t tmpStatus = module->loadOutput(node); + if (status == NAME_NOT_FOUND || status == NO_ERROR) { + status = tmpStatus; + } + node = node->next; + } + } + node = config_find(root, INPUTS_TAG); + if (node != NULL) { + node = node->first_child; + while (node) { + ALOGV("loadHwModule() loading input %s", node->name); + status_t tmpStatus = module->loadInput(node); + if (status == NAME_NOT_FOUND || status == NO_ERROR) { + status = tmpStatus; + } + node = node->next; + } + } + loadGlobalConfig(root, module); + + if (status == NO_ERROR) { + mHwModules.add(module); + } +} + +// TODO candidate to be moved to ConfigParsingUtils +void AudioPolicyManager::loadHwModules(cnode *root) +{ + cnode *node = config_find(root, AUDIO_HW_MODULE_TAG); + if (node == NULL) { + return; + } + + node = node->first_child; + while (node) { + ALOGV("loadHwModules() loading module %s", node->name); + loadHwModule(node); + node = node->next; + } +} + +// TODO candidate to be moved to ConfigParsingUtils +void AudioPolicyManager::loadGlobalConfig(cnode *root, const sp& module) +{ + cnode *node = config_find(root, GLOBAL_CONFIG_TAG); + + if (node == NULL) { + return; + } + DeviceVector declaredDevices; + if (module != NULL) { + declaredDevices = module->mDeclaredDevices; + } + + node = node->first_child; + while (node) { + if (strcmp(ATTACHED_OUTPUT_DEVICES_TAG, node->name) == 0) { + mAvailableOutputDevices.loadDevicesFromName((char *)node->value, + declaredDevices); + ALOGV("loadGlobalConfig() Attached Output Devices %08x", + mAvailableOutputDevices.types()); + } else if (strcmp(DEFAULT_OUTPUT_DEVICE_TAG, node->name) == 0) { + audio_devices_t device = (audio_devices_t)ConfigParsingUtils::stringToEnum( + sDeviceNameToEnumTable, + ARRAY_SIZE(sDeviceNameToEnumTable), + (char *)node->value); + if (device != AUDIO_DEVICE_NONE) { + mDefaultOutputDevice = new DeviceDescriptor(String8("default-output"), device); + } else { + ALOGW("loadGlobalConfig() default device not specified"); + } + ALOGV("loadGlobalConfig() mDefaultOutputDevice %08x", mDefaultOutputDevice->mDeviceType); + } else if (strcmp(ATTACHED_INPUT_DEVICES_TAG, node->name) == 0) { + mAvailableInputDevices.loadDevicesFromName((char *)node->value, + declaredDevices); + ALOGV("loadGlobalConfig() Available InputDevices %08x", mAvailableInputDevices.types()); + } else if (strcmp(SPEAKER_DRC_ENABLED_TAG, node->name) == 0) { + mSpeakerDrcEnabled = ConfigParsingUtils::stringToBool((char *)node->value); + ALOGV("loadGlobalConfig() mSpeakerDrcEnabled = %d", mSpeakerDrcEnabled); + } else if (strcmp(AUDIO_HAL_VERSION_TAG, node->name) == 0) { + uint32_t major, minor; + sscanf((char *)node->value, "%u.%u", &major, &minor); + module->mHalVersion = HARDWARE_DEVICE_API_VERSION(major, minor); + ALOGV("loadGlobalConfig() mHalVersion = %04x major %u minor %u", + module->mHalVersion, major, minor); + } + node = node->next; + } +} + +// TODO candidate to be moved to ConfigParsingUtils +status_t AudioPolicyManager::loadAudioPolicyConfig(const char *path) +{ + cnode *root; + char *data; + + data = (char *)load_file(path, NULL); + if (data == NULL) { + return -ENODEV; + } + root = config_node("", ""); + config_load(root, data); + + loadHwModules(root); + // legacy audio_policy.conf files have one global_configuration section + loadGlobalConfig(root, getModuleFromName(AUDIO_HARDWARE_MODULE_ID_PRIMARY)); + config_free(root); + free(root); + free(data); + + ALOGI("loadAudioPolicyConfig() loaded %s\n", path); + + return NO_ERROR; +} + +void AudioPolicyManager::defaultAudioPolicyConfig(void) +{ + sp module; + sp profile; + sp defaultInputDevice = + new DeviceDescriptor(String8("builtin-mic"), AUDIO_DEVICE_IN_BUILTIN_MIC); + mAvailableOutputDevices.add(mDefaultOutputDevice); + mAvailableInputDevices.add(defaultInputDevice); + + module = new HwModule("primary"); + + profile = new IOProfile(String8("primary"), AUDIO_PORT_ROLE_SOURCE, module); + profile->mSamplingRates.add(44100); + profile->mFormats.add(AUDIO_FORMAT_PCM_16_BIT); + profile->mChannelMasks.add(AUDIO_CHANNEL_OUT_STEREO); + profile->mSupportedDevices.add(mDefaultOutputDevice); + profile->mFlags = AUDIO_OUTPUT_FLAG_PRIMARY; + module->mOutputProfiles.add(profile); + + profile = new IOProfile(String8("primary"), AUDIO_PORT_ROLE_SINK, module); + profile->mSamplingRates.add(8000); + profile->mFormats.add(AUDIO_FORMAT_PCM_16_BIT); + profile->mChannelMasks.add(AUDIO_CHANNEL_IN_MONO); + profile->mSupportedDevices.add(defaultInputDevice); + module->mInputProfiles.add(profile); + + mHwModules.add(module); +} + +audio_stream_type_t AudioPolicyManager::streamTypefromAttributesInt(const audio_attributes_t *attr) +{ + // flags to stream type mapping + if ((attr->flags & AUDIO_FLAG_AUDIBILITY_ENFORCED) == AUDIO_FLAG_AUDIBILITY_ENFORCED) { + return AUDIO_STREAM_ENFORCED_AUDIBLE; + } + if ((attr->flags & AUDIO_FLAG_SCO) == AUDIO_FLAG_SCO) { + return AUDIO_STREAM_BLUETOOTH_SCO; + } + if ((attr->flags & AUDIO_FLAG_BEACON) == AUDIO_FLAG_BEACON) { + return AUDIO_STREAM_TTS; + } + + // usage to stream type mapping + switch (attr->usage) { + case AUDIO_USAGE_MEDIA: + case AUDIO_USAGE_GAME: + case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE: + return AUDIO_STREAM_MUSIC; + case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY: + if (isStreamActive(AUDIO_STREAM_ALARM)) { + return AUDIO_STREAM_ALARM; + } + if (isStreamActive(AUDIO_STREAM_RING)) { + return AUDIO_STREAM_RING; + } + if (isInCall()) { + return AUDIO_STREAM_VOICE_CALL; + } + return AUDIO_STREAM_ACCESSIBILITY; + case AUDIO_USAGE_ASSISTANCE_SONIFICATION: + return AUDIO_STREAM_SYSTEM; + case AUDIO_USAGE_VOICE_COMMUNICATION: + return AUDIO_STREAM_VOICE_CALL; + + case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING: + return AUDIO_STREAM_DTMF; + + case AUDIO_USAGE_ALARM: + return AUDIO_STREAM_ALARM; + case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE: + return AUDIO_STREAM_RING; + + case AUDIO_USAGE_NOTIFICATION: + case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST: + case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT: + case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED: + case AUDIO_USAGE_NOTIFICATION_EVENT: + return AUDIO_STREAM_NOTIFICATION; + + case AUDIO_USAGE_UNKNOWN: + default: + return AUDIO_STREAM_MUSIC; + } +} + +bool AudioPolicyManager::isValidAttributes(const audio_attributes_t *paa) { + // has flags that map to a strategy? + if ((paa->flags & (AUDIO_FLAG_AUDIBILITY_ENFORCED | AUDIO_FLAG_SCO | AUDIO_FLAG_BEACON)) != 0) { + return true; + } + + // has known usage? + switch (paa->usage) { + case AUDIO_USAGE_UNKNOWN: + case AUDIO_USAGE_MEDIA: + case AUDIO_USAGE_VOICE_COMMUNICATION: + case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING: + case AUDIO_USAGE_ALARM: + case AUDIO_USAGE_NOTIFICATION: + case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE: + case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST: + case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT: + case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED: + case AUDIO_USAGE_NOTIFICATION_EVENT: + case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY: + case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE: + case AUDIO_USAGE_ASSISTANCE_SONIFICATION: + case AUDIO_USAGE_GAME: + case AUDIO_USAGE_VIRTUAL_SOURCE: + break; + default: + return false; + } + return true; +} + +}; // namespace android diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.h b/services/audiopolicy/managerdefault/AudioPolicyManager.h new file mode 100644 index 0000000..61ea6f2 --- /dev/null +++ b/services/audiopolicy/managerdefault/AudioPolicyManager.h @@ -0,0 +1,560 @@ +/* + * Copyright (C) 2009 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "AudioPolicyInterface.h" + +#include "Gains.h" +#include "Ports.h" +#include "ConfigParsingUtils.h" +#include "Devices.h" +#include "IOProfile.h" +#include "HwModule.h" +#include "AudioInputDescriptor.h" +#include "AudioOutputDescriptor.h" + +namespace android { + +// ---------------------------------------------------------------------------- + +// Attenuation applied to STRATEGY_SONIFICATION streams when a headset is connected: 6dB +#define SONIFICATION_HEADSET_VOLUME_FACTOR 0.5 +// Min volume for STRATEGY_SONIFICATION streams when limited by music volume: -36dB +#define SONIFICATION_HEADSET_VOLUME_MIN 0.016 +// Time in milliseconds during which we consider that music is still active after a music +// track was stopped - see computeVolume() +#define SONIFICATION_HEADSET_MUSIC_DELAY 5000 +// Time in milliseconds after media stopped playing during which we consider that the +// sonification should be as unobtrusive as during the time media was playing. +#define SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY 5000 +// Time in milliseconds during witch some streams are muted while the audio path +// is switched +#define MUTE_TIME_MS 2000 + +#define NUM_TEST_OUTPUTS 5 + +#define NUM_VOL_CURVE_KNEES 2 + +// Default minimum length allowed for offloading a compressed track +// Can be overridden by the audio.offload.min.duration.secs property +#define OFFLOAD_DEFAULT_MIN_DURATION_SECS 60 + +#define MAX_MIXER_SAMPLING_RATE 48000 +#define MAX_MIXER_CHANNEL_COUNT 8 + +// ---------------------------------------------------------------------------- +// AudioPolicyManager implements audio policy manager behavior common to all platforms. +// ---------------------------------------------------------------------------- + +class AudioPolicyManager: public AudioPolicyInterface +#ifdef AUDIO_POLICY_TEST + , public Thread +#endif //AUDIO_POLICY_TEST +{ + +public: + AudioPolicyManager(AudioPolicyClientInterface *clientInterface); + virtual ~AudioPolicyManager(); + + // AudioPolicyInterface + virtual status_t setDeviceConnectionState(audio_devices_t device, + audio_policy_dev_state_t state, + const char *device_address, + const char *device_name); + virtual audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device, + const char *device_address); + virtual void setPhoneState(audio_mode_t state); + virtual void setForceUse(audio_policy_force_use_t usage, + audio_policy_forced_cfg_t config); + virtual audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage); + virtual void setSystemProperty(const char* property, const char* value); + virtual status_t initCheck(); + virtual audio_io_handle_t getOutput(audio_stream_type_t stream, + uint32_t samplingRate, + audio_format_t format, + audio_channel_mask_t channelMask, + audio_output_flags_t flags, + const audio_offload_info_t *offloadInfo); + virtual status_t getOutputForAttr(const audio_attributes_t *attr, + audio_io_handle_t *output, + audio_session_t session, + audio_stream_type_t *stream, + uint32_t samplingRate, + audio_format_t format, + audio_channel_mask_t channelMask, + audio_output_flags_t flags, + const audio_offload_info_t *offloadInfo); + virtual status_t startOutput(audio_io_handle_t output, + audio_stream_type_t stream, + audio_session_t session); + virtual status_t stopOutput(audio_io_handle_t output, + audio_stream_type_t stream, + audio_session_t session); + virtual void releaseOutput(audio_io_handle_t output, + audio_stream_type_t stream, + audio_session_t session); + virtual status_t getInputForAttr(const audio_attributes_t *attr, + audio_io_handle_t *input, + audio_session_t session, + uint32_t samplingRate, + audio_format_t format, + audio_channel_mask_t channelMask, + audio_input_flags_t flags, + input_type_t *inputType); + + // indicates to the audio policy manager that the input starts being used. + virtual status_t startInput(audio_io_handle_t input, + audio_session_t session); + + // indicates to the audio policy manager that the input stops being used. + virtual status_t stopInput(audio_io_handle_t input, + audio_session_t session); + virtual void releaseInput(audio_io_handle_t input, + audio_session_t session); + virtual void closeAllInputs(); + virtual void initStreamVolume(audio_stream_type_t stream, + int indexMin, + int indexMax); + virtual status_t setStreamVolumeIndex(audio_stream_type_t stream, + int index, + audio_devices_t device); + virtual status_t getStreamVolumeIndex(audio_stream_type_t stream, + int *index, + audio_devices_t device); + + // return the strategy corresponding to a given stream type + virtual uint32_t getStrategyForStream(audio_stream_type_t stream); + // return the strategy corresponding to the given audio attributes + virtual uint32_t getStrategyForAttr(const audio_attributes_t *attr); + + // return the enabled output devices for the given stream type + virtual audio_devices_t getDevicesForStream(audio_stream_type_t stream); + + virtual audio_io_handle_t getOutputForEffect(const effect_descriptor_t *desc = NULL); + virtual status_t registerEffect(const effect_descriptor_t *desc, + audio_io_handle_t io, + uint32_t strategy, + int session, + int id); + virtual status_t unregisterEffect(int id); + virtual status_t setEffectEnabled(int id, bool enabled); + + virtual bool isStreamActive(audio_stream_type_t stream, uint32_t inPastMs = 0) const; + // return whether a stream is playing remotely, override to change the definition of + // local/remote playback, used for instance by notification manager to not make + // media players lose audio focus when not playing locally + // For the base implementation, "remotely" means playing during screen mirroring which + // uses an output for playback with a non-empty, non "0" address. + virtual bool isStreamActiveRemotely(audio_stream_type_t stream, uint32_t inPastMs = 0) const; + virtual bool isSourceActive(audio_source_t source) const; + + virtual status_t dump(int fd); + + virtual bool isOffloadSupported(const audio_offload_info_t& offloadInfo); + + virtual status_t listAudioPorts(audio_port_role_t role, + audio_port_type_t type, + unsigned int *num_ports, + struct audio_port *ports, + unsigned int *generation); + virtual status_t getAudioPort(struct audio_port *port); + virtual status_t createAudioPatch(const struct audio_patch *patch, + audio_patch_handle_t *handle, + uid_t uid); + virtual status_t releaseAudioPatch(audio_patch_handle_t handle, + uid_t uid); + virtual status_t listAudioPatches(unsigned int *num_patches, + struct audio_patch *patches, + unsigned int *generation); + virtual status_t setAudioPortConfig(const struct audio_port_config *config); + virtual void clearAudioPatches(uid_t uid); + + virtual status_t acquireSoundTriggerSession(audio_session_t *session, + audio_io_handle_t *ioHandle, + audio_devices_t *device); + + virtual status_t releaseSoundTriggerSession(audio_session_t session); + + virtual status_t registerPolicyMixes(Vector mixes); + virtual status_t unregisterPolicyMixes(Vector mixes); + + // Audio policy configuration file parsing (audio_policy.conf) + // TODO candidates to be moved to ConfigParsingUtils + void loadHwModule(cnode *root); + void loadHwModules(cnode *root); + void loadGlobalConfig(cnode *root, const sp& module); + status_t loadAudioPolicyConfig(const char *path); + void defaultAudioPolicyConfig(void); + + // return the strategy corresponding to a given stream type + static routing_strategy getStrategy(audio_stream_type_t stream); + + static uint32_t nextUniqueId(); +protected: + + class EffectDescriptor : public RefBase + { + public: + + status_t dump(int fd); + + int mIo; // io the effect is attached to + routing_strategy mStrategy; // routing strategy the effect is associated to + int mSession; // audio session the effect is on + effect_descriptor_t mDesc; // effect descriptor + bool mEnabled; // enabled state: CPU load being used or not + }; + + void addOutput(audio_io_handle_t output, sp outputDesc); + void addInput(audio_io_handle_t input, sp inputDesc); + + // return appropriate device for streams handled by the specified strategy according to current + // phone state, connected devices... + // if fromCache is true, the device is returned from mDeviceForStrategy[], + // otherwise it is determine by current state + // (device connected,phone state, force use, a2dp output...) + // This allows to: + // 1 speed up process when the state is stable (when starting or stopping an output) + // 2 access to either current device selection (fromCache == true) or + // "future" device selection (fromCache == false) when called from a context + // where conditions are changing (setDeviceConnectionState(), setPhoneState()...) AND + // before updateDevicesAndOutputs() is called. + virtual audio_devices_t getDeviceForStrategy(routing_strategy strategy, + bool fromCache); + + // change the route of the specified output. Returns the number of ms we have slept to + // allow new routing to take effect in certain cases. + virtual uint32_t setOutputDevice(audio_io_handle_t output, + audio_devices_t device, + bool force = false, + int delayMs = 0, + audio_patch_handle_t *patchHandle = NULL, + const char* address = NULL); + status_t resetOutputDevice(audio_io_handle_t output, + int delayMs = 0, + audio_patch_handle_t *patchHandle = NULL); + status_t setInputDevice(audio_io_handle_t input, + audio_devices_t device, + bool force = false, + audio_patch_handle_t *patchHandle = NULL); + status_t resetInputDevice(audio_io_handle_t input, + audio_patch_handle_t *patchHandle = NULL); + + // select input device corresponding to requested audio source + virtual audio_devices_t getDeviceForInputSource(audio_source_t inputSource); + + // return io handle of active input or 0 if no input is active + // Only considers inputs from physical devices (e.g. main mic, headset mic) when + // ignoreVirtualInputs is true. + audio_io_handle_t getActiveInput(bool ignoreVirtualInputs = true); + + uint32_t activeInputsCount() const; + + // initialize volume curves for each strategy and device category + void initializeVolumeCurves(); + + // compute the actual volume for a given stream according to the requested index and a particular + // device + virtual float computeVolume(audio_stream_type_t stream, int index, + audio_io_handle_t output, audio_devices_t device); + + // check that volume change is permitted, compute and send new volume to audio hardware + virtual status_t checkAndSetVolume(audio_stream_type_t stream, int index, + audio_io_handle_t output, + audio_devices_t device, + int delayMs = 0, bool force = false); + + // apply all stream volumes to the specified output and device + void applyStreamVolumes(audio_io_handle_t output, audio_devices_t device, int delayMs = 0, bool force = false); + + // Mute or unmute all streams handled by the specified strategy on the specified output + void setStrategyMute(routing_strategy strategy, + bool on, + audio_io_handle_t output, + int delayMs = 0, + audio_devices_t device = (audio_devices_t)0); + + // Mute or unmute the stream on the specified output + void setStreamMute(audio_stream_type_t stream, + bool on, + audio_io_handle_t output, + int delayMs = 0, + audio_devices_t device = (audio_devices_t)0); + + // handle special cases for sonification strategy while in call: mute streams or replace by + // a special tone in the device used for communication + void handleIncallSonification(audio_stream_type_t stream, bool starting, bool stateChange); + + // true if device is in a telephony or VoIP call + virtual bool isInCall(); + + // true if given state represents a device in a telephony or VoIP call + virtual bool isStateInCall(int state); + + // when a device is connected, checks if an open output can be routed + // to this device. If none is open, tries to open one of the available outputs. + // Returns an output suitable to this device or 0. + // when a device is disconnected, checks if an output is not used any more and + // returns its handle if any. + // transfers the audio tracks and effects from one output thread to another accordingly. + status_t checkOutputsForDevice(const sp devDesc, + audio_policy_dev_state_t state, + SortedVector& outputs, + const String8 address); + + status_t checkInputsForDevice(audio_devices_t device, + audio_policy_dev_state_t state, + SortedVector& inputs, + const String8 address); + + // close an output and its companion duplicating output. + void closeOutput(audio_io_handle_t output); + + // close an input. + void closeInput(audio_io_handle_t input); + + // checks and if necessary changes outputs used for all strategies. + // must be called every time a condition that affects the output choice for a given strategy + // changes: connected device, phone state, force use... + // Must be called before updateDevicesAndOutputs() + void checkOutputForStrategy(routing_strategy strategy); + + // Same as checkOutputForStrategy() but for a all strategies in order of priority + void checkOutputForAllStrategies(); + + // manages A2DP output suspend/restore according to phone state and BT SCO usage + void checkA2dpSuspend(); + + // returns the A2DP output handle if it is open or 0 otherwise + audio_io_handle_t getA2dpOutput(); + + // selects the most appropriate device on output for current state + // must be called every time a condition that affects the device choice for a given output is + // changed: connected device, phone state, force use, output start, output stop.. + // see getDeviceForStrategy() for the use of fromCache parameter + audio_devices_t getNewOutputDevice(audio_io_handle_t output, bool fromCache); + + // updates cache of device used by all strategies (mDeviceForStrategy[]) + // must be called every time a condition that affects the device choice for a given strategy is + // changed: connected device, phone state, force use... + // cached values are used by getDeviceForStrategy() if parameter fromCache is true. + // Must be called after checkOutputForAllStrategies() + void updateDevicesAndOutputs(); + + // selects the most appropriate device on input for current state + audio_devices_t getNewInputDevice(audio_io_handle_t input); + + virtual uint32_t getMaxEffectsCpuLoad(); + virtual uint32_t getMaxEffectsMemory(); +#ifdef AUDIO_POLICY_TEST + virtual bool threadLoop(); + void exit(); + int testOutputIndex(audio_io_handle_t output); +#endif //AUDIO_POLICY_TEST + + status_t setEffectEnabled(const sp& effectDesc, bool enabled); + + SortedVector getOutputsForDevice(audio_devices_t device, + DefaultKeyedVector > openOutputs); + bool vectorsEqual(SortedVector& outputs1, + SortedVector& outputs2); + + // mute/unmute strategies using an incompatible device combination + // if muting, wait for the audio in pcm buffer to be drained before proceeding + // if unmuting, unmute only after the specified delay + // Returns the number of ms waited + virtual uint32_t checkDeviceMuteStrategies(sp outputDesc, + audio_devices_t prevDevice, + uint32_t delayMs); + + audio_io_handle_t selectOutput(const SortedVector& outputs, + audio_output_flags_t flags, + audio_format_t format); + // samplingRate parameter is an in/out and so may be modified + sp getInputProfile(audio_devices_t device, + String8 address, + uint32_t& samplingRate, + audio_format_t format, + audio_channel_mask_t channelMask, + audio_input_flags_t flags); + sp getProfileForDirectOutput(audio_devices_t device, + uint32_t samplingRate, + audio_format_t format, + audio_channel_mask_t channelMask, + audio_output_flags_t flags); + + audio_io_handle_t selectOutputForEffects(const SortedVector& outputs); + + bool isNonOffloadableEffectEnabled(); + + virtual status_t addAudioPatch(audio_patch_handle_t handle, + const sp& patch); + virtual status_t removeAudioPatch(audio_patch_handle_t handle); + + sp getOutputFromId(audio_port_handle_t id) const; + sp getInputFromId(audio_port_handle_t id) const; + sp getModuleForDevice(audio_devices_t device) const; + sp getModuleFromName(const char *name) const; + audio_devices_t availablePrimaryOutputDevices(); + audio_devices_t availablePrimaryInputDevices(); + + void updateCallRouting(audio_devices_t rxDevice, int delayMs = 0); + + + uid_t mUidCached; + AudioPolicyClientInterface *mpClientInterface; // audio policy client interface + audio_io_handle_t mPrimaryOutput; // primary output handle + // list of descriptors for outputs currently opened + DefaultKeyedVector > mOutputs; + // copy of mOutputs before setDeviceConnectionState() opens new outputs + // reset to mOutputs when updateDevicesAndOutputs() is called. + DefaultKeyedVector > mPreviousOutputs; + DefaultKeyedVector > mInputs; // list of input descriptors + DeviceVector mAvailableOutputDevices; // all available output devices + DeviceVector mAvailableInputDevices; // all available input devices + int mPhoneState; // current phone state + audio_policy_forced_cfg_t mForceUse[AUDIO_POLICY_FORCE_USE_CNT]; // current forced use configuration + + StreamDescriptor mStreams[AUDIO_STREAM_CNT]; // stream descriptors for volume control + bool mLimitRingtoneVolume; // limit ringtone volume to music volume if headset connected + audio_devices_t mDeviceForStrategy[NUM_STRATEGIES]; + float mLastVoiceVolume; // last voice volume value sent to audio HAL + + // Maximum CPU load allocated to audio effects in 0.1 MIPS (ARMv5TE, 0 WS memory) units + static const uint32_t MAX_EFFECTS_CPU_LOAD = 1000; + // Maximum memory allocated to audio effects in KB + static const uint32_t MAX_EFFECTS_MEMORY = 512; + uint32_t mTotalEffectsCpuLoad; // current CPU load used by effects + uint32_t mTotalEffectsMemory; // current memory used by effects + KeyedVector > mEffects; // list of registered audio effects + bool mA2dpSuspended; // true if A2DP output is suspended + sp mDefaultOutputDevice; // output device selected by default at boot time + bool mSpeakerDrcEnabled;// true on devices that use DRC on the DEVICE_CATEGORY_SPEAKER path + // to boost soft sounds, used to adjust volume curves accordingly + + Vector < sp > mHwModules; + static volatile int32_t mNextUniqueId; + volatile int32_t mAudioPortGeneration; + + DefaultKeyedVector > mAudioPatches; + + DefaultKeyedVector mSoundTriggerSessions; + + sp mCallTxPatch; + sp mCallRxPatch; + + // for supporting "beacon" streams, i.e. streams that only play on speaker, and never + // when something other than STREAM_TTS (a.k.a. "Transmitted Through Speaker") is playing + enum { + STARTING_OUTPUT, + STARTING_BEACON, + STOPPING_OUTPUT, + STOPPING_BEACON + }; + uint32_t mBeaconMuteRefCount; // ref count for stream that would mute beacon + uint32_t mBeaconPlayingRefCount;// ref count for the playing beacon streams + bool mBeaconMuted; // has STREAM_TTS been muted + + // custom mix entry in mPolicyMixes + class AudioPolicyMix : public RefBase { + public: + AudioPolicyMix() {} + + AudioMix mMix; // Audio policy mix descriptor + sp mOutput; // Corresponding output stream + }; + DefaultKeyedVector > mPolicyMixes; // list of registered mixes + + +#ifdef AUDIO_POLICY_TEST + Mutex mLock; + Condition mWaitWorkCV; + + int mCurOutput; + bool mDirectOutput; + audio_io_handle_t mTestOutputs[NUM_TEST_OUTPUTS]; + int mTestInput; + uint32_t mTestDevice; + uint32_t mTestSamplingRate; + uint32_t mTestFormat; + uint32_t mTestChannels; + uint32_t mTestLatencyMs; +#endif //AUDIO_POLICY_TEST + + static bool isVirtualInputDevice(audio_devices_t device); + + uint32_t nextAudioPortGeneration(); +private: + // updates device caching and output for streams that can influence the + // routing of notifications + void handleNotificationRoutingForStream(audio_stream_type_t stream); + static bool deviceDistinguishesOnAddress(audio_devices_t device); + // find the outputs on a given output descriptor that have the given address. + // to be called on an AudioOutputDescriptor whose supported devices (as defined + // in mProfile->mSupportedDevices) matches the device whose address is to be matched. + // see deviceDistinguishesOnAddress(audio_devices_t) for whether the device type is one + // where addresses are used to distinguish between one connected device and another. + void findIoHandlesByAddress(sp desc /*in*/, + const audio_devices_t device /*in*/, + const String8 address /*in*/, + SortedVector& outputs /*out*/); + uint32_t curAudioPortGeneration() const { return mAudioPortGeneration; } + // internal method to return the output handle for the given device and format + audio_io_handle_t getOutputForDevice( + audio_devices_t device, + audio_session_t session, + audio_stream_type_t stream, + uint32_t samplingRate, + audio_format_t format, + audio_channel_mask_t channelMask, + audio_output_flags_t flags, + const audio_offload_info_t *offloadInfo); + // internal function to derive a stream type value from audio attributes + audio_stream_type_t streamTypefromAttributesInt(const audio_attributes_t *attr); + // return true if any output is playing anything besides the stream to ignore + bool isAnyOutputActive(audio_stream_type_t streamToIgnore); + // event is one of STARTING_OUTPUT, STARTING_BEACON, STOPPING_OUTPUT, STOPPING_BEACON + // returns 0 if no mute/unmute event happened, the largest latency of the device where + // the mute/unmute happened + uint32_t handleEventForBeacon(int event); + uint32_t setBeaconMute(bool mute); + bool isValidAttributes(const audio_attributes_t *paa); + + // select input device corresponding to requested audio source and return associated policy + // mix if any. Calls getDeviceForInputSource(). + audio_devices_t getDeviceAndMixForInputSource(audio_source_t inputSource, + AudioMix **policyMix = NULL); + + // Called by setDeviceConnectionState(). + status_t setDeviceConnectionStateInt(audio_devices_t device, + audio_policy_dev_state_t state, + const char *device_address, + const char *device_name); + sp getDeviceDescriptor(const audio_devices_t device, + const char *device_address, + const char *device_name); +}; + +}; diff --git a/services/audiopolicy/managerdefault/ConfigParsingUtils.cpp b/services/audiopolicy/managerdefault/ConfigParsingUtils.cpp new file mode 100644 index 0000000..1afd487 --- /dev/null +++ b/services/audiopolicy/managerdefault/ConfigParsingUtils.cpp @@ -0,0 +1,121 @@ +/* + * Copyright (C) 2015 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#define LOG_TAG "APM::ConfigParsingUtils" +//#define LOG_NDEBUG 0 + +#include "AudioPolicyManager.h" + +namespace android { + +//static +uint32_t ConfigParsingUtils::stringToEnum(const struct StringToEnum *table, + size_t size, + const char *name) +{ + for (size_t i = 0; i < size; i++) { + if (strcmp(table[i].name, name) == 0) { + ALOGV("stringToEnum() found %s", table[i].name); + return table[i].value; + } + } + return 0; +} + +//static +const char *ConfigParsingUtils::enumToString(const struct StringToEnum *table, + size_t size, + uint32_t value) +{ + for (size_t i = 0; i < size; i++) { + if (table[i].value == value) { + return table[i].name; + } + } + return ""; +} + +//static +bool ConfigParsingUtils::stringToBool(const char *value) +{ + return ((strcasecmp("true", value) == 0) || (strcmp("1", value) == 0)); +} + + +// --- audio_policy.conf file parsing +//static +uint32_t ConfigParsingUtils::parseOutputFlagNames(char *name) +{ + uint32_t flag = 0; + + // it is OK to cast name to non const here as we are not going to use it after + // strtok() modifies it + char *flagName = strtok(name, "|"); + while (flagName != NULL) { + if (strlen(flagName) != 0) { + flag |= ConfigParsingUtils::stringToEnum(sOutputFlagNameToEnumTable, + ARRAY_SIZE(sOutputFlagNameToEnumTable), + flagName); + } + flagName = strtok(NULL, "|"); + } + //force direct flag if offload flag is set: offloading implies a direct output stream + // and all common behaviors are driven by checking only the direct flag + // this should normally be set appropriately in the policy configuration file + if ((flag & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) { + flag |= AUDIO_OUTPUT_FLAG_DIRECT; + } + + return flag; +} + +//static +uint32_t ConfigParsingUtils::parseInputFlagNames(char *name) +{ + uint32_t flag = 0; + + // it is OK to cast name to non const here as we are not going to use it after + // strtok() modifies it + char *flagName = strtok(name, "|"); + while (flagName != NULL) { + if (strlen(flagName) != 0) { + flag |= stringToEnum(sInputFlagNameToEnumTable, + ARRAY_SIZE(sInputFlagNameToEnumTable), + flagName); + } + flagName = strtok(NULL, "|"); + } + return flag; +} + +//static +audio_devices_t ConfigParsingUtils::parseDeviceNames(char *name) +{ + uint32_t device = 0; + + char *devName = strtok(name, "|"); + while (devName != NULL) { + if (strlen(devName) != 0) { + device |= stringToEnum(sDeviceNameToEnumTable, + ARRAY_SIZE(sDeviceNameToEnumTable), + devName); + } + devName = strtok(NULL, "|"); + } + return device; +} + +}; // namespace android diff --git a/services/audiopolicy/managerdefault/ConfigParsingUtils.h b/services/audiopolicy/managerdefault/ConfigParsingUtils.h new file mode 100644 index 0000000..7969661 --- /dev/null +++ b/services/audiopolicy/managerdefault/ConfigParsingUtils.h @@ -0,0 +1,159 @@ +/* + * Copyright (C) 2015 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +namespace android { + +// ---------------------------------------------------------------------------- +// Definitions for audio_policy.conf file parsing +// ---------------------------------------------------------------------------- + +struct StringToEnum { + const char *name; + uint32_t value; +}; + +#define STRING_TO_ENUM(string) { #string, string } +#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0])) + +const StringToEnum sDeviceNameToEnumTable[] = { + STRING_TO_ENUM(AUDIO_DEVICE_OUT_EARPIECE), + STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPEAKER), + STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPEAKER_SAFE), + STRING_TO_ENUM(AUDIO_DEVICE_OUT_WIRED_HEADSET), + STRING_TO_ENUM(AUDIO_DEVICE_OUT_WIRED_HEADPHONE), + STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_SCO), + STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET), + STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT), + STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_SCO), + STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_A2DP), + STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES), + STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER), + STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_A2DP), + STRING_TO_ENUM(AUDIO_DEVICE_OUT_AUX_DIGITAL), + STRING_TO_ENUM(AUDIO_DEVICE_OUT_HDMI), + STRING_TO_ENUM(AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET), + STRING_TO_ENUM(AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET), + STRING_TO_ENUM(AUDIO_DEVICE_OUT_USB_ACCESSORY), + STRING_TO_ENUM(AUDIO_DEVICE_OUT_USB_DEVICE), + STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_USB), + STRING_TO_ENUM(AUDIO_DEVICE_OUT_REMOTE_SUBMIX), + STRING_TO_ENUM(AUDIO_DEVICE_OUT_TELEPHONY_TX), + STRING_TO_ENUM(AUDIO_DEVICE_OUT_LINE), + STRING_TO_ENUM(AUDIO_DEVICE_OUT_HDMI_ARC), + STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPDIF), + STRING_TO_ENUM(AUDIO_DEVICE_OUT_FM), + STRING_TO_ENUM(AUDIO_DEVICE_OUT_AUX_LINE), + STRING_TO_ENUM(AUDIO_DEVICE_IN_AMBIENT), + STRING_TO_ENUM(AUDIO_DEVICE_IN_BUILTIN_MIC), + STRING_TO_ENUM(AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET), + STRING_TO_ENUM(AUDIO_DEVICE_IN_ALL_SCO), + STRING_TO_ENUM(AUDIO_DEVICE_IN_WIRED_HEADSET), + STRING_TO_ENUM(AUDIO_DEVICE_IN_AUX_DIGITAL), + STRING_TO_ENUM(AUDIO_DEVICE_IN_HDMI), + STRING_TO_ENUM(AUDIO_DEVICE_IN_TELEPHONY_RX), + STRING_TO_ENUM(AUDIO_DEVICE_IN_VOICE_CALL), + STRING_TO_ENUM(AUDIO_DEVICE_IN_BACK_MIC), + STRING_TO_ENUM(AUDIO_DEVICE_IN_REMOTE_SUBMIX), + STRING_TO_ENUM(AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET), + STRING_TO_ENUM(AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET), + STRING_TO_ENUM(AUDIO_DEVICE_IN_USB_ACCESSORY), + STRING_TO_ENUM(AUDIO_DEVICE_IN_USB_DEVICE), + STRING_TO_ENUM(AUDIO_DEVICE_IN_FM_TUNER), + STRING_TO_ENUM(AUDIO_DEVICE_IN_TV_TUNER), + STRING_TO_ENUM(AUDIO_DEVICE_IN_LINE), + STRING_TO_ENUM(AUDIO_DEVICE_IN_SPDIF), + STRING_TO_ENUM(AUDIO_DEVICE_IN_BLUETOOTH_A2DP), + STRING_TO_ENUM(AUDIO_DEVICE_IN_LOOPBACK), +}; + +const StringToEnum sOutputFlagNameToEnumTable[] = { + STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_DIRECT), + STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_PRIMARY), + STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_FAST), + STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_DEEP_BUFFER), + STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD), + STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_NON_BLOCKING), + STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_HW_AV_SYNC), +}; + +const StringToEnum sInputFlagNameToEnumTable[] = { + STRING_TO_ENUM(AUDIO_INPUT_FLAG_FAST), + STRING_TO_ENUM(AUDIO_INPUT_FLAG_HW_HOTWORD), +}; + +const StringToEnum sFormatNameToEnumTable[] = { + STRING_TO_ENUM(AUDIO_FORMAT_PCM_16_BIT), + STRING_TO_ENUM(AUDIO_FORMAT_PCM_8_BIT), + STRING_TO_ENUM(AUDIO_FORMAT_PCM_32_BIT), + STRING_TO_ENUM(AUDIO_FORMAT_PCM_8_24_BIT), + STRING_TO_ENUM(AUDIO_FORMAT_PCM_FLOAT), + STRING_TO_ENUM(AUDIO_FORMAT_PCM_24_BIT_PACKED), + STRING_TO_ENUM(AUDIO_FORMAT_MP3), + STRING_TO_ENUM(AUDIO_FORMAT_AAC), + STRING_TO_ENUM(AUDIO_FORMAT_AAC_MAIN), + STRING_TO_ENUM(AUDIO_FORMAT_AAC_LC), + STRING_TO_ENUM(AUDIO_FORMAT_AAC_SSR), + STRING_TO_ENUM(AUDIO_FORMAT_AAC_LTP), + STRING_TO_ENUM(AUDIO_FORMAT_AAC_HE_V1), + STRING_TO_ENUM(AUDIO_FORMAT_AAC_SCALABLE), + STRING_TO_ENUM(AUDIO_FORMAT_AAC_ERLC), + STRING_TO_ENUM(AUDIO_FORMAT_AAC_LD), + STRING_TO_ENUM(AUDIO_FORMAT_AAC_HE_V2), + STRING_TO_ENUM(AUDIO_FORMAT_AAC_ELD), + STRING_TO_ENUM(AUDIO_FORMAT_VORBIS), + STRING_TO_ENUM(AUDIO_FORMAT_HE_AAC_V1), + STRING_TO_ENUM(AUDIO_FORMAT_HE_AAC_V2), + STRING_TO_ENUM(AUDIO_FORMAT_OPUS), + STRING_TO_ENUM(AUDIO_FORMAT_AC3), + STRING_TO_ENUM(AUDIO_FORMAT_E_AC3), +}; + +const StringToEnum sOutChannelsNameToEnumTable[] = { + STRING_TO_ENUM(AUDIO_CHANNEL_OUT_MONO), + STRING_TO_ENUM(AUDIO_CHANNEL_OUT_STEREO), + STRING_TO_ENUM(AUDIO_CHANNEL_OUT_QUAD), + STRING_TO_ENUM(AUDIO_CHANNEL_OUT_5POINT1), + STRING_TO_ENUM(AUDIO_CHANNEL_OUT_7POINT1), +}; + +const StringToEnum sInChannelsNameToEnumTable[] = { + STRING_TO_ENUM(AUDIO_CHANNEL_IN_MONO), + STRING_TO_ENUM(AUDIO_CHANNEL_IN_STEREO), + STRING_TO_ENUM(AUDIO_CHANNEL_IN_FRONT_BACK), +}; + +const StringToEnum sGainModeNameToEnumTable[] = { + STRING_TO_ENUM(AUDIO_GAIN_MODE_JOINT), + STRING_TO_ENUM(AUDIO_GAIN_MODE_CHANNELS), + STRING_TO_ENUM(AUDIO_GAIN_MODE_RAMP), +}; + +class ConfigParsingUtils +{ +public: + static uint32_t stringToEnum(const struct StringToEnum *table, + size_t size, + const char *name); + static const char *enumToString(const struct StringToEnum *table, + size_t size, + uint32_t value); + static bool stringToBool(const char *value); + static uint32_t parseOutputFlagNames(char *name); + static uint32_t parseInputFlagNames(char *name); + static audio_devices_t parseDeviceNames(char *name); +}; + +}; // namespace android diff --git a/services/audiopolicy/managerdefault/Devices.cpp b/services/audiopolicy/managerdefault/Devices.cpp new file mode 100644 index 0000000..13c8bbc --- /dev/null +++ b/services/audiopolicy/managerdefault/Devices.cpp @@ -0,0 +1,282 @@ +/* + * Copyright (C) 2015 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#define LOG_TAG "APM::Devices" +//#define LOG_NDEBUG 0 + +#include "AudioPolicyManager.h" + +namespace android { + +String8 DeviceDescriptor::emptyNameStr = String8(""); + +DeviceDescriptor::DeviceDescriptor(const String8& name, audio_devices_t type) : + AudioPort(name, AUDIO_PORT_TYPE_DEVICE, + audio_is_output_device(type) ? AUDIO_PORT_ROLE_SINK : + AUDIO_PORT_ROLE_SOURCE, + NULL), + mDeviceType(type), mAddress("") +{ + +} + +bool DeviceDescriptor::equals(const sp& other) const +{ + // Devices are considered equal if they: + // - are of the same type (a device type cannot be AUDIO_DEVICE_NONE) + // - have the same address or one device does not specify the address + // - have the same channel mask or one device does not specify the channel mask + return (mDeviceType == other->mDeviceType) && + (mAddress == "" || other->mAddress == "" || mAddress == other->mAddress) && + (mChannelMask == 0 || other->mChannelMask == 0 || + mChannelMask == other->mChannelMask); +} + +void DeviceDescriptor::loadGains(cnode *root) +{ + AudioPort::loadGains(root); + if (mGains.size() > 0) { + mGains[0]->getDefaultConfig(&mGain); + } +} + +void DeviceVector::refreshTypes() +{ + mDeviceTypes = AUDIO_DEVICE_NONE; + for(size_t i = 0; i < size(); i++) { + mDeviceTypes |= itemAt(i)->mDeviceType; + } + ALOGV("DeviceVector::refreshTypes() mDeviceTypes %08x", mDeviceTypes); +} + +ssize_t DeviceVector::indexOf(const sp& item) const +{ + for(size_t i = 0; i < size(); i++) { + if (item->equals(itemAt(i))) { + return i; + } + } + return -1; +} + +ssize_t DeviceVector::add(const sp& item) +{ + ssize_t ret = indexOf(item); + + if (ret < 0) { + ret = SortedVector::add(item); + if (ret >= 0) { + refreshTypes(); + } + } else { + ALOGW("DeviceVector::add device %08x already in", item->mDeviceType); + ret = -1; + } + return ret; +} + +ssize_t DeviceVector::remove(const sp& item) +{ + size_t i; + ssize_t ret = indexOf(item); + + if (ret < 0) { + ALOGW("DeviceVector::remove device %08x not in", item->mDeviceType); + } else { + ret = SortedVector::removeAt(ret); + if (ret >= 0) { + refreshTypes(); + } + } + return ret; +} + +void DeviceVector::loadDevicesFromType(audio_devices_t types) +{ + DeviceVector deviceList; + + uint32_t role_bit = AUDIO_DEVICE_BIT_IN & types; + types &= ~role_bit; + + while (types) { + uint32_t i = 31 - __builtin_clz(types); + uint32_t type = 1 << i; + types &= ~type; + add(new DeviceDescriptor(String8("device_type"), type | role_bit)); + } +} + +void DeviceVector::loadDevicesFromName(char *name, + const DeviceVector& declaredDevices) +{ + char *devName = strtok(name, "|"); + while (devName != NULL) { + if (strlen(devName) != 0) { + audio_devices_t type = ConfigParsingUtils::stringToEnum(sDeviceNameToEnumTable, + ARRAY_SIZE(sDeviceNameToEnumTable), + devName); + if (type != AUDIO_DEVICE_NONE) { + sp dev = new DeviceDescriptor(String8(name), type); + if (type == AUDIO_DEVICE_IN_REMOTE_SUBMIX || + type == AUDIO_DEVICE_OUT_REMOTE_SUBMIX ) { + dev->mAddress = String8("0"); + } + add(dev); + } else { + sp deviceDesc = + declaredDevices.getDeviceFromName(String8(devName)); + if (deviceDesc != 0) { + add(deviceDesc); + } + } + } + devName = strtok(NULL, "|"); + } +} + +sp DeviceVector::getDevice(audio_devices_t type, String8 address) const +{ + sp device; + for (size_t i = 0; i < size(); i++) { + if (itemAt(i)->mDeviceType == type) { + if (address == "" || itemAt(i)->mAddress == address) { + device = itemAt(i); + if (itemAt(i)->mAddress == address) { + break; + } + } + } + } + ALOGV("DeviceVector::getDevice() for type %08x address %s found %p", + type, address.string(), device.get()); + return device; +} + +sp DeviceVector::getDeviceFromId(audio_port_handle_t id) const +{ + sp device; + for (size_t i = 0; i < size(); i++) { + if (itemAt(i)->getHandle() == id) { + device = itemAt(i); + break; + } + } + return device; +} + +DeviceVector DeviceVector::getDevicesFromType(audio_devices_t type) const +{ + DeviceVector devices; + for (size_t i = 0; (i < size()) && (type != AUDIO_DEVICE_NONE); i++) { + if (itemAt(i)->mDeviceType & type & ~AUDIO_DEVICE_BIT_IN) { + devices.add(itemAt(i)); + type &= ~itemAt(i)->mDeviceType; + ALOGV("DeviceVector::getDevicesFromType() for type %x found %p", + itemAt(i)->mDeviceType, itemAt(i).get()); + } + } + return devices; +} + +DeviceVector DeviceVector::getDevicesFromTypeAddr( + audio_devices_t type, String8 address) const +{ + DeviceVector devices; + for (size_t i = 0; i < size(); i++) { + if (itemAt(i)->mDeviceType == type) { + if (itemAt(i)->mAddress == address) { + devices.add(itemAt(i)); + } + } + } + return devices; +} + +sp DeviceVector::getDeviceFromName(const String8& name) const +{ + sp device; + for (size_t i = 0; i < size(); i++) { + if (itemAt(i)->mName == name) { + device = itemAt(i); + break; + } + } + return device; +} + +void DeviceDescriptor::toAudioPortConfig(struct audio_port_config *dstConfig, + const struct audio_port_config *srcConfig) const +{ + dstConfig->config_mask = AUDIO_PORT_CONFIG_CHANNEL_MASK|AUDIO_PORT_CONFIG_GAIN; + if (srcConfig != NULL) { + dstConfig->config_mask |= srcConfig->config_mask; + } + + AudioPortConfig::toAudioPortConfig(dstConfig, srcConfig); + + dstConfig->id = mId; + dstConfig->role = audio_is_output_device(mDeviceType) ? + AUDIO_PORT_ROLE_SINK : AUDIO_PORT_ROLE_SOURCE; + dstConfig->type = AUDIO_PORT_TYPE_DEVICE; + dstConfig->ext.device.type = mDeviceType; + + //TODO Understand why this test is necessary. i.e. why at boot time does it crash + // without the test? + // This has been demonstrated to NOT be true (at start up) + // ALOG_ASSERT(mModule != NULL); + dstConfig->ext.device.hw_module = mModule != NULL ? mModule->mHandle : NULL; + strncpy(dstConfig->ext.device.address, mAddress.string(), AUDIO_DEVICE_MAX_ADDRESS_LEN); +} + +void DeviceDescriptor::toAudioPort(struct audio_port *port) const +{ + ALOGV("DeviceDescriptor::toAudioPort() handle %d type %x", mId, mDeviceType); + AudioPort::toAudioPort(port); + port->id = mId; + toAudioPortConfig(&port->active_config); + port->ext.device.type = mDeviceType; + port->ext.device.hw_module = mModule->mHandle; + strncpy(port->ext.device.address, mAddress.string(), AUDIO_DEVICE_MAX_ADDRESS_LEN); +} + +status_t DeviceDescriptor::dump(int fd, int spaces, int index) const +{ + const size_t SIZE = 256; + char buffer[SIZE]; + String8 result; + + snprintf(buffer, SIZE, "%*sDevice %d:\n", spaces, "", index+1); + result.append(buffer); + if (mId != 0) { + snprintf(buffer, SIZE, "%*s- id: %2d\n", spaces, "", mId); + result.append(buffer); + } + snprintf(buffer, SIZE, "%*s- type: %-48s\n", spaces, "", + ConfigParsingUtils::enumToString(sDeviceNameToEnumTable, + ARRAY_SIZE(sDeviceNameToEnumTable), + mDeviceType)); + result.append(buffer); + if (mAddress.size() != 0) { + snprintf(buffer, SIZE, "%*s- address: %-32s\n", spaces, "", mAddress.string()); + result.append(buffer); + } + write(fd, result.string(), result.size()); + AudioPort::dump(fd, spaces); + + return NO_ERROR; +} + +}; // namespace android diff --git a/services/audiopolicy/managerdefault/Devices.h b/services/audiopolicy/managerdefault/Devices.h new file mode 100644 index 0000000..65e1416 --- /dev/null +++ b/services/audiopolicy/managerdefault/Devices.h @@ -0,0 +1,75 @@ +/* + * Copyright (C) 2015 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +namespace android { + +class AudioPort; +class AudioPortConfig; + +class DeviceDescriptor: public AudioPort, public AudioPortConfig +{ +public: + DeviceDescriptor(const String8& name, audio_devices_t type); + + virtual ~DeviceDescriptor() {} + + bool equals(const sp& other) const; + + // AudioPortConfig + virtual sp getAudioPort() const { return (AudioPort*) this; } + virtual void toAudioPortConfig(struct audio_port_config *dstConfig, + const struct audio_port_config *srcConfig = NULL) const; + + // AudioPort + virtual void loadGains(cnode *root); + virtual void toAudioPort(struct audio_port *port) const; + + status_t dump(int fd, int spaces, int index) const; + + audio_devices_t mDeviceType; + String8 mAddress; + audio_port_handle_t mId; + + static String8 emptyNameStr; +}; + +class DeviceVector : public SortedVector< sp > +{ +public: + DeviceVector() : SortedVector(), mDeviceTypes(AUDIO_DEVICE_NONE) {} + + ssize_t add(const sp& item); + ssize_t remove(const sp& item); + ssize_t indexOf(const sp& item) const; + + audio_devices_t types() const { return mDeviceTypes; } + + void loadDevicesFromType(audio_devices_t types); + void loadDevicesFromName(char *name, const DeviceVector& declaredDevices); + + sp getDevice(audio_devices_t type, String8 address) const; + DeviceVector getDevicesFromType(audio_devices_t types) const; + sp getDeviceFromId(audio_port_handle_t id) const; + sp getDeviceFromName(const String8& name) const; + DeviceVector getDevicesFromTypeAddr(audio_devices_t type, String8 address) + const; + +private: + void refreshTypes(); + audio_devices_t mDeviceTypes; +}; + +}; // namespace android diff --git a/services/audiopolicy/managerdefault/Gains.cpp b/services/audiopolicy/managerdefault/Gains.cpp new file mode 100644 index 0000000..4aca26d --- /dev/null +++ b/services/audiopolicy/managerdefault/Gains.cpp @@ -0,0 +1,446 @@ +/* + * Copyright (C) 2015 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#define LOG_TAG "APM::Gains" +//#define LOG_NDEBUG 0 + +//#define VERY_VERBOSE_LOGGING +#ifdef VERY_VERBOSE_LOGGING +#define ALOGVV ALOGV +#else +#define ALOGVV(a...) do { } while(0) +#endif + +#include "AudioPolicyManager.h" + +#include + +namespace android { + +const VolumeCurvePoint +ApmGains::sDefaultVolumeCurve[ApmGains::VOLCNT] = { + {1, -49.5f}, {33, -33.5f}, {66, -17.0f}, {100, 0.0f} +}; + + +const VolumeCurvePoint +ApmGains::sDefaultMediaVolumeCurve[ApmGains::VOLCNT] = { + {1, -58.0f}, {20, -40.0f}, {60, -17.0f}, {100, 0.0f} +}; + +const VolumeCurvePoint +ApmGains::sExtMediaSystemVolumeCurve[ApmGains::VOLCNT] = { + {1, -58.0f}, {20, -40.0f}, {60, -21.0f}, {100, -10.0f} +}; + +const VolumeCurvePoint +ApmGains::sSpeakerMediaVolumeCurve[ApmGains::VOLCNT] = { + {1, -56.0f}, {20, -34.0f}, {60, -11.0f}, {100, 0.0f} +}; + +const VolumeCurvePoint +ApmGains::sSpeakerMediaVolumeCurveDrc[ApmGains::VOLCNT] = { + {1, -55.0f}, {20, -43.0f}, {86, -12.0f}, {100, 0.0f} +}; + +const VolumeCurvePoint +ApmGains::sSpeakerSonificationVolumeCurve[ApmGains::VOLCNT] = { + {1, -29.7f}, {33, -20.1f}, {66, -10.2f}, {100, 0.0f} +}; + +const VolumeCurvePoint +ApmGains::sSpeakerSonificationVolumeCurveDrc[ApmGains::VOLCNT] = { + {1, -35.7f}, {33, -26.1f}, {66, -13.2f}, {100, 0.0f} +}; + +// AUDIO_STREAM_SYSTEM, AUDIO_STREAM_ENFORCED_AUDIBLE and AUDIO_STREAM_DTMF volume tracks +// AUDIO_STREAM_RING on phones and AUDIO_STREAM_MUSIC on tablets. +// AUDIO_STREAM_DTMF tracks AUDIO_STREAM_VOICE_CALL while in call (See AudioService.java). +// The range is constrained between -24dB and -6dB over speaker and -30dB and -18dB over headset. + +const VolumeCurvePoint +ApmGains::sDefaultSystemVolumeCurve[ApmGains::VOLCNT] = { + {1, -24.0f}, {33, -18.0f}, {66, -12.0f}, {100, -6.0f} +}; + +const VolumeCurvePoint +ApmGains::sDefaultSystemVolumeCurveDrc[ApmGains::VOLCNT] = { + {1, -34.0f}, {33, -24.0f}, {66, -15.0f}, {100, -6.0f} +}; + +const VolumeCurvePoint +ApmGains::sHeadsetSystemVolumeCurve[ApmGains::VOLCNT] = { + {1, -30.0f}, {33, -26.0f}, {66, -22.0f}, {100, -18.0f} +}; + +const VolumeCurvePoint +ApmGains::sDefaultVoiceVolumeCurve[ApmGains::VOLCNT] = { + {0, -42.0f}, {33, -28.0f}, {66, -14.0f}, {100, 0.0f} +}; + +const VolumeCurvePoint +ApmGains::sSpeakerVoiceVolumeCurve[ApmGains::VOLCNT] = { + {0, -24.0f}, {33, -16.0f}, {66, -8.0f}, {100, 0.0f} +}; + +const VolumeCurvePoint +ApmGains::sLinearVolumeCurve[ApmGains::VOLCNT] = { + {0, -96.0f}, {33, -68.0f}, {66, -34.0f}, {100, 0.0f} +}; + +const VolumeCurvePoint +ApmGains::sSilentVolumeCurve[ApmGains::VOLCNT] = { + {0, -96.0f}, {1, -96.0f}, {2, -96.0f}, {100, -96.0f} +}; + +const VolumeCurvePoint +ApmGains::sFullScaleVolumeCurve[ApmGains::VOLCNT] = { + {0, 0.0f}, {1, 0.0f}, {2, 0.0f}, {100, 0.0f} +}; + +const VolumeCurvePoint *ApmGains::sVolumeProfiles[AUDIO_STREAM_CNT] + [ApmGains::DEVICE_CATEGORY_CNT] = { + { // AUDIO_STREAM_VOICE_CALL + ApmGains::sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_HEADSET + ApmGains::sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_SPEAKER + ApmGains::sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_EARPIECE + ApmGains::sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA + }, + { // AUDIO_STREAM_SYSTEM + ApmGains::sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET + ApmGains::sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER + ApmGains::sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_EARPIECE + ApmGains::sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA + }, + { // AUDIO_STREAM_RING + ApmGains::sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET + ApmGains::sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER + ApmGains::sDefaultVolumeCurve, // DEVICE_CATEGORY_EARPIECE + ApmGains::sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA + }, + { // AUDIO_STREAM_MUSIC + ApmGains::sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_HEADSET + ApmGains::sSpeakerMediaVolumeCurve, // DEVICE_CATEGORY_SPEAKER + ApmGains::sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_EARPIECE + ApmGains::sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA + }, + { // AUDIO_STREAM_ALARM + ApmGains::sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET + ApmGains::sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER + ApmGains::sDefaultVolumeCurve, // DEVICE_CATEGORY_EARPIECE + ApmGains::sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA + }, + { // AUDIO_STREAM_NOTIFICATION + ApmGains::sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET + ApmGains::sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER + ApmGains::sDefaultVolumeCurve, // DEVICE_CATEGORY_EARPIECE + ApmGains::sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA + }, + { // AUDIO_STREAM_BLUETOOTH_SCO + ApmGains::sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_HEADSET + ApmGains::sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_SPEAKER + ApmGains::sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_EARPIECE + ApmGains::sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA + }, + { // AUDIO_STREAM_ENFORCED_AUDIBLE + ApmGains::sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET + ApmGains::sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER + ApmGains::sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_EARPIECE + ApmGains::sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA + }, + { // AUDIO_STREAM_DTMF + ApmGains::sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET + ApmGains::sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER + ApmGains::sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_EARPIECE + ApmGains::sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA + }, + { // AUDIO_STREAM_TTS + // "Transmitted Through Speaker": always silent except on DEVICE_CATEGORY_SPEAKER + ApmGains::sSilentVolumeCurve, // DEVICE_CATEGORY_HEADSET + ApmGains::sLinearVolumeCurve, // DEVICE_CATEGORY_SPEAKER + ApmGains::sSilentVolumeCurve, // DEVICE_CATEGORY_EARPIECE + ApmGains::sSilentVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA + }, + { // AUDIO_STREAM_ACCESSIBILITY + ApmGains::sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_HEADSET + ApmGains::sSpeakerMediaVolumeCurve, // DEVICE_CATEGORY_SPEAKER + ApmGains::sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_EARPIECE + ApmGains::sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA + }, + { // AUDIO_STREAM_REROUTING + ApmGains::sFullScaleVolumeCurve, // DEVICE_CATEGORY_HEADSET + ApmGains::sFullScaleVolumeCurve, // DEVICE_CATEGORY_SPEAKER + ApmGains::sFullScaleVolumeCurve, // DEVICE_CATEGORY_EARPIECE + ApmGains::sFullScaleVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA + }, + { // AUDIO_STREAM_PATCH + ApmGains::sFullScaleVolumeCurve, // DEVICE_CATEGORY_HEADSET + ApmGains::sFullScaleVolumeCurve, // DEVICE_CATEGORY_SPEAKER + ApmGains::sFullScaleVolumeCurve, // DEVICE_CATEGORY_EARPIECE + ApmGains::sFullScaleVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA + }, +}; + +//static +audio_devices_t ApmGains::getDeviceForVolume(audio_devices_t device) +{ + if (device == AUDIO_DEVICE_NONE) { + // this happens when forcing a route update and no track is active on an output. + // In this case the returned category is not important. + device = AUDIO_DEVICE_OUT_SPEAKER; + } else if (popcount(device) > 1) { + // Multiple device selection is either: + // - speaker + one other device: give priority to speaker in this case. + // - one A2DP device + another device: happens with duplicated output. In this case + // retain the device on the A2DP output as the other must not correspond to an active + // selection if not the speaker. + // - HDMI-CEC system audio mode only output: give priority to available item in order. + if (device & AUDIO_DEVICE_OUT_SPEAKER) { + device = AUDIO_DEVICE_OUT_SPEAKER; + } else if (device & AUDIO_DEVICE_OUT_HDMI_ARC) { + device = AUDIO_DEVICE_OUT_HDMI_ARC; + } else if (device & AUDIO_DEVICE_OUT_AUX_LINE) { + device = AUDIO_DEVICE_OUT_AUX_LINE; + } else if (device & AUDIO_DEVICE_OUT_SPDIF) { + device = AUDIO_DEVICE_OUT_SPDIF; + } else { + device = (audio_devices_t)(device & AUDIO_DEVICE_OUT_ALL_A2DP); + } + } + + /*SPEAKER_SAFE is an alias of SPEAKER for purposes of volume control*/ + if (device == AUDIO_DEVICE_OUT_SPEAKER_SAFE) + device = AUDIO_DEVICE_OUT_SPEAKER; + + ALOGW_IF(popcount(device) != 1, + "getDeviceForVolume() invalid device combination: %08x", + device); + + return device; +} + +//static +ApmGains::device_category ApmGains::getDeviceCategory(audio_devices_t device) +{ + switch(getDeviceForVolume(device)) { + case AUDIO_DEVICE_OUT_EARPIECE: + return ApmGains::DEVICE_CATEGORY_EARPIECE; + case AUDIO_DEVICE_OUT_WIRED_HEADSET: + case AUDIO_DEVICE_OUT_WIRED_HEADPHONE: + case AUDIO_DEVICE_OUT_BLUETOOTH_SCO: + case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET: + case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP: + case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES: + return ApmGains::DEVICE_CATEGORY_HEADSET; + case AUDIO_DEVICE_OUT_LINE: + case AUDIO_DEVICE_OUT_AUX_DIGITAL: + /*USB? Remote submix?*/ + return ApmGains::DEVICE_CATEGORY_EXT_MEDIA; + case AUDIO_DEVICE_OUT_SPEAKER: + case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT: + case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER: + case AUDIO_DEVICE_OUT_USB_ACCESSORY: + case AUDIO_DEVICE_OUT_USB_DEVICE: + case AUDIO_DEVICE_OUT_REMOTE_SUBMIX: + default: + return ApmGains::DEVICE_CATEGORY_SPEAKER; + } +} + +//static +float ApmGains::volIndexToAmpl(audio_devices_t device, const StreamDescriptor& streamDesc, + int indexInUi) +{ + ApmGains::device_category deviceCategory = ApmGains::getDeviceCategory(device); + const VolumeCurvePoint *curve = streamDesc.mVolumeCurve[deviceCategory]; + + // the volume index in the UI is relative to the min and max volume indices for this stream type + int nbSteps = 1 + curve[ApmGains::VOLMAX].mIndex - + curve[ApmGains::VOLMIN].mIndex; + int volIdx = (nbSteps * (indexInUi - streamDesc.mIndexMin)) / + (streamDesc.mIndexMax - streamDesc.mIndexMin); + + // find what part of the curve this index volume belongs to, or if it's out of bounds + int segment = 0; + if (volIdx < curve[ApmGains::VOLMIN].mIndex) { // out of bounds + return 0.0f; + } else if (volIdx < curve[ApmGains::VOLKNEE1].mIndex) { + segment = 0; + } else if (volIdx < curve[ApmGains::VOLKNEE2].mIndex) { + segment = 1; + } else if (volIdx <= curve[ApmGains::VOLMAX].mIndex) { + segment = 2; + } else { // out of bounds + return 1.0f; + } + + // linear interpolation in the attenuation table in dB + float decibels = curve[segment].mDBAttenuation + + ((float)(volIdx - curve[segment].mIndex)) * + ( (curve[segment+1].mDBAttenuation - + curve[segment].mDBAttenuation) / + ((float)(curve[segment+1].mIndex - + curve[segment].mIndex)) ); + + float amplification = exp( decibels * 0.115129f); // exp( dB * ln(10) / 20 ) + + ALOGVV("VOLUME vol index=[%d %d %d], dB=[%.1f %.1f %.1f] ampl=%.5f", + curve[segment].mIndex, volIdx, + curve[segment+1].mIndex, + curve[segment].mDBAttenuation, + decibels, + curve[segment+1].mDBAttenuation, + amplification); + + return amplification; +} + + + +AudioGain::AudioGain(int index, bool useInChannelMask) +{ + mIndex = index; + mUseInChannelMask = useInChannelMask; + memset(&mGain, 0, sizeof(struct audio_gain)); +} + +void AudioGain::getDefaultConfig(struct audio_gain_config *config) +{ + config->index = mIndex; + config->mode = mGain.mode; + config->channel_mask = mGain.channel_mask; + if ((mGain.mode & AUDIO_GAIN_MODE_JOINT) == AUDIO_GAIN_MODE_JOINT) { + config->values[0] = mGain.default_value; + } else { + uint32_t numValues; + if (mUseInChannelMask) { + numValues = audio_channel_count_from_in_mask(mGain.channel_mask); + } else { + numValues = audio_channel_count_from_out_mask(mGain.channel_mask); + } + for (size_t i = 0; i < numValues; i++) { + config->values[i] = mGain.default_value; + } + } + if ((mGain.mode & AUDIO_GAIN_MODE_RAMP) == AUDIO_GAIN_MODE_RAMP) { + config->ramp_duration_ms = mGain.min_ramp_ms; + } +} + +status_t AudioGain::checkConfig(const struct audio_gain_config *config) +{ + if ((config->mode & ~mGain.mode) != 0) { + return BAD_VALUE; + } + if ((config->mode & AUDIO_GAIN_MODE_JOINT) == AUDIO_GAIN_MODE_JOINT) { + if ((config->values[0] < mGain.min_value) || + (config->values[0] > mGain.max_value)) { + return BAD_VALUE; + } + } else { + if ((config->channel_mask & ~mGain.channel_mask) != 0) { + return BAD_VALUE; + } + uint32_t numValues; + if (mUseInChannelMask) { + numValues = audio_channel_count_from_in_mask(config->channel_mask); + } else { + numValues = audio_channel_count_from_out_mask(config->channel_mask); + } + for (size_t i = 0; i < numValues; i++) { + if ((config->values[i] < mGain.min_value) || + (config->values[i] > mGain.max_value)) { + return BAD_VALUE; + } + } + } + if ((config->mode & AUDIO_GAIN_MODE_RAMP) == AUDIO_GAIN_MODE_RAMP) { + if ((config->ramp_duration_ms < mGain.min_ramp_ms) || + (config->ramp_duration_ms > mGain.max_ramp_ms)) { + return BAD_VALUE; + } + } + return NO_ERROR; +} + +void AudioGain::dump(int fd, int spaces, int index) const +{ + const size_t SIZE = 256; + char buffer[SIZE]; + String8 result; + + snprintf(buffer, SIZE, "%*sGain %d:\n", spaces, "", index+1); + result.append(buffer); + snprintf(buffer, SIZE, "%*s- mode: %08x\n", spaces, "", mGain.mode); + result.append(buffer); + snprintf(buffer, SIZE, "%*s- channel_mask: %08x\n", spaces, "", mGain.channel_mask); + result.append(buffer); + snprintf(buffer, SIZE, "%*s- min_value: %d mB\n", spaces, "", mGain.min_value); + result.append(buffer); + snprintf(buffer, SIZE, "%*s- max_value: %d mB\n", spaces, "", mGain.max_value); + result.append(buffer); + snprintf(buffer, SIZE, "%*s- default_value: %d mB\n", spaces, "", mGain.default_value); + result.append(buffer); + snprintf(buffer, SIZE, "%*s- step_value: %d mB\n", spaces, "", mGain.step_value); + result.append(buffer); + snprintf(buffer, SIZE, "%*s- min_ramp_ms: %d ms\n", spaces, "", mGain.min_ramp_ms); + result.append(buffer); + snprintf(buffer, SIZE, "%*s- max_ramp_ms: %d ms\n", spaces, "", mGain.max_ramp_ms); + result.append(buffer); + + write(fd, result.string(), result.size()); +} + + +// --- StreamDescriptor class implementation + +StreamDescriptor::StreamDescriptor() + : mIndexMin(0), mIndexMax(1), mCanBeMuted(true) +{ + mIndexCur.add(AUDIO_DEVICE_OUT_DEFAULT, 0); +} + +int StreamDescriptor::getVolumeIndex(audio_devices_t device) +{ + device = ApmGains::getDeviceForVolume(device); + // there is always a valid entry for AUDIO_DEVICE_OUT_DEFAULT + if (mIndexCur.indexOfKey(device) < 0) { + device = AUDIO_DEVICE_OUT_DEFAULT; + } + return mIndexCur.valueFor(device); +} + +void StreamDescriptor::dump(int fd) +{ + const size_t SIZE = 256; + char buffer[SIZE]; + String8 result; + + snprintf(buffer, SIZE, "%s %02d %02d ", + mCanBeMuted ? "true " : "false", mIndexMin, mIndexMax); + result.append(buffer); + for (size_t i = 0; i < mIndexCur.size(); i++) { + snprintf(buffer, SIZE, "%04x : %02d, ", + mIndexCur.keyAt(i), + mIndexCur.valueAt(i)); + result.append(buffer); + } + result.append("\n"); + + write(fd, result.string(), result.size()); +} + +}; // namespace android diff --git a/services/audiopolicy/managerdefault/Gains.h b/services/audiopolicy/managerdefault/Gains.h new file mode 100644 index 0000000..b4ab129 --- /dev/null +++ b/services/audiopolicy/managerdefault/Gains.h @@ -0,0 +1,112 @@ +/* + * Copyright (C) 2015 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +namespace android { + +class VolumeCurvePoint +{ +public: + int mIndex; + float mDBAttenuation; +}; + +class StreamDescriptor; + +class ApmGains +{ +public : + // 4 points to define the volume attenuation curve, each characterized by the volume + // index (from 0 to 100) at which they apply, and the attenuation in dB at that index. + // we use 100 steps to avoid rounding errors when computing the volume in volIndexToAmpl() + enum { VOLMIN = 0, VOLKNEE1 = 1, VOLKNEE2 = 2, VOLMAX = 3, VOLCNT = 4}; + + // device categories used for volume curve management. + enum device_category { + DEVICE_CATEGORY_HEADSET, + DEVICE_CATEGORY_SPEAKER, + DEVICE_CATEGORY_EARPIECE, + DEVICE_CATEGORY_EXT_MEDIA, + DEVICE_CATEGORY_CNT + }; + + // returns the category the device belongs to with regard to volume curve management + static ApmGains::device_category getDeviceCategory(audio_devices_t device); + + // extract one device relevant for volume control from multiple device selection + static audio_devices_t getDeviceForVolume(audio_devices_t device); + + static float volIndexToAmpl(audio_devices_t device, const StreamDescriptor& streamDesc, + int indexInUi); + + // default volume curve + static const VolumeCurvePoint sDefaultVolumeCurve[ApmGains::VOLCNT]; + // default volume curve for media strategy + static const VolumeCurvePoint sDefaultMediaVolumeCurve[ApmGains::VOLCNT]; + // volume curve for non-media audio on ext media outputs (HDMI, Line, etc) + static const VolumeCurvePoint sExtMediaSystemVolumeCurve[ApmGains::VOLCNT]; + // volume curve for media strategy on speakers + static const VolumeCurvePoint sSpeakerMediaVolumeCurve[ApmGains::VOLCNT]; + static const VolumeCurvePoint sSpeakerMediaVolumeCurveDrc[ApmGains::VOLCNT]; + // volume curve for sonification strategy on speakers + static const VolumeCurvePoint sSpeakerSonificationVolumeCurve[ApmGains::VOLCNT]; + static const VolumeCurvePoint sSpeakerSonificationVolumeCurveDrc[ApmGains::VOLCNT]; + static const VolumeCurvePoint sDefaultSystemVolumeCurve[ApmGains::VOLCNT]; + static const VolumeCurvePoint sDefaultSystemVolumeCurveDrc[ApmGains::VOLCNT]; + static const VolumeCurvePoint sHeadsetSystemVolumeCurve[ApmGains::VOLCNT]; + static const VolumeCurvePoint sDefaultVoiceVolumeCurve[ApmGains::VOLCNT]; + static const VolumeCurvePoint sSpeakerVoiceVolumeCurve[ApmGains::VOLCNT]; + static const VolumeCurvePoint sLinearVolumeCurve[ApmGains::VOLCNT]; + static const VolumeCurvePoint sSilentVolumeCurve[ApmGains::VOLCNT]; + static const VolumeCurvePoint sFullScaleVolumeCurve[ApmGains::VOLCNT]; + // default volume curves per stream and device category. See initializeVolumeCurves() + static const VolumeCurvePoint *sVolumeProfiles[AUDIO_STREAM_CNT][ApmGains::DEVICE_CATEGORY_CNT]; +}; + + +class AudioGain: public RefBase +{ +public: + AudioGain(int index, bool useInChannelMask); + virtual ~AudioGain() {} + + void dump(int fd, int spaces, int index) const; + + void getDefaultConfig(struct audio_gain_config *config); + status_t checkConfig(const struct audio_gain_config *config); + int mIndex; + struct audio_gain mGain; + bool mUseInChannelMask; +}; + + +// stream descriptor used for volume control +class StreamDescriptor +{ +public: + StreamDescriptor(); + + int getVolumeIndex(audio_devices_t device); + void dump(int fd); + + int mIndexMin; // min volume index + int mIndexMax; // max volume index + KeyedVector mIndexCur; // current volume index per device + bool mCanBeMuted; // true is the stream can be muted + + const VolumeCurvePoint *mVolumeCurve[ApmGains::DEVICE_CATEGORY_CNT]; +}; + +}; // namespace android diff --git a/services/audiopolicy/managerdefault/HwModule.cpp b/services/audiopolicy/managerdefault/HwModule.cpp new file mode 100644 index 0000000..a04bdc8 --- /dev/null +++ b/services/audiopolicy/managerdefault/HwModule.cpp @@ -0,0 +1,279 @@ +/* + * Copyright (C) 2015 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#define LOG_TAG "APM::HwModule" +//#define LOG_NDEBUG 0 + +#include "AudioPolicyManager.h" +#include "audio_policy_conf.h" +#include + +namespace android { + +HwModule::HwModule(const char *name) + : mName(strndup(name, AUDIO_HARDWARE_MODULE_ID_MAX_LEN)), + mHalVersion(AUDIO_DEVICE_API_VERSION_MIN), mHandle(0) +{ +} + +HwModule::~HwModule() +{ + for (size_t i = 0; i < mOutputProfiles.size(); i++) { + mOutputProfiles[i]->mSupportedDevices.clear(); + } + for (size_t i = 0; i < mInputProfiles.size(); i++) { + mInputProfiles[i]->mSupportedDevices.clear(); + } + free((void *)mName); +} + +status_t HwModule::loadInput(cnode *root) +{ + cnode *node = root->first_child; + + sp profile = new IOProfile(String8(root->name), AUDIO_PORT_ROLE_SINK, this); + + while (node) { + if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) { + profile->loadSamplingRates((char *)node->value); + } else if (strcmp(node->name, FORMATS_TAG) == 0) { + profile->loadFormats((char *)node->value); + } else if (strcmp(node->name, CHANNELS_TAG) == 0) { + profile->loadInChannels((char *)node->value); + } else if (strcmp(node->name, DEVICES_TAG) == 0) { + profile->mSupportedDevices.loadDevicesFromName((char *)node->value, + mDeclaredDevices); + } else if (strcmp(node->name, FLAGS_TAG) == 0) { + profile->mFlags = ConfigParsingUtils::parseInputFlagNames((char *)node->value); + } else if (strcmp(node->name, GAINS_TAG) == 0) { + profile->loadGains(node); + } + node = node->next; + } + ALOGW_IF(profile->mSupportedDevices.isEmpty(), + "loadInput() invalid supported devices"); + ALOGW_IF(profile->mChannelMasks.size() == 0, + "loadInput() invalid supported channel masks"); + ALOGW_IF(profile->mSamplingRates.size() == 0, + "loadInput() invalid supported sampling rates"); + ALOGW_IF(profile->mFormats.size() == 0, + "loadInput() invalid supported formats"); + if (!profile->mSupportedDevices.isEmpty() && + (profile->mChannelMasks.size() != 0) && + (profile->mSamplingRates.size() != 0) && + (profile->mFormats.size() != 0)) { + + ALOGV("loadInput() adding input Supported Devices %04x", + profile->mSupportedDevices.types()); + + mInputProfiles.add(profile); + return NO_ERROR; + } else { + return BAD_VALUE; + } +} + +status_t HwModule::loadOutput(cnode *root) +{ + cnode *node = root->first_child; + + sp profile = new IOProfile(String8(root->name), AUDIO_PORT_ROLE_SOURCE, this); + + while (node) { + if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) { + profile->loadSamplingRates((char *)node->value); + } else if (strcmp(node->name, FORMATS_TAG) == 0) { + profile->loadFormats((char *)node->value); + } else if (strcmp(node->name, CHANNELS_TAG) == 0) { + profile->loadOutChannels((char *)node->value); + } else if (strcmp(node->name, DEVICES_TAG) == 0) { + profile->mSupportedDevices.loadDevicesFromName((char *)node->value, + mDeclaredDevices); + } else if (strcmp(node->name, FLAGS_TAG) == 0) { + profile->mFlags = ConfigParsingUtils::parseOutputFlagNames((char *)node->value); + } else if (strcmp(node->name, GAINS_TAG) == 0) { + profile->loadGains(node); + } + node = node->next; + } + ALOGW_IF(profile->mSupportedDevices.isEmpty(), + "loadOutput() invalid supported devices"); + ALOGW_IF(profile->mChannelMasks.size() == 0, + "loadOutput() invalid supported channel masks"); + ALOGW_IF(profile->mSamplingRates.size() == 0, + "loadOutput() invalid supported sampling rates"); + ALOGW_IF(profile->mFormats.size() == 0, + "loadOutput() invalid supported formats"); + if (!profile->mSupportedDevices.isEmpty() && + (profile->mChannelMasks.size() != 0) && + (profile->mSamplingRates.size() != 0) && + (profile->mFormats.size() != 0)) { + + ALOGV("loadOutput() adding output Supported Devices %04x, mFlags %04x", + profile->mSupportedDevices.types(), profile->mFlags); + + mOutputProfiles.add(profile); + return NO_ERROR; + } else { + return BAD_VALUE; + } +} + +status_t HwModule::loadDevice(cnode *root) +{ + cnode *node = root->first_child; + + audio_devices_t type = AUDIO_DEVICE_NONE; + while (node) { + if (strcmp(node->name, DEVICE_TYPE) == 0) { + type = ConfigParsingUtils::parseDeviceNames((char *)node->value); + break; + } + node = node->next; + } + if (type == AUDIO_DEVICE_NONE || + (!audio_is_input_device(type) && !audio_is_output_device(type))) { + ALOGW("loadDevice() bad type %08x", type); + return BAD_VALUE; + } + sp deviceDesc = new DeviceDescriptor(String8(root->name), type); + deviceDesc->mModule = this; + + node = root->first_child; + while (node) { + if (strcmp(node->name, DEVICE_ADDRESS) == 0) { + deviceDesc->mAddress = String8((char *)node->value); + } else if (strcmp(node->name, CHANNELS_TAG) == 0) { + if (audio_is_input_device(type)) { + deviceDesc->loadInChannels((char *)node->value); + } else { + deviceDesc->loadOutChannels((char *)node->value); + } + } else if (strcmp(node->name, GAINS_TAG) == 0) { + deviceDesc->loadGains(node); + } + node = node->next; + } + + ALOGV("loadDevice() adding device name %s type %08x address %s", + deviceDesc->mName.string(), type, deviceDesc->mAddress.string()); + + mDeclaredDevices.add(deviceDesc); + + return NO_ERROR; +} + +status_t HwModule::addOutputProfile(String8 name, const audio_config_t *config, + audio_devices_t device, String8 address) +{ + sp profile = new IOProfile(name, AUDIO_PORT_ROLE_SOURCE, this); + + profile->mSamplingRates.add(config->sample_rate); + profile->mChannelMasks.add(config->channel_mask); + profile->mFormats.add(config->format); + + sp devDesc = new DeviceDescriptor(name, device); + devDesc->mAddress = address; + profile->mSupportedDevices.add(devDesc); + + mOutputProfiles.add(profile); + + return NO_ERROR; +} + +status_t HwModule::removeOutputProfile(String8 name) +{ + for (size_t i = 0; i < mOutputProfiles.size(); i++) { + if (mOutputProfiles[i]->mName == name) { + mOutputProfiles.removeAt(i); + break; + } + } + + return NO_ERROR; +} + +status_t HwModule::addInputProfile(String8 name, const audio_config_t *config, + audio_devices_t device, String8 address) +{ + sp profile = new IOProfile(name, AUDIO_PORT_ROLE_SINK, this); + + profile->mSamplingRates.add(config->sample_rate); + profile->mChannelMasks.add(config->channel_mask); + profile->mFormats.add(config->format); + + sp devDesc = new DeviceDescriptor(name, device); + devDesc->mAddress = address; + profile->mSupportedDevices.add(devDesc); + + ALOGV("addInputProfile() name %s rate %d mask 0x08", name.string(), config->sample_rate, config->channel_mask); + + mInputProfiles.add(profile); + + return NO_ERROR; +} + +status_t HwModule::removeInputProfile(String8 name) +{ + for (size_t i = 0; i < mInputProfiles.size(); i++) { + if (mInputProfiles[i]->mName == name) { + mInputProfiles.removeAt(i); + break; + } + } + + return NO_ERROR; +} + + +void HwModule::dump(int fd) +{ + const size_t SIZE = 256; + char buffer[SIZE]; + String8 result; + + snprintf(buffer, SIZE, " - name: %s\n", mName); + result.append(buffer); + snprintf(buffer, SIZE, " - handle: %d\n", mHandle); + result.append(buffer); + snprintf(buffer, SIZE, " - version: %u.%u\n", mHalVersion >> 8, mHalVersion & 0xFF); + result.append(buffer); + write(fd, result.string(), result.size()); + if (mOutputProfiles.size()) { + write(fd, " - outputs:\n", strlen(" - outputs:\n")); + for (size_t i = 0; i < mOutputProfiles.size(); i++) { + snprintf(buffer, SIZE, " output %zu:\n", i); + write(fd, buffer, strlen(buffer)); + mOutputProfiles[i]->dump(fd); + } + } + if (mInputProfiles.size()) { + write(fd, " - inputs:\n", strlen(" - inputs:\n")); + for (size_t i = 0; i < mInputProfiles.size(); i++) { + snprintf(buffer, SIZE, " input %zu:\n", i); + write(fd, buffer, strlen(buffer)); + mInputProfiles[i]->dump(fd); + } + } + if (mDeclaredDevices.size()) { + write(fd, " - devices:\n", strlen(" - devices:\n")); + for (size_t i = 0; i < mDeclaredDevices.size(); i++) { + mDeclaredDevices[i]->dump(fd, 4, i); + } + } +} + +} //namespace android diff --git a/services/audiopolicy/managerdefault/HwModule.h b/services/audiopolicy/managerdefault/HwModule.h new file mode 100644 index 0000000..f814dd9 --- /dev/null +++ b/services/audiopolicy/managerdefault/HwModule.h @@ -0,0 +1,46 @@ +/* + * Copyright (C) 2015 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +namespace android { + +class HwModule : public RefBase +{ +public: + HwModule(const char *name); + ~HwModule(); + + status_t loadOutput(cnode *root); + status_t loadInput(cnode *root); + status_t loadDevice(cnode *root); + + status_t addOutputProfile(String8 name, const audio_config_t *config, + audio_devices_t device, String8 address); + status_t removeOutputProfile(String8 name); + status_t addInputProfile(String8 name, const audio_config_t *config, + audio_devices_t device, String8 address); + status_t removeInputProfile(String8 name); + + void dump(int fd); + + const char *const mName; // base name of the audio HW module (primary, a2dp ...) + uint32_t mHalVersion; // audio HAL API version + audio_module_handle_t mHandle; + Vector < sp > mOutputProfiles; // output profiles exposed by this module + Vector < sp > mInputProfiles; // input profiles exposed by this module + DeviceVector mDeclaredDevices; // devices declared in audio_policy.conf +}; + +}; // namespace android diff --git a/services/audiopolicy/managerdefault/IOProfile.cpp b/services/audiopolicy/managerdefault/IOProfile.cpp new file mode 100644 index 0000000..538ac1a --- /dev/null +++ b/services/audiopolicy/managerdefault/IOProfile.cpp @@ -0,0 +1,139 @@ +/* + * Copyright (C) 2015 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#define LOG_TAG "APM::IOProfile" +//#define LOG_NDEBUG 0 + +#include "AudioPolicyManager.h" + +namespace android { + +IOProfile::IOProfile(const String8& name, audio_port_role_t role, + const sp& module) + : AudioPort(name, AUDIO_PORT_TYPE_MIX, role, module) +{ +} + +IOProfile::~IOProfile() +{ +} + +// checks if the IO profile is compatible with specified parameters. +// Sampling rate, format and channel mask must be specified in order to +// get a valid a match +bool IOProfile::isCompatibleProfile(audio_devices_t device, + String8 address, + uint32_t samplingRate, + uint32_t *updatedSamplingRate, + audio_format_t format, + audio_channel_mask_t channelMask, + uint32_t flags) const +{ + const bool isPlaybackThread = mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SOURCE; + const bool isRecordThread = mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SINK; + ALOG_ASSERT(isPlaybackThread != isRecordThread); + + if (device != AUDIO_DEVICE_NONE && mSupportedDevices.getDevice(device, address) == 0) { + return false; + } + + if (samplingRate == 0) { + return false; + } + uint32_t myUpdatedSamplingRate = samplingRate; + if (isPlaybackThread && checkExactSamplingRate(samplingRate) != NO_ERROR) { + return false; + } + if (isRecordThread && checkCompatibleSamplingRate(samplingRate, &myUpdatedSamplingRate) != + NO_ERROR) { + return false; + } + + if (!audio_is_valid_format(format) || checkFormat(format) != NO_ERROR) { + return false; + } + + if (isPlaybackThread && (!audio_is_output_channel(channelMask) || + checkExactChannelMask(channelMask) != NO_ERROR)) { + return false; + } + if (isRecordThread && (!audio_is_input_channel(channelMask) || + checkCompatibleChannelMask(channelMask) != NO_ERROR)) { + return false; + } + + if (isPlaybackThread && (mFlags & flags) != flags) { + return false; + } + // The only input flag that is allowed to be different is the fast flag. + // An existing fast stream is compatible with a normal track request. + // An existing normal stream is compatible with a fast track request, + // but the fast request will be denied by AudioFlinger and converted to normal track. + if (isRecordThread && ((mFlags ^ flags) & + ~AUDIO_INPUT_FLAG_FAST)) { + return false; + } + + if (updatedSamplingRate != NULL) { + *updatedSamplingRate = myUpdatedSamplingRate; + } + return true; +} + +void IOProfile::dump(int fd) +{ + const size_t SIZE = 256; + char buffer[SIZE]; + String8 result; + + AudioPort::dump(fd, 4); + + snprintf(buffer, SIZE, " - flags: 0x%04x\n", mFlags); + result.append(buffer); + snprintf(buffer, SIZE, " - devices:\n"); + result.append(buffer); + write(fd, result.string(), result.size()); + for (size_t i = 0; i < mSupportedDevices.size(); i++) { + mSupportedDevices[i]->dump(fd, 6, i); + } +} + +void IOProfile::log() +{ + const size_t SIZE = 256; + char buffer[SIZE]; + String8 result; + + ALOGV(" - sampling rates: "); + for (size_t i = 0; i < mSamplingRates.size(); i++) { + ALOGV(" %d", mSamplingRates[i]); + } + + ALOGV(" - channel masks: "); + for (size_t i = 0; i < mChannelMasks.size(); i++) { + ALOGV(" 0x%04x", mChannelMasks[i]); + } + + ALOGV(" - formats: "); + for (size_t i = 0; i < mFormats.size(); i++) { + ALOGV(" 0x%08x", mFormats[i]); + } + + ALOGV(" - devices: 0x%04x\n", mSupportedDevices.types()); + ALOGV(" - flags: 0x%04x\n", mFlags); +} + +}; // namespace android diff --git a/services/audiopolicy/managerdefault/IOProfile.h b/services/audiopolicy/managerdefault/IOProfile.h new file mode 100644 index 0000000..3317969 --- /dev/null +++ b/services/audiopolicy/managerdefault/IOProfile.h @@ -0,0 +1,51 @@ +/* + * Copyright (C) 2015 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +namespace android { + +class HwModule; + +// the IOProfile class describes the capabilities of an output or input stream. +// It is currently assumed that all combination of listed parameters are supported. +// It is used by the policy manager to determine if an output or input is suitable for +// a given use case, open/close it accordingly and connect/disconnect audio tracks +// to/from it. +class IOProfile : public AudioPort +{ +public: + IOProfile(const String8& name, audio_port_role_t role, const sp& module); + virtual ~IOProfile(); + + // This method is used for both output and input. + // If parameter updatedSamplingRate is non-NULL, it is assigned the actual sample rate. + // For input, flags is interpreted as audio_input_flags_t. + // TODO: merge audio_output_flags_t and audio_input_flags_t. + bool isCompatibleProfile(audio_devices_t device, + String8 address, + uint32_t samplingRate, + uint32_t *updatedSamplingRate, + audio_format_t format, + audio_channel_mask_t channelMask, + uint32_t flags) const; + + void dump(int fd); + void log(); + + DeviceVector mSupportedDevices; // supported devices + // (devices this output can be routed to) +}; + +}; // namespace android diff --git a/services/audiopolicy/managerdefault/Ports.cpp b/services/audiopolicy/managerdefault/Ports.cpp new file mode 100644 index 0000000..3e55cee --- /dev/null +++ b/services/audiopolicy/managerdefault/Ports.cpp @@ -0,0 +1,844 @@ +/* + * Copyright (C) 2015 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#define LOG_TAG "APM::Ports" +//#define LOG_NDEBUG 0 + +#include "AudioPolicyManager.h" + +#include "audio_policy_conf.h" + +namespace android { + +// --- AudioPort class implementation + +AudioPort::AudioPort(const String8& name, audio_port_type_t type, + audio_port_role_t role, const sp& module) : + mName(name), mType(type), mRole(role), mModule(module), mFlags(0), mId(0) +{ + mUseInChannelMask = ((type == AUDIO_PORT_TYPE_DEVICE) && (role == AUDIO_PORT_ROLE_SOURCE)) || + ((type == AUDIO_PORT_TYPE_MIX) && (role == AUDIO_PORT_ROLE_SINK)); +} + +void AudioPort::attach(const sp& module) { + mId = AudioPolicyManager::nextUniqueId(); + mModule = module; +} + +void AudioPort::toAudioPort(struct audio_port *port) const +{ + port->role = mRole; + port->type = mType; + strlcpy(port->name, mName, AUDIO_PORT_MAX_NAME_LEN); + unsigned int i; + for (i = 0; i < mSamplingRates.size() && i < AUDIO_PORT_MAX_SAMPLING_RATES; i++) { + if (mSamplingRates[i] != 0) { + port->sample_rates[i] = mSamplingRates[i]; + } + } + port->num_sample_rates = i; + for (i = 0; i < mChannelMasks.size() && i < AUDIO_PORT_MAX_CHANNEL_MASKS; i++) { + if (mChannelMasks[i] != 0) { + port->channel_masks[i] = mChannelMasks[i]; + } + } + port->num_channel_masks = i; + for (i = 0; i < mFormats.size() && i < AUDIO_PORT_MAX_FORMATS; i++) { + if (mFormats[i] != 0) { + port->formats[i] = mFormats[i]; + } + } + port->num_formats = i; + + ALOGV("AudioPort::toAudioPort() num gains %zu", mGains.size()); + + for (i = 0; i < mGains.size() && i < AUDIO_PORT_MAX_GAINS; i++) { + port->gains[i] = mGains[i]->mGain; + } + port->num_gains = i; +} + +void AudioPort::importAudioPort(const sp port) { + for (size_t k = 0 ; k < port->mSamplingRates.size() ; k++) { + const uint32_t rate = port->mSamplingRates.itemAt(k); + if (rate != 0) { // skip "dynamic" rates + bool hasRate = false; + for (size_t l = 0 ; l < mSamplingRates.size() ; l++) { + if (rate == mSamplingRates.itemAt(l)) { + hasRate = true; + break; + } + } + if (!hasRate) { // never import a sampling rate twice + mSamplingRates.add(rate); + } + } + } + for (size_t k = 0 ; k < port->mChannelMasks.size() ; k++) { + const audio_channel_mask_t mask = port->mChannelMasks.itemAt(k); + if (mask != 0) { // skip "dynamic" masks + bool hasMask = false; + for (size_t l = 0 ; l < mChannelMasks.size() ; l++) { + if (mask == mChannelMasks.itemAt(l)) { + hasMask = true; + break; + } + } + if (!hasMask) { // never import a channel mask twice + mChannelMasks.add(mask); + } + } + } + for (size_t k = 0 ; k < port->mFormats.size() ; k++) { + const audio_format_t format = port->mFormats.itemAt(k); + if (format != 0) { // skip "dynamic" formats + bool hasFormat = false; + for (size_t l = 0 ; l < mFormats.size() ; l++) { + if (format == mFormats.itemAt(l)) { + hasFormat = true; + break; + } + } + if (!hasFormat) { // never import a channel mask twice + mFormats.add(format); + } + } + } + for (size_t k = 0 ; k < port->mGains.size() ; k++) { + sp gain = port->mGains.itemAt(k); + if (gain != 0) { + bool hasGain = false; + for (size_t l = 0 ; l < mGains.size() ; l++) { + if (gain == mGains.itemAt(l)) { + hasGain = true; + break; + } + } + if (!hasGain) { // never import a gain twice + mGains.add(gain); + } + } + } +} + +void AudioPort::clearCapabilities() { + mChannelMasks.clear(); + mFormats.clear(); + mSamplingRates.clear(); + mGains.clear(); +} + +void AudioPort::loadSamplingRates(char *name) +{ + char *str = strtok(name, "|"); + + // by convention, "0' in the first entry in mSamplingRates indicates the supported sampling + // rates should be read from the output stream after it is opened for the first time + if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) { + mSamplingRates.add(0); + return; + } + + while (str != NULL) { + uint32_t rate = atoi(str); + if (rate != 0) { + ALOGV("loadSamplingRates() adding rate %d", rate); + mSamplingRates.add(rate); + } + str = strtok(NULL, "|"); + } +} + +void AudioPort::loadFormats(char *name) +{ + char *str = strtok(name, "|"); + + // by convention, "0' in the first entry in mFormats indicates the supported formats + // should be read from the output stream after it is opened for the first time + if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) { + mFormats.add(AUDIO_FORMAT_DEFAULT); + return; + } + + while (str != NULL) { + audio_format_t format = (audio_format_t)ConfigParsingUtils::stringToEnum(sFormatNameToEnumTable, + ARRAY_SIZE(sFormatNameToEnumTable), + str); + if (format != AUDIO_FORMAT_DEFAULT) { + mFormats.add(format); + } + str = strtok(NULL, "|"); + } +} + +void AudioPort::loadInChannels(char *name) +{ + const char *str = strtok(name, "|"); + + ALOGV("loadInChannels() %s", name); + + if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) { + mChannelMasks.add(0); + return; + } + + while (str != NULL) { + audio_channel_mask_t channelMask = + (audio_channel_mask_t)ConfigParsingUtils::stringToEnum(sInChannelsNameToEnumTable, + ARRAY_SIZE(sInChannelsNameToEnumTable), + str); + if (channelMask != 0) { + ALOGV("loadInChannels() adding channelMask %04x", channelMask); + mChannelMasks.add(channelMask); + } + str = strtok(NULL, "|"); + } +} + +void AudioPort::loadOutChannels(char *name) +{ + const char *str = strtok(name, "|"); + + ALOGV("loadOutChannels() %s", name); + + // by convention, "0' in the first entry in mChannelMasks indicates the supported channel + // masks should be read from the output stream after it is opened for the first time + if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) { + mChannelMasks.add(0); + return; + } + + while (str != NULL) { + audio_channel_mask_t channelMask = + (audio_channel_mask_t)ConfigParsingUtils::stringToEnum(sOutChannelsNameToEnumTable, + ARRAY_SIZE(sOutChannelsNameToEnumTable), + str); + if (channelMask != 0) { + mChannelMasks.add(channelMask); + } + str = strtok(NULL, "|"); + } + return; +} + +audio_gain_mode_t AudioPort::loadGainMode(char *name) +{ + const char *str = strtok(name, "|"); + + ALOGV("loadGainMode() %s", name); + audio_gain_mode_t mode = 0; + while (str != NULL) { + mode |= (audio_gain_mode_t)ConfigParsingUtils::stringToEnum(sGainModeNameToEnumTable, + ARRAY_SIZE(sGainModeNameToEnumTable), + str); + str = strtok(NULL, "|"); + } + return mode; +} + +void AudioPort::loadGain(cnode *root, int index) +{ + cnode *node = root->first_child; + + sp gain = new AudioGain(index, mUseInChannelMask); + + while (node) { + if (strcmp(node->name, GAIN_MODE) == 0) { + gain->mGain.mode = loadGainMode((char *)node->value); + } else if (strcmp(node->name, GAIN_CHANNELS) == 0) { + if (mUseInChannelMask) { + gain->mGain.channel_mask = + (audio_channel_mask_t)ConfigParsingUtils::stringToEnum(sInChannelsNameToEnumTable, + ARRAY_SIZE(sInChannelsNameToEnumTable), + (char *)node->value); + } else { + gain->mGain.channel_mask = + (audio_channel_mask_t)ConfigParsingUtils::stringToEnum(sOutChannelsNameToEnumTable, + ARRAY_SIZE(sOutChannelsNameToEnumTable), + (char *)node->value); + } + } else if (strcmp(node->name, GAIN_MIN_VALUE) == 0) { + gain->mGain.min_value = atoi((char *)node->value); + } else if (strcmp(node->name, GAIN_MAX_VALUE) == 0) { + gain->mGain.max_value = atoi((char *)node->value); + } else if (strcmp(node->name, GAIN_DEFAULT_VALUE) == 0) { + gain->mGain.default_value = atoi((char *)node->value); + } else if (strcmp(node->name, GAIN_STEP_VALUE) == 0) { + gain->mGain.step_value = atoi((char *)node->value); + } else if (strcmp(node->name, GAIN_MIN_RAMP_MS) == 0) { + gain->mGain.min_ramp_ms = atoi((char *)node->value); + } else if (strcmp(node->name, GAIN_MAX_RAMP_MS) == 0) { + gain->mGain.max_ramp_ms = atoi((char *)node->value); + } + node = node->next; + } + + ALOGV("loadGain() adding new gain mode %08x channel mask %08x min mB %d max mB %d", + gain->mGain.mode, gain->mGain.channel_mask, gain->mGain.min_value, gain->mGain.max_value); + + if (gain->mGain.mode == 0) { + return; + } + mGains.add(gain); +} + +void AudioPort::loadGains(cnode *root) +{ + cnode *node = root->first_child; + int index = 0; + while (node) { + ALOGV("loadGains() loading gain %s", node->name); + loadGain(node, index++); + node = node->next; + } +} + +status_t AudioPort::checkExactSamplingRate(uint32_t samplingRate) const +{ + if (mSamplingRates.isEmpty()) { + return NO_ERROR; + } + + for (size_t i = 0; i < mSamplingRates.size(); i ++) { + if (mSamplingRates[i] == samplingRate) { + return NO_ERROR; + } + } + return BAD_VALUE; +} + +status_t AudioPort::checkCompatibleSamplingRate(uint32_t samplingRate, + uint32_t *updatedSamplingRate) const +{ + if (mSamplingRates.isEmpty()) { + return NO_ERROR; + } + + // Search for the closest supported sampling rate that is above (preferred) + // or below (acceptable) the desired sampling rate, within a permitted ratio. + // The sampling rates do not need to be sorted in ascending order. + ssize_t maxBelow = -1; + ssize_t minAbove = -1; + uint32_t candidate; + for (size_t i = 0; i < mSamplingRates.size(); i++) { + candidate = mSamplingRates[i]; + if (candidate == samplingRate) { + if (updatedSamplingRate != NULL) { + *updatedSamplingRate = candidate; + } + return NO_ERROR; + } + // candidate < desired + if (candidate < samplingRate) { + if (maxBelow < 0 || candidate > mSamplingRates[maxBelow]) { + maxBelow = i; + } + // candidate > desired + } else { + if (minAbove < 0 || candidate < mSamplingRates[minAbove]) { + minAbove = i; + } + } + } + // This uses hard-coded knowledge about AudioFlinger resampling ratios. + // TODO Move these assumptions out. + static const uint32_t kMaxDownSampleRatio = 6; // beyond this aliasing occurs + static const uint32_t kMaxUpSampleRatio = 256; // beyond this sample rate inaccuracies occur + // due to approximation by an int32_t of the + // phase increments + // Prefer to down-sample from a higher sampling rate, as we get the desired frequency spectrum. + if (minAbove >= 0) { + candidate = mSamplingRates[minAbove]; + if (candidate / kMaxDownSampleRatio <= samplingRate) { + if (updatedSamplingRate != NULL) { + *updatedSamplingRate = candidate; + } + return NO_ERROR; + } + } + // But if we have to up-sample from a lower sampling rate, that's OK. + if (maxBelow >= 0) { + candidate = mSamplingRates[maxBelow]; + if (candidate * kMaxUpSampleRatio >= samplingRate) { + if (updatedSamplingRate != NULL) { + *updatedSamplingRate = candidate; + } + return NO_ERROR; + } + } + // leave updatedSamplingRate unmodified + return BAD_VALUE; +} + +status_t AudioPort::checkExactChannelMask(audio_channel_mask_t channelMask) const +{ + if (mChannelMasks.isEmpty()) { + return NO_ERROR; + } + + for (size_t i = 0; i < mChannelMasks.size(); i++) { + if (mChannelMasks[i] == channelMask) { + return NO_ERROR; + } + } + return BAD_VALUE; +} + +status_t AudioPort::checkCompatibleChannelMask(audio_channel_mask_t channelMask) + const +{ + if (mChannelMasks.isEmpty()) { + return NO_ERROR; + } + + const bool isRecordThread = mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SINK; + for (size_t i = 0; i < mChannelMasks.size(); i ++) { + // FIXME Does not handle multi-channel automatic conversions yet + audio_channel_mask_t supported = mChannelMasks[i]; + if (supported == channelMask) { + return NO_ERROR; + } + if (isRecordThread) { + // This uses hard-coded knowledge that AudioFlinger can silently down-mix and up-mix. + // FIXME Abstract this out to a table. + if (((supported == AUDIO_CHANNEL_IN_FRONT_BACK || supported == AUDIO_CHANNEL_IN_STEREO) + && channelMask == AUDIO_CHANNEL_IN_MONO) || + (supported == AUDIO_CHANNEL_IN_MONO && (channelMask == AUDIO_CHANNEL_IN_FRONT_BACK + || channelMask == AUDIO_CHANNEL_IN_STEREO))) { + return NO_ERROR; + } + } + } + return BAD_VALUE; +} + +status_t AudioPort::checkFormat(audio_format_t format) const +{ + if (mFormats.isEmpty()) { + return NO_ERROR; + } + + for (size_t i = 0; i < mFormats.size(); i ++) { + if (mFormats[i] == format) { + return NO_ERROR; + } + } + return BAD_VALUE; +} + + +uint32_t AudioPort::pickSamplingRate() const +{ + // special case for uninitialized dynamic profile + if (mSamplingRates.size() == 1 && mSamplingRates[0] == 0) { + return 0; + } + + // For direct outputs, pick minimum sampling rate: this helps ensuring that the + // channel count / sampling rate combination chosen will be supported by the connected + // sink + if ((mType == AUDIO_PORT_TYPE_MIX) && (mRole == AUDIO_PORT_ROLE_SOURCE) && + (mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD))) { + uint32_t samplingRate = UINT_MAX; + for (size_t i = 0; i < mSamplingRates.size(); i ++) { + if ((mSamplingRates[i] < samplingRate) && (mSamplingRates[i] > 0)) { + samplingRate = mSamplingRates[i]; + } + } + return (samplingRate == UINT_MAX) ? 0 : samplingRate; + } + + uint32_t samplingRate = 0; + uint32_t maxRate = MAX_MIXER_SAMPLING_RATE; + + // For mixed output and inputs, use max mixer sampling rates. Do not + // limit sampling rate otherwise + if (mType != AUDIO_PORT_TYPE_MIX) { + maxRate = UINT_MAX; + } + for (size_t i = 0; i < mSamplingRates.size(); i ++) { + if ((mSamplingRates[i] > samplingRate) && (mSamplingRates[i] <= maxRate)) { + samplingRate = mSamplingRates[i]; + } + } + return samplingRate; +} + +audio_channel_mask_t AudioPort::pickChannelMask() const +{ + // special case for uninitialized dynamic profile + if (mChannelMasks.size() == 1 && mChannelMasks[0] == 0) { + return AUDIO_CHANNEL_NONE; + } + audio_channel_mask_t channelMask = AUDIO_CHANNEL_NONE; + + // For direct outputs, pick minimum channel count: this helps ensuring that the + // channel count / sampling rate combination chosen will be supported by the connected + // sink + if ((mType == AUDIO_PORT_TYPE_MIX) && (mRole == AUDIO_PORT_ROLE_SOURCE) && + (mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD))) { + uint32_t channelCount = UINT_MAX; + for (size_t i = 0; i < mChannelMasks.size(); i ++) { + uint32_t cnlCount; + if (mUseInChannelMask) { + cnlCount = audio_channel_count_from_in_mask(mChannelMasks[i]); + } else { + cnlCount = audio_channel_count_from_out_mask(mChannelMasks[i]); + } + if ((cnlCount < channelCount) && (cnlCount > 0)) { + channelMask = mChannelMasks[i]; + channelCount = cnlCount; + } + } + return channelMask; + } + + uint32_t channelCount = 0; + uint32_t maxCount = MAX_MIXER_CHANNEL_COUNT; + + // For mixed output and inputs, use max mixer channel count. Do not + // limit channel count otherwise + if (mType != AUDIO_PORT_TYPE_MIX) { + maxCount = UINT_MAX; + } + for (size_t i = 0; i < mChannelMasks.size(); i ++) { + uint32_t cnlCount; + if (mUseInChannelMask) { + cnlCount = audio_channel_count_from_in_mask(mChannelMasks[i]); + } else { + cnlCount = audio_channel_count_from_out_mask(mChannelMasks[i]); + } + if ((cnlCount > channelCount) && (cnlCount <= maxCount)) { + channelMask = mChannelMasks[i]; + channelCount = cnlCount; + } + } + return channelMask; +} + +/* format in order of increasing preference */ +const audio_format_t AudioPort::sPcmFormatCompareTable[] = { + AUDIO_FORMAT_DEFAULT, + AUDIO_FORMAT_PCM_16_BIT, + AUDIO_FORMAT_PCM_8_24_BIT, + AUDIO_FORMAT_PCM_24_BIT_PACKED, + AUDIO_FORMAT_PCM_32_BIT, + AUDIO_FORMAT_PCM_FLOAT, +}; + +int AudioPort::compareFormats(audio_format_t format1, + audio_format_t format2) +{ + // NOTE: AUDIO_FORMAT_INVALID is also considered not PCM and will be compared equal to any + // compressed format and better than any PCM format. This is by design of pickFormat() + if (!audio_is_linear_pcm(format1)) { + if (!audio_is_linear_pcm(format2)) { + return 0; + } + return 1; + } + if (!audio_is_linear_pcm(format2)) { + return -1; + } + + int index1 = -1, index2 = -1; + for (size_t i = 0; + (i < ARRAY_SIZE(sPcmFormatCompareTable)) && ((index1 == -1) || (index2 == -1)); + i ++) { + if (sPcmFormatCompareTable[i] == format1) { + index1 = i; + } + if (sPcmFormatCompareTable[i] == format2) { + index2 = i; + } + } + // format1 not found => index1 < 0 => format2 > format1 + // format2 not found => index2 < 0 => format2 < format1 + return index1 - index2; +} + +audio_format_t AudioPort::pickFormat() const +{ + // special case for uninitialized dynamic profile + if (mFormats.size() == 1 && mFormats[0] == 0) { + return AUDIO_FORMAT_DEFAULT; + } + + audio_format_t format = AUDIO_FORMAT_DEFAULT; + audio_format_t bestFormat = + AudioPort::sPcmFormatCompareTable[ + ARRAY_SIZE(AudioPort::sPcmFormatCompareTable) - 1]; + // For mixed output and inputs, use best mixer output format. Do not + // limit format otherwise + if ((mType != AUDIO_PORT_TYPE_MIX) || + ((mRole == AUDIO_PORT_ROLE_SOURCE) && + (((mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)) != 0)))) { + bestFormat = AUDIO_FORMAT_INVALID; + } + + for (size_t i = 0; i < mFormats.size(); i ++) { + if ((compareFormats(mFormats[i], format) > 0) && + (compareFormats(mFormats[i], bestFormat) <= 0)) { + format = mFormats[i]; + } + } + return format; +} + +status_t AudioPort::checkGain(const struct audio_gain_config *gainConfig, + int index) const +{ + if (index < 0 || (size_t)index >= mGains.size()) { + return BAD_VALUE; + } + return mGains[index]->checkConfig(gainConfig); +} + +void AudioPort::dump(int fd, int spaces) const +{ + const size_t SIZE = 256; + char buffer[SIZE]; + String8 result; + + if (mName.size() != 0) { + snprintf(buffer, SIZE, "%*s- name: %s\n", spaces, "", mName.string()); + result.append(buffer); + } + + if (mSamplingRates.size() != 0) { + snprintf(buffer, SIZE, "%*s- sampling rates: ", spaces, ""); + result.append(buffer); + for (size_t i = 0; i < mSamplingRates.size(); i++) { + if (i == 0 && mSamplingRates[i] == 0) { + snprintf(buffer, SIZE, "Dynamic"); + } else { + snprintf(buffer, SIZE, "%d", mSamplingRates[i]); + } + result.append(buffer); + result.append(i == (mSamplingRates.size() - 1) ? "" : ", "); + } + result.append("\n"); + } + + if (mChannelMasks.size() != 0) { + snprintf(buffer, SIZE, "%*s- channel masks: ", spaces, ""); + result.append(buffer); + for (size_t i = 0; i < mChannelMasks.size(); i++) { + ALOGV("AudioPort::dump mChannelMasks %zu %08x", i, mChannelMasks[i]); + + if (i == 0 && mChannelMasks[i] == 0) { + snprintf(buffer, SIZE, "Dynamic"); + } else { + snprintf(buffer, SIZE, "0x%04x", mChannelMasks[i]); + } + result.append(buffer); + result.append(i == (mChannelMasks.size() - 1) ? "" : ", "); + } + result.append("\n"); + } + + if (mFormats.size() != 0) { + snprintf(buffer, SIZE, "%*s- formats: ", spaces, ""); + result.append(buffer); + for (size_t i = 0; i < mFormats.size(); i++) { + const char *formatStr = ConfigParsingUtils::enumToString(sFormatNameToEnumTable, + ARRAY_SIZE(sFormatNameToEnumTable), + mFormats[i]); + if (i == 0 && strcmp(formatStr, "") == 0) { + snprintf(buffer, SIZE, "Dynamic"); + } else { + snprintf(buffer, SIZE, "%s", formatStr); + } + result.append(buffer); + result.append(i == (mFormats.size() - 1) ? "" : ", "); + } + result.append("\n"); + } + write(fd, result.string(), result.size()); + if (mGains.size() != 0) { + snprintf(buffer, SIZE, "%*s- gains:\n", spaces, ""); + write(fd, buffer, strlen(buffer) + 1); + result.append(buffer); + for (size_t i = 0; i < mGains.size(); i++) { + mGains[i]->dump(fd, spaces + 2, i); + } + } +} + + +// --- AudioPortConfig class implementation + +AudioPortConfig::AudioPortConfig() +{ + mSamplingRate = 0; + mChannelMask = AUDIO_CHANNEL_NONE; + mFormat = AUDIO_FORMAT_INVALID; + mGain.index = -1; +} + +status_t AudioPortConfig::applyAudioPortConfig( + const struct audio_port_config *config, + struct audio_port_config *backupConfig) +{ + struct audio_port_config localBackupConfig; + status_t status = NO_ERROR; + + localBackupConfig.config_mask = config->config_mask; + toAudioPortConfig(&localBackupConfig); + + sp audioport = getAudioPort(); + if (audioport == 0) { + status = NO_INIT; + goto exit; + } + if (config->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) { + status = audioport->checkExactSamplingRate(config->sample_rate); + if (status != NO_ERROR) { + goto exit; + } + mSamplingRate = config->sample_rate; + } + if (config->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) { + status = audioport->checkExactChannelMask(config->channel_mask); + if (status != NO_ERROR) { + goto exit; + } + mChannelMask = config->channel_mask; + } + if (config->config_mask & AUDIO_PORT_CONFIG_FORMAT) { + status = audioport->checkFormat(config->format); + if (status != NO_ERROR) { + goto exit; + } + mFormat = config->format; + } + if (config->config_mask & AUDIO_PORT_CONFIG_GAIN) { + status = audioport->checkGain(&config->gain, config->gain.index); + if (status != NO_ERROR) { + goto exit; + } + mGain = config->gain; + } + +exit: + if (status != NO_ERROR) { + applyAudioPortConfig(&localBackupConfig); + } + if (backupConfig != NULL) { + *backupConfig = localBackupConfig; + } + return status; +} + +void AudioPortConfig::toAudioPortConfig(struct audio_port_config *dstConfig, + const struct audio_port_config *srcConfig) const +{ + if (dstConfig->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) { + dstConfig->sample_rate = mSamplingRate; + if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE)) { + dstConfig->sample_rate = srcConfig->sample_rate; + } + } else { + dstConfig->sample_rate = 0; + } + if (dstConfig->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) { + dstConfig->channel_mask = mChannelMask; + if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK)) { + dstConfig->channel_mask = srcConfig->channel_mask; + } + } else { + dstConfig->channel_mask = AUDIO_CHANNEL_NONE; + } + if (dstConfig->config_mask & AUDIO_PORT_CONFIG_FORMAT) { + dstConfig->format = mFormat; + if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_FORMAT)) { + dstConfig->format = srcConfig->format; + } + } else { + dstConfig->format = AUDIO_FORMAT_INVALID; + } + if (dstConfig->config_mask & AUDIO_PORT_CONFIG_GAIN) { + dstConfig->gain = mGain; + if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_GAIN)) { + dstConfig->gain = srcConfig->gain; + } + } else { + dstConfig->gain.index = -1; + } + if (dstConfig->gain.index != -1) { + dstConfig->config_mask |= AUDIO_PORT_CONFIG_GAIN; + } else { + dstConfig->config_mask &= ~AUDIO_PORT_CONFIG_GAIN; + } +} + + +// --- AudioPatch class implementation + +AudioPatch::AudioPatch(audio_patch_handle_t handle, + const struct audio_patch *patch, uid_t uid) : + mHandle(handle), mPatch(*patch), mUid(uid), mAfPatchHandle(0) +{} + +status_t AudioPatch::dump(int fd, int spaces, int index) const +{ + const size_t SIZE = 256; + char buffer[SIZE]; + String8 result; + + snprintf(buffer, SIZE, "%*sAudio patch %d:\n", spaces, "", index+1); + result.append(buffer); + snprintf(buffer, SIZE, "%*s- handle: %2d\n", spaces, "", mHandle); + result.append(buffer); + snprintf(buffer, SIZE, "%*s- audio flinger handle: %2d\n", spaces, "", mAfPatchHandle); + result.append(buffer); + snprintf(buffer, SIZE, "%*s- owner uid: %2d\n", spaces, "", mUid); + result.append(buffer); + snprintf(buffer, SIZE, "%*s- %d sources:\n", spaces, "", mPatch.num_sources); + result.append(buffer); + for (size_t i = 0; i < mPatch.num_sources; i++) { + if (mPatch.sources[i].type == AUDIO_PORT_TYPE_DEVICE) { + snprintf(buffer, SIZE, "%*s- Device ID %d %s\n", spaces + 2, "", + mPatch.sources[i].id, ConfigParsingUtils::enumToString(sDeviceNameToEnumTable, + ARRAY_SIZE(sDeviceNameToEnumTable), + mPatch.sources[i].ext.device.type)); + } else { + snprintf(buffer, SIZE, "%*s- Mix ID %d I/O handle %d\n", spaces + 2, "", + mPatch.sources[i].id, mPatch.sources[i].ext.mix.handle); + } + result.append(buffer); + } + snprintf(buffer, SIZE, "%*s- %d sinks:\n", spaces, "", mPatch.num_sinks); + result.append(buffer); + for (size_t i = 0; i < mPatch.num_sinks; i++) { + if (mPatch.sinks[i].type == AUDIO_PORT_TYPE_DEVICE) { + snprintf(buffer, SIZE, "%*s- Device ID %d %s\n", spaces + 2, "", + mPatch.sinks[i].id, ConfigParsingUtils::enumToString(sDeviceNameToEnumTable, + ARRAY_SIZE(sDeviceNameToEnumTable), + mPatch.sinks[i].ext.device.type)); + } else { + snprintf(buffer, SIZE, "%*s- Mix ID %d I/O handle %d\n", spaces + 2, "", + mPatch.sinks[i].id, mPatch.sinks[i].ext.mix.handle); + } + result.append(buffer); + } + + write(fd, result.string(), result.size()); + return NO_ERROR; +} + + +}; // namespace android diff --git a/services/audiopolicy/managerdefault/Ports.h b/services/audiopolicy/managerdefault/Ports.h new file mode 100644 index 0000000..f6e0e93 --- /dev/null +++ b/services/audiopolicy/managerdefault/Ports.h @@ -0,0 +1,122 @@ +/* + * Copyright (C) 2015 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +namespace android { + +class HwModule; + +class AudioPort: public virtual RefBase +{ +public: + AudioPort(const String8& name, audio_port_type_t type, + audio_port_role_t role, const sp& module); + virtual ~AudioPort() {} + + audio_port_handle_t getHandle() { return mId; } + + void attach(const sp& module); + bool isAttached() { return mId != 0; } + + virtual void toAudioPort(struct audio_port *port) const; + + void importAudioPort(const sp port); + void clearCapabilities(); + + void loadSamplingRates(char *name); + void loadFormats(char *name); + void loadOutChannels(char *name); + void loadInChannels(char *name); + + audio_gain_mode_t loadGainMode(char *name); + void loadGain(cnode *root, int index); + virtual void loadGains(cnode *root); + + // searches for an exact match + status_t checkExactSamplingRate(uint32_t samplingRate) const; + // searches for a compatible match, and returns the best match via updatedSamplingRate + status_t checkCompatibleSamplingRate(uint32_t samplingRate, + uint32_t *updatedSamplingRate) const; + // searches for an exact match + status_t checkExactChannelMask(audio_channel_mask_t channelMask) const; + // searches for a compatible match, currently implemented for input channel masks only + status_t checkCompatibleChannelMask(audio_channel_mask_t channelMask) const; + status_t checkFormat(audio_format_t format) const; + status_t checkGain(const struct audio_gain_config *gainConfig, int index) const; + + uint32_t pickSamplingRate() const; + audio_channel_mask_t pickChannelMask() const; + audio_format_t pickFormat() const; + + static const audio_format_t sPcmFormatCompareTable[]; + static int compareFormats(audio_format_t format1, audio_format_t format2); + + void dump(int fd, int spaces) const; + + String8 mName; + audio_port_type_t mType; + audio_port_role_t mRole; + bool mUseInChannelMask; + // by convention, "0' in the first entry in mSamplingRates, mChannelMasks or mFormats + // indicates the supported parameters should be read from the output stream + // after it is opened for the first time + Vector mSamplingRates; // supported sampling rates + Vector mChannelMasks; // supported channel masks + Vector mFormats; // supported audio formats + Vector < sp > mGains; // gain controllers + sp mModule; // audio HW module exposing this I/O stream + uint32_t mFlags; // attribute flags (e.g primary output, + // direct output...). + + +protected: + //TODO - clarify the role of mId in this case, both an "attached" indicator + // and a unique ID for identifying a port to the (upcoming) selection API, + // and its relationship to the mId in AudioOutputDescriptor and AudioInputDescriptor. + audio_port_handle_t mId; +}; + +class AudioPortConfig: public virtual RefBase +{ +public: + AudioPortConfig(); + virtual ~AudioPortConfig() {} + + status_t applyAudioPortConfig(const struct audio_port_config *config, + struct audio_port_config *backupConfig = NULL); + virtual void toAudioPortConfig(struct audio_port_config *dstConfig, + const struct audio_port_config *srcConfig = NULL) const = 0; + virtual sp getAudioPort() const = 0; + uint32_t mSamplingRate; + audio_format_t mFormat; + audio_channel_mask_t mChannelMask; + struct audio_gain_config mGain; +}; + + +class AudioPatch: public RefBase +{ +public: + AudioPatch(audio_patch_handle_t handle, const struct audio_patch *patch, uid_t uid); + + status_t dump(int fd, int spaces, int index) const; + + audio_patch_handle_t mHandle; + struct audio_patch mPatch; + uid_t mUid; + audio_patch_handle_t mAfPatchHandle; +}; + +}; // namespace android diff --git a/services/audiopolicy/managerdefault/audio_policy_conf.h b/services/audiopolicy/managerdefault/audio_policy_conf.h new file mode 100644 index 0000000..2535a67 --- /dev/null +++ b/services/audiopolicy/managerdefault/audio_policy_conf.h @@ -0,0 +1,77 @@ +/* + * Copyright (C) 2012 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + + +#ifndef ANDROID_AUDIO_POLICY_CONF_H +#define ANDROID_AUDIO_POLICY_CONF_H + + +///////////////////////////////////////////////// +// Definitions for audio policy configuration file (audio_policy.conf) +///////////////////////////////////////////////// + +#define AUDIO_HARDWARE_MODULE_ID_MAX_LEN 32 + +#define AUDIO_POLICY_CONFIG_FILE "/system/etc/audio_policy.conf" +#define AUDIO_POLICY_VENDOR_CONFIG_FILE "/vendor/etc/audio_policy.conf" + +// global configuration +#define GLOBAL_CONFIG_TAG "global_configuration" + +#define ATTACHED_OUTPUT_DEVICES_TAG "attached_output_devices" +#define DEFAULT_OUTPUT_DEVICE_TAG "default_output_device" +#define ATTACHED_INPUT_DEVICES_TAG "attached_input_devices" +#define SPEAKER_DRC_ENABLED_TAG "speaker_drc_enabled" +#define AUDIO_HAL_VERSION_TAG "audio_hal_version" + +// hw modules descriptions +#define AUDIO_HW_MODULE_TAG "audio_hw_modules" + +#define OUTPUTS_TAG "outputs" +#define INPUTS_TAG "inputs" + +#define SAMPLING_RATES_TAG "sampling_rates" +#define FORMATS_TAG "formats" +#define CHANNELS_TAG "channel_masks" +#define DEVICES_TAG "devices" +#define FLAGS_TAG "flags" + +#define DYNAMIC_VALUE_TAG "dynamic" // special value for "channel_masks", "sampling_rates" and + // "formats" in outputs descriptors indicating that supported + // values should be queried after opening the output. + +#define DEVICES_TAG "devices" +#define DEVICE_TYPE "type" +#define DEVICE_ADDRESS "address" + +#define MIXERS_TAG "mixers" +#define MIXER_TYPE "type" +#define MIXER_TYPE_MUX "mux" +#define MIXER_TYPE_MIX "mix" + +#define GAINS_TAG "gains" +#define GAIN_MODE "mode" +#define GAIN_CHANNELS "channel_mask" +#define GAIN_MIN_VALUE "min_value_mB" +#define GAIN_MAX_VALUE "max_value_mB" +#define GAIN_DEFAULT_VALUE "default_value_mB" +#define GAIN_STEP_VALUE "step_value_mB" +#define GAIN_MIN_RAMP_MS "min_ramp_ms" +#define GAIN_MAX_RAMP_MS "max_ramp_ms" + + + +#endif // ANDROID_AUDIO_POLICY_CONF_H diff --git a/services/audiopolicy/service/AudioPolicyClientImpl.cpp b/services/audiopolicy/service/AudioPolicyClientImpl.cpp new file mode 100644 index 0000000..3e090e9 --- /dev/null +++ b/services/audiopolicy/service/AudioPolicyClientImpl.cpp @@ -0,0 +1,221 @@ +/* + * Copyright (C) 2009 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#define LOG_TAG "AudioPolicyClientImpl" +//#define LOG_NDEBUG 0 + +#include +#include +#include "AudioPolicyService.h" + +namespace android { + +/* implementation of the client interface from the policy manager */ + +audio_module_handle_t AudioPolicyService::AudioPolicyClient::loadHwModule(const char *name) +{ + sp af = AudioSystem::get_audio_flinger(); + if (af == 0) { + ALOGW("%s: could not get AudioFlinger", __func__); + return 0; + } + + return af->loadHwModule(name); +} + +status_t AudioPolicyService::AudioPolicyClient::openOutput(audio_module_handle_t module, + audio_io_handle_t *output, + audio_config_t *config, + audio_devices_t *devices, + const String8& address, + uint32_t *latencyMs, + audio_output_flags_t flags) +{ + sp af = AudioSystem::get_audio_flinger(); + if (af == 0) { + ALOGW("%s: could not get AudioFlinger", __func__); + return PERMISSION_DENIED; + } + return af->openOutput(module, output, config, devices, address, latencyMs, flags); +} + +audio_io_handle_t AudioPolicyService::AudioPolicyClient::openDuplicateOutput( + audio_io_handle_t output1, + audio_io_handle_t output2) +{ + sp af = AudioSystem::get_audio_flinger(); + if (af == 0) { + ALOGW("%s: could not get AudioFlinger", __func__); + return 0; + } + return af->openDuplicateOutput(output1, output2); +} + +status_t AudioPolicyService::AudioPolicyClient::closeOutput(audio_io_handle_t output) +{ + sp af = AudioSystem::get_audio_flinger(); + if (af == 0) { + return PERMISSION_DENIED; + } + + return af->closeOutput(output); +} + +status_t AudioPolicyService::AudioPolicyClient::suspendOutput(audio_io_handle_t output) +{ + sp af = AudioSystem::get_audio_flinger(); + if (af == 0) { + ALOGW("%s: could not get AudioFlinger", __func__); + return PERMISSION_DENIED; + } + + return af->suspendOutput(output); +} + +status_t AudioPolicyService::AudioPolicyClient::restoreOutput(audio_io_handle_t output) +{ + sp af = AudioSystem::get_audio_flinger(); + if (af == 0) { + ALOGW("%s: could not get AudioFlinger", __func__); + return PERMISSION_DENIED; + } + + return af->restoreOutput(output); +} + +status_t AudioPolicyService::AudioPolicyClient::openInput(audio_module_handle_t module, + audio_io_handle_t *input, + audio_config_t *config, + audio_devices_t *device, + const String8& address, + audio_source_t source, + audio_input_flags_t flags) +{ + sp af = AudioSystem::get_audio_flinger(); + if (af == 0) { + ALOGW("%s: could not get AudioFlinger", __func__); + return PERMISSION_DENIED; + } + + return af->openInput(module, input, config, device, address, source, flags); +} + +status_t AudioPolicyService::AudioPolicyClient::closeInput(audio_io_handle_t input) +{ + sp af = AudioSystem::get_audio_flinger(); + if (af == 0) { + return PERMISSION_DENIED; + } + + return af->closeInput(input); +} + +status_t AudioPolicyService::AudioPolicyClient::setStreamVolume(audio_stream_type_t stream, + float volume, audio_io_handle_t output, + int delay_ms) +{ + return mAudioPolicyService->setStreamVolume(stream, volume, output, + delay_ms); +} + +status_t AudioPolicyService::AudioPolicyClient::invalidateStream(audio_stream_type_t stream) +{ + sp af = AudioSystem::get_audio_flinger(); + if (af == 0) { + return PERMISSION_DENIED; + } + + return af->invalidateStream(stream); +} + +void AudioPolicyService::AudioPolicyClient::setParameters(audio_io_handle_t io_handle, + const String8& keyValuePairs, + int delay_ms) +{ + mAudioPolicyService->setParameters(io_handle, keyValuePairs.string(), delay_ms); +} + +String8 AudioPolicyService::AudioPolicyClient::getParameters(audio_io_handle_t io_handle, + const String8& keys) +{ + String8 result = AudioSystem::getParameters(io_handle, keys); + return result; +} + +status_t AudioPolicyService::AudioPolicyClient::startTone(audio_policy_tone_t tone, + audio_stream_type_t stream) +{ + return mAudioPolicyService->startTone(tone, stream); +} + +status_t AudioPolicyService::AudioPolicyClient::stopTone() +{ + return mAudioPolicyService->stopTone(); +} + +status_t AudioPolicyService::AudioPolicyClient::setVoiceVolume(float volume, int delay_ms) +{ + return mAudioPolicyService->setVoiceVolume(volume, delay_ms); +} + +status_t AudioPolicyService::AudioPolicyClient::moveEffects(int session, + audio_io_handle_t src_output, + audio_io_handle_t dst_output) +{ + sp af = AudioSystem::get_audio_flinger(); + if (af == 0) { + return PERMISSION_DENIED; + } + + return af->moveEffects(session, src_output, dst_output); +} + +status_t AudioPolicyService::AudioPolicyClient::createAudioPatch(const struct audio_patch *patch, + audio_patch_handle_t *handle, + int delayMs) +{ + return mAudioPolicyService->clientCreateAudioPatch(patch, handle, delayMs); +} + +status_t AudioPolicyService::AudioPolicyClient::releaseAudioPatch(audio_patch_handle_t handle, + int delayMs) +{ + return mAudioPolicyService->clientReleaseAudioPatch(handle, delayMs); +} + +status_t AudioPolicyService::AudioPolicyClient::setAudioPortConfig( + const struct audio_port_config *config, + int delayMs) +{ + return mAudioPolicyService->clientSetAudioPortConfig(config, delayMs); +} + +void AudioPolicyService::AudioPolicyClient::onAudioPortListUpdate() +{ + mAudioPolicyService->onAudioPortListUpdate(); +} + +void AudioPolicyService::AudioPolicyClient::onAudioPatchListUpdate() +{ + mAudioPolicyService->onAudioPatchListUpdate(); +} + +audio_unique_id_t AudioPolicyService::AudioPolicyClient::newAudioUniqueId() +{ + return AudioSystem::newAudioUniqueId(); +} + +}; // namespace android diff --git a/services/audiopolicy/service/AudioPolicyClientImplLegacy.cpp b/services/audiopolicy/service/AudioPolicyClientImplLegacy.cpp new file mode 100644 index 0000000..a79f8ae --- /dev/null +++ b/services/audiopolicy/service/AudioPolicyClientImplLegacy.cpp @@ -0,0 +1,316 @@ +/* + * Copyright (C) 2009 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#define LOG_TAG "AudioPolicyService" +//#define LOG_NDEBUG 0 + +#include "Configuration.h" +#undef __STRICT_ANSI__ +#define __STDINT_LIMITS +#define __STDC_LIMIT_MACROS +#include + +#include +#include +#include +#include +#include +#include +#include +#include "AudioPolicyService.h" +#include "ServiceUtilities.h" +#include +#include +#include +//#include + +#include +#include +#include +#include +#include +#include + + +namespace android { + +/* implementation of the interface to the policy manager */ +extern "C" { + +audio_module_handle_t aps_load_hw_module(void *service __unused, + const char *name) +{ + sp af = AudioSystem::get_audio_flinger(); + if (af == 0) { + ALOGW("%s: could not get AudioFlinger", __func__); + return 0; + } + + return af->loadHwModule(name); +} + +static audio_io_handle_t open_output(audio_module_handle_t module, + audio_devices_t *pDevices, + uint32_t *pSamplingRate, + audio_format_t *pFormat, + audio_channel_mask_t *pChannelMask, + uint32_t *pLatencyMs, + audio_output_flags_t flags, + const audio_offload_info_t *offloadInfo) +{ + sp af = AudioSystem::get_audio_flinger(); + if (af == 0) { + ALOGW("%s: could not get AudioFlinger", __func__); + return AUDIO_IO_HANDLE_NONE; + } + + if (pSamplingRate == NULL || pFormat == NULL || pChannelMask == NULL || + pDevices == NULL || pLatencyMs == NULL) { + return AUDIO_IO_HANDLE_NONE; + } + audio_config_t config = AUDIO_CONFIG_INITIALIZER; + config.sample_rate = *pSamplingRate; + config.format = *pFormat; + config.channel_mask = *pChannelMask; + if (offloadInfo != NULL) { + config.offload_info = *offloadInfo; + } + audio_io_handle_t output = AUDIO_IO_HANDLE_NONE; + status_t status = af->openOutput(module, &output, &config, pDevices, + String8(""), pLatencyMs, flags); + if (status == NO_ERROR) { + *pSamplingRate = config.sample_rate; + *pFormat = config.format; + *pChannelMask = config.channel_mask; + if (offloadInfo != NULL) { + *((audio_offload_info_t *)offloadInfo) = config.offload_info; + } + } + return output; +} + +// deprecated: replaced by aps_open_output_on_module() +audio_io_handle_t aps_open_output(void *service __unused, + audio_devices_t *pDevices, + uint32_t *pSamplingRate, + audio_format_t *pFormat, + audio_channel_mask_t *pChannelMask, + uint32_t *pLatencyMs, + audio_output_flags_t flags) +{ + return open_output((audio_module_handle_t)0, pDevices, pSamplingRate, pFormat, pChannelMask, + pLatencyMs, flags, NULL); +} + +audio_io_handle_t aps_open_output_on_module(void *service __unused, + audio_module_handle_t module, + audio_devices_t *pDevices, + uint32_t *pSamplingRate, + audio_format_t *pFormat, + audio_channel_mask_t *pChannelMask, + uint32_t *pLatencyMs, + audio_output_flags_t flags, + const audio_offload_info_t *offloadInfo) +{ + return open_output(module, pDevices, pSamplingRate, pFormat, pChannelMask, + pLatencyMs, flags, offloadInfo); +} + +audio_io_handle_t aps_open_dup_output(void *service __unused, + audio_io_handle_t output1, + audio_io_handle_t output2) +{ + sp af = AudioSystem::get_audio_flinger(); + if (af == 0) { + ALOGW("%s: could not get AudioFlinger", __func__); + return 0; + } + return af->openDuplicateOutput(output1, output2); +} + +int aps_close_output(void *service __unused, audio_io_handle_t output) +{ + sp af = AudioSystem::get_audio_flinger(); + if (af == 0) { + return PERMISSION_DENIED; + } + + return af->closeOutput(output); +} + +int aps_suspend_output(void *service __unused, audio_io_handle_t output) +{ + sp af = AudioSystem::get_audio_flinger(); + if (af == 0) { + ALOGW("%s: could not get AudioFlinger", __func__); + return PERMISSION_DENIED; + } + + return af->suspendOutput(output); +} + +int aps_restore_output(void *service __unused, audio_io_handle_t output) +{ + sp af = AudioSystem::get_audio_flinger(); + if (af == 0) { + ALOGW("%s: could not get AudioFlinger", __func__); + return PERMISSION_DENIED; + } + + return af->restoreOutput(output); +} + +static audio_io_handle_t open_input(audio_module_handle_t module, + audio_devices_t *pDevices, + uint32_t *pSamplingRate, + audio_format_t *pFormat, + audio_channel_mask_t *pChannelMask) +{ + sp af = AudioSystem::get_audio_flinger(); + if (af == 0) { + ALOGW("%s: could not get AudioFlinger", __func__); + return AUDIO_IO_HANDLE_NONE; + } + + if (pSamplingRate == NULL || pFormat == NULL || pChannelMask == NULL || pDevices == NULL) { + return AUDIO_IO_HANDLE_NONE; + } + + if (((*pDevices & AUDIO_DEVICE_IN_REMOTE_SUBMIX) == AUDIO_DEVICE_IN_REMOTE_SUBMIX) + && !captureAudioOutputAllowed()) { + ALOGE("open_input() permission denied: capture not allowed"); + return AUDIO_IO_HANDLE_NONE; + } + + audio_config_t config = AUDIO_CONFIG_INITIALIZER;; + config.sample_rate = *pSamplingRate; + config.format = *pFormat; + config.channel_mask = *pChannelMask; + audio_io_handle_t input = AUDIO_IO_HANDLE_NONE; + status_t status = af->openInput(module, &input, &config, pDevices, + String8(""), AUDIO_SOURCE_MIC, AUDIO_INPUT_FLAG_FAST /*FIXME*/); + if (status == NO_ERROR) { + *pSamplingRate = config.sample_rate; + *pFormat = config.format; + *pChannelMask = config.channel_mask; + } + return input; +} + + +// deprecated: replaced by aps_open_input_on_module(), and acoustics parameter is ignored +audio_io_handle_t aps_open_input(void *service __unused, + audio_devices_t *pDevices, + uint32_t *pSamplingRate, + audio_format_t *pFormat, + audio_channel_mask_t *pChannelMask, + audio_in_acoustics_t acoustics __unused) +{ + return open_input((audio_module_handle_t)0, pDevices, pSamplingRate, pFormat, pChannelMask); +} + +audio_io_handle_t aps_open_input_on_module(void *service __unused, + audio_module_handle_t module, + audio_devices_t *pDevices, + uint32_t *pSamplingRate, + audio_format_t *pFormat, + audio_channel_mask_t *pChannelMask) +{ + return open_input(module, pDevices, pSamplingRate, pFormat, pChannelMask); +} + +int aps_close_input(void *service __unused, audio_io_handle_t input) +{ + sp af = AudioSystem::get_audio_flinger(); + if (af == 0) { + return PERMISSION_DENIED; + } + + return af->closeInput(input); +} + +int aps_invalidate_stream(void *service __unused, audio_stream_type_t stream) +{ + sp af = AudioSystem::get_audio_flinger(); + if (af == 0) { + return PERMISSION_DENIED; + } + + return af->invalidateStream(stream); +} + +int aps_move_effects(void *service __unused, int session, + audio_io_handle_t src_output, + audio_io_handle_t dst_output) +{ + sp af = AudioSystem::get_audio_flinger(); + if (af == 0) { + return PERMISSION_DENIED; + } + + return af->moveEffects(session, src_output, dst_output); +} + +char * aps_get_parameters(void *service __unused, audio_io_handle_t io_handle, + const char *keys) +{ + String8 result = AudioSystem::getParameters(io_handle, String8(keys)); + return strdup(result.string()); +} + +void aps_set_parameters(void *service, audio_io_handle_t io_handle, + const char *kv_pairs, int delay_ms) +{ + AudioPolicyService *audioPolicyService = (AudioPolicyService *)service; + + audioPolicyService->setParameters(io_handle, kv_pairs, delay_ms); +} + +int aps_set_stream_volume(void *service, audio_stream_type_t stream, + float volume, audio_io_handle_t output, + int delay_ms) +{ + AudioPolicyService *audioPolicyService = (AudioPolicyService *)service; + + return audioPolicyService->setStreamVolume(stream, volume, output, + delay_ms); +} + +int aps_start_tone(void *service, audio_policy_tone_t tone, + audio_stream_type_t stream) +{ + AudioPolicyService *audioPolicyService = (AudioPolicyService *)service; + + return audioPolicyService->startTone(tone, stream); +} + +int aps_stop_tone(void *service) +{ + AudioPolicyService *audioPolicyService = (AudioPolicyService *)service; + + return audioPolicyService->stopTone(); +} + +int aps_set_voice_volume(void *service, float volume, int delay_ms) +{ + AudioPolicyService *audioPolicyService = (AudioPolicyService *)service; + + return audioPolicyService->setVoiceVolume(volume, delay_ms); +} + +}; // extern "C" + +}; // namespace android diff --git a/services/audiopolicy/service/AudioPolicyEffects.cpp b/services/audiopolicy/service/AudioPolicyEffects.cpp new file mode 100644 index 0000000..e6ace20 --- /dev/null +++ b/services/audiopolicy/service/AudioPolicyEffects.cpp @@ -0,0 +1,673 @@ +/* + * Copyright (C) 2014 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#define LOG_TAG "AudioPolicyEffects" +//#define LOG_NDEBUG 0 + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "AudioPolicyEffects.h" +#include "ServiceUtilities.h" + +namespace android { + +// ---------------------------------------------------------------------------- +// AudioPolicyEffects Implementation +// ---------------------------------------------------------------------------- + +AudioPolicyEffects::AudioPolicyEffects() +{ + // load automatic audio effect modules + if (access(AUDIO_EFFECT_VENDOR_CONFIG_FILE, R_OK) == 0) { + loadAudioEffectConfig(AUDIO_EFFECT_VENDOR_CONFIG_FILE); + } else if (access(AUDIO_EFFECT_DEFAULT_CONFIG_FILE, R_OK) == 0) { + loadAudioEffectConfig(AUDIO_EFFECT_DEFAULT_CONFIG_FILE); + } +} + + +AudioPolicyEffects::~AudioPolicyEffects() +{ + size_t i = 0; + // release audio input processing resources + for (i = 0; i < mInputSources.size(); i++) { + delete mInputSources.valueAt(i); + } + mInputSources.clear(); + + for (i = 0; i < mInputs.size(); i++) { + mInputs.valueAt(i)->mEffects.clear(); + delete mInputs.valueAt(i); + } + mInputs.clear(); + + // release audio output processing resources + for (i = 0; i < mOutputStreams.size(); i++) { + delete mOutputStreams.valueAt(i); + } + mOutputStreams.clear(); + + for (i = 0; i < mOutputSessions.size(); i++) { + mOutputSessions.valueAt(i)->mEffects.clear(); + delete mOutputSessions.valueAt(i); + } + mOutputSessions.clear(); +} + + +status_t AudioPolicyEffects::addInputEffects(audio_io_handle_t input, + audio_source_t inputSource, + int audioSession) +{ + status_t status = NO_ERROR; + + // create audio pre processors according to input source + audio_source_t aliasSource = (inputSource == AUDIO_SOURCE_HOTWORD) ? + AUDIO_SOURCE_VOICE_RECOGNITION : inputSource; + + Mutex::Autolock _l(mLock); + ssize_t index = mInputSources.indexOfKey(aliasSource); + if (index < 0) { + ALOGV("addInputEffects(): no processing needs to be attached to this source"); + return status; + } + ssize_t idx = mInputs.indexOfKey(input); + EffectVector *inputDesc; + if (idx < 0) { + inputDesc = new EffectVector(audioSession); + mInputs.add(input, inputDesc); + } else { + // EffectVector is existing and we just need to increase ref count + inputDesc = mInputs.valueAt(idx); + } + inputDesc->mRefCount++; + + ALOGV("addInputEffects(): input: %d, refCount: %d", input, inputDesc->mRefCount); + if (inputDesc->mRefCount == 1) { + Vector effects = mInputSources.valueAt(index)->mEffects; + for (size_t i = 0; i < effects.size(); i++) { + EffectDesc *effect = effects[i]; + sp fx = new AudioEffect(NULL, &effect->mUuid, -1, 0, 0, + audioSession, input); + status_t status = fx->initCheck(); + if (status != NO_ERROR && status != ALREADY_EXISTS) { + ALOGW("addInputEffects(): failed to create Fx %s on source %d", + effect->mName, (int32_t)aliasSource); + // fx goes out of scope and strong ref on AudioEffect is released + continue; + } + for (size_t j = 0; j < effect->mParams.size(); j++) { + fx->setParameter(effect->mParams[j]); + } + ALOGV("addInputEffects(): added Fx %s on source: %d", + effect->mName, (int32_t)aliasSource); + inputDesc->mEffects.add(fx); + } + inputDesc->setProcessorEnabled(true); + } + return status; +} + + +status_t AudioPolicyEffects::releaseInputEffects(audio_io_handle_t input) +{ + status_t status = NO_ERROR; + + Mutex::Autolock _l(mLock); + ssize_t index = mInputs.indexOfKey(input); + if (index < 0) { + return status; + } + EffectVector *inputDesc = mInputs.valueAt(index); + inputDesc->mRefCount--; + ALOGV("releaseInputEffects(): input: %d, refCount: %d", input, inputDesc->mRefCount); + if (inputDesc->mRefCount == 0) { + inputDesc->setProcessorEnabled(false); + delete inputDesc; + mInputs.removeItemsAt(index); + ALOGV("releaseInputEffects(): all effects released"); + } + return status; +} + +status_t AudioPolicyEffects::queryDefaultInputEffects(int audioSession, + effect_descriptor_t *descriptors, + uint32_t *count) +{ + status_t status = NO_ERROR; + + Mutex::Autolock _l(mLock); + size_t index; + for (index = 0; index < mInputs.size(); index++) { + if (mInputs.valueAt(index)->mSessionId == audioSession) { + break; + } + } + if (index == mInputs.size()) { + *count = 0; + return BAD_VALUE; + } + Vector< sp > effects = mInputs.valueAt(index)->mEffects; + + for (size_t i = 0; i < effects.size(); i++) { + effect_descriptor_t desc = effects[i]->descriptor(); + if (i < *count) { + descriptors[i] = desc; + } + } + if (effects.size() > *count) { + status = NO_MEMORY; + } + *count = effects.size(); + return status; +} + + +status_t AudioPolicyEffects::queryDefaultOutputSessionEffects(int audioSession, + effect_descriptor_t *descriptors, + uint32_t *count) +{ + status_t status = NO_ERROR; + + Mutex::Autolock _l(mLock); + size_t index; + for (index = 0; index < mOutputSessions.size(); index++) { + if (mOutputSessions.valueAt(index)->mSessionId == audioSession) { + break; + } + } + if (index == mOutputSessions.size()) { + *count = 0; + return BAD_VALUE; + } + Vector< sp > effects = mOutputSessions.valueAt(index)->mEffects; + + for (size_t i = 0; i < effects.size(); i++) { + effect_descriptor_t desc = effects[i]->descriptor(); + if (i < *count) { + descriptors[i] = desc; + } + } + if (effects.size() > *count) { + status = NO_MEMORY; + } + *count = effects.size(); + return status; +} + + +status_t AudioPolicyEffects::addOutputSessionEffects(audio_io_handle_t output, + audio_stream_type_t stream, + int audioSession) +{ + status_t status = NO_ERROR; + + Mutex::Autolock _l(mLock); + // create audio processors according to stream + // FIXME: should we have specific post processing settings for internal streams? + // default to media for now. + if (stream >= AUDIO_STREAM_PUBLIC_CNT) { + stream = AUDIO_STREAM_MUSIC; + } + ssize_t index = mOutputStreams.indexOfKey(stream); + if (index < 0) { + ALOGV("addOutputSessionEffects(): no output processing needed for this stream"); + return NO_ERROR; + } + + ssize_t idx = mOutputSessions.indexOfKey(audioSession); + EffectVector *procDesc; + if (idx < 0) { + procDesc = new EffectVector(audioSession); + mOutputSessions.add(audioSession, procDesc); + } else { + // EffectVector is existing and we just need to increase ref count + procDesc = mOutputSessions.valueAt(idx); + } + procDesc->mRefCount++; + + ALOGV("addOutputSessionEffects(): session: %d, refCount: %d", + audioSession, procDesc->mRefCount); + if (procDesc->mRefCount == 1) { + Vector effects = mOutputStreams.valueAt(index)->mEffects; + for (size_t i = 0; i < effects.size(); i++) { + EffectDesc *effect = effects[i]; + sp fx = new AudioEffect(NULL, &effect->mUuid, 0, 0, 0, + audioSession, output); + status_t status = fx->initCheck(); + if (status != NO_ERROR && status != ALREADY_EXISTS) { + ALOGE("addOutputSessionEffects(): failed to create Fx %s on session %d", + effect->mName, audioSession); + // fx goes out of scope and strong ref on AudioEffect is released + continue; + } + ALOGV("addOutputSessionEffects(): added Fx %s on session: %d for stream: %d", + effect->mName, audioSession, (int32_t)stream); + procDesc->mEffects.add(fx); + } + + procDesc->setProcessorEnabled(true); + } + return status; +} + +status_t AudioPolicyEffects::releaseOutputSessionEffects(audio_io_handle_t output, + audio_stream_type_t stream, + int audioSession) +{ + status_t status = NO_ERROR; + (void) output; // argument not used for now + (void) stream; // argument not used for now + + Mutex::Autolock _l(mLock); + ssize_t index = mOutputSessions.indexOfKey(audioSession); + if (index < 0) { + ALOGV("releaseOutputSessionEffects: no output processing was attached to this stream"); + return NO_ERROR; + } + + EffectVector *procDesc = mOutputSessions.valueAt(index); + procDesc->mRefCount--; + ALOGV("releaseOutputSessionEffects(): session: %d, refCount: %d", + audioSession, procDesc->mRefCount); + if (procDesc->mRefCount == 0) { + procDesc->setProcessorEnabled(false); + procDesc->mEffects.clear(); + delete procDesc; + mOutputSessions.removeItemsAt(index); + ALOGV("releaseOutputSessionEffects(): output processing released from session: %d", + audioSession); + } + return status; +} + + +void AudioPolicyEffects::EffectVector::setProcessorEnabled(bool enabled) +{ + for (size_t i = 0; i < mEffects.size(); i++) { + mEffects.itemAt(i)->setEnabled(enabled); + } +} + + +// ---------------------------------------------------------------------------- +// Audio processing configuration +// ---------------------------------------------------------------------------- + +/*static*/ const char * const AudioPolicyEffects::kInputSourceNames[AUDIO_SOURCE_CNT -1] = { + MIC_SRC_TAG, + VOICE_UL_SRC_TAG, + VOICE_DL_SRC_TAG, + VOICE_CALL_SRC_TAG, + CAMCORDER_SRC_TAG, + VOICE_REC_SRC_TAG, + VOICE_COMM_SRC_TAG +}; + +// returns the audio_source_t enum corresponding to the input source name or +// AUDIO_SOURCE_CNT is no match found +/*static*/ audio_source_t AudioPolicyEffects::inputSourceNameToEnum(const char *name) +{ + int i; + for (i = AUDIO_SOURCE_MIC; i < AUDIO_SOURCE_CNT; i++) { + if (strcmp(name, kInputSourceNames[i - AUDIO_SOURCE_MIC]) == 0) { + ALOGV("inputSourceNameToEnum found source %s %d", name, i); + break; + } + } + return (audio_source_t)i; +} + +const char *AudioPolicyEffects::kStreamNames[AUDIO_STREAM_PUBLIC_CNT+1] = { + AUDIO_STREAM_DEFAULT_TAG, + AUDIO_STREAM_VOICE_CALL_TAG, + AUDIO_STREAM_SYSTEM_TAG, + AUDIO_STREAM_RING_TAG, + AUDIO_STREAM_MUSIC_TAG, + AUDIO_STREAM_ALARM_TAG, + AUDIO_STREAM_NOTIFICATION_TAG, + AUDIO_STREAM_BLUETOOTH_SCO_TAG, + AUDIO_STREAM_ENFORCED_AUDIBLE_TAG, + AUDIO_STREAM_DTMF_TAG, + AUDIO_STREAM_TTS_TAG +}; + +// returns the audio_stream_t enum corresponding to the output stream name or +// AUDIO_STREAM_PUBLIC_CNT is no match found +audio_stream_type_t AudioPolicyEffects::streamNameToEnum(const char *name) +{ + int i; + for (i = AUDIO_STREAM_DEFAULT; i < AUDIO_STREAM_PUBLIC_CNT; i++) { + if (strcmp(name, kStreamNames[i - AUDIO_STREAM_DEFAULT]) == 0) { + ALOGV("streamNameToEnum found stream %s %d", name, i); + break; + } + } + return (audio_stream_type_t)i; +} + +// ---------------------------------------------------------------------------- +// Audio Effect Config parser +// ---------------------------------------------------------------------------- + +size_t AudioPolicyEffects::growParamSize(char *param, + size_t size, + size_t *curSize, + size_t *totSize) +{ + // *curSize is at least sizeof(effect_param_t) + 2 * sizeof(int) + size_t pos = ((*curSize - 1 ) / size + 1) * size; + + if (pos + size > *totSize) { + while (pos + size > *totSize) { + *totSize += ((*totSize + 7) / 8) * 4; + } + param = (char *)realloc(param, *totSize); + } + *curSize = pos + size; + return pos; +} + +size_t AudioPolicyEffects::readParamValue(cnode *node, + char *param, + size_t *curSize, + size_t *totSize) +{ + if (strncmp(node->name, SHORT_TAG, sizeof(SHORT_TAG) + 1) == 0) { + size_t pos = growParamSize(param, sizeof(short), curSize, totSize); + *(short *)((char *)param + pos) = (short)atoi(node->value); + ALOGV("readParamValue() reading short %d", *(short *)((char *)param + pos)); + return sizeof(short); + } else if (strncmp(node->name, INT_TAG, sizeof(INT_TAG) + 1) == 0) { + size_t pos = growParamSize(param, sizeof(int), curSize, totSize); + *(int *)((char *)param + pos) = atoi(node->value); + ALOGV("readParamValue() reading int %d", *(int *)((char *)param + pos)); + return sizeof(int); + } else if (strncmp(node->name, FLOAT_TAG, sizeof(FLOAT_TAG) + 1) == 0) { + size_t pos = growParamSize(param, sizeof(float), curSize, totSize); + *(float *)((char *)param + pos) = (float)atof(node->value); + ALOGV("readParamValue() reading float %f",*(float *)((char *)param + pos)); + return sizeof(float); + } else if (strncmp(node->name, BOOL_TAG, sizeof(BOOL_TAG) + 1) == 0) { + size_t pos = growParamSize(param, sizeof(bool), curSize, totSize); + if (strncmp(node->value, "false", strlen("false") + 1) == 0) { + *(bool *)((char *)param + pos) = false; + } else { + *(bool *)((char *)param + pos) = true; + } + ALOGV("readParamValue() reading bool %s",*(bool *)((char *)param + pos) ? "true" : "false"); + return sizeof(bool); + } else if (strncmp(node->name, STRING_TAG, sizeof(STRING_TAG) + 1) == 0) { + size_t len = strnlen(node->value, EFFECT_STRING_LEN_MAX); + if (*curSize + len + 1 > *totSize) { + *totSize = *curSize + len + 1; + param = (char *)realloc(param, *totSize); + } + strncpy(param + *curSize, node->value, len); + *curSize += len; + param[*curSize] = '\0'; + ALOGV("readParamValue() reading string %s", param + *curSize - len); + return len; + } + ALOGW("readParamValue() unknown param type %s", node->name); + return 0; +} + +effect_param_t *AudioPolicyEffects::loadEffectParameter(cnode *root) +{ + cnode *param; + cnode *value; + size_t curSize = sizeof(effect_param_t); + size_t totSize = sizeof(effect_param_t) + 2 * sizeof(int); + effect_param_t *fx_param = (effect_param_t *)malloc(totSize); + + param = config_find(root, PARAM_TAG); + value = config_find(root, VALUE_TAG); + if (param == NULL && value == NULL) { + // try to parse simple parameter form {int int} + param = root->first_child; + if (param != NULL) { + // Note: that a pair of random strings is read as 0 0 + int *ptr = (int *)fx_param->data; + int *ptr2 = (int *)((char *)param + sizeof(effect_param_t)); + ALOGW("loadEffectParameter() ptr %p ptr2 %p", ptr, ptr2); + *ptr++ = atoi(param->name); + *ptr = atoi(param->value); + fx_param->psize = sizeof(int); + fx_param->vsize = sizeof(int); + return fx_param; + } + } + if (param == NULL || value == NULL) { + ALOGW("loadEffectParameter() invalid parameter description %s", root->name); + goto error; + } + + fx_param->psize = 0; + param = param->first_child; + while (param) { + ALOGV("loadEffectParameter() reading param of type %s", param->name); + size_t size = readParamValue(param, (char *)fx_param, &curSize, &totSize); + if (size == 0) { + goto error; + } + fx_param->psize += size; + param = param->next; + } + + // align start of value field on 32 bit boundary + curSize = ((curSize - 1 ) / sizeof(int) + 1) * sizeof(int); + + fx_param->vsize = 0; + value = value->first_child; + while (value) { + ALOGV("loadEffectParameter() reading value of type %s", value->name); + size_t size = readParamValue(value, (char *)fx_param, &curSize, &totSize); + if (size == 0) { + goto error; + } + fx_param->vsize += size; + value = value->next; + } + + return fx_param; + +error: + delete fx_param; + return NULL; +} + +void AudioPolicyEffects::loadEffectParameters(cnode *root, Vector & params) +{ + cnode *node = root->first_child; + while (node) { + ALOGV("loadEffectParameters() loading param %s", node->name); + effect_param_t *param = loadEffectParameter(node); + if (param == NULL) { + node = node->next; + continue; + } + params.add(param); + node = node->next; + } +} + + +AudioPolicyEffects::EffectDescVector *AudioPolicyEffects::loadEffectConfig( + cnode *root, + const Vector & effects) +{ + cnode *node = root->first_child; + if (node == NULL) { + ALOGW("loadInputSource() empty element %s", root->name); + return NULL; + } + EffectDescVector *desc = new EffectDescVector(); + while (node) { + size_t i; + for (i = 0; i < effects.size(); i++) { + if (strncmp(effects[i]->mName, node->name, EFFECT_STRING_LEN_MAX) == 0) { + ALOGV("loadEffectConfig() found effect %s in list", node->name); + break; + } + } + if (i == effects.size()) { + ALOGV("loadEffectConfig() effect %s not in list", node->name); + node = node->next; + continue; + } + EffectDesc *effect = new EffectDesc(*effects[i]); // deep copy + loadEffectParameters(node, effect->mParams); + ALOGV("loadEffectConfig() adding effect %s uuid %08x", + effect->mName, effect->mUuid.timeLow); + desc->mEffects.add(effect); + node = node->next; + } + if (desc->mEffects.size() == 0) { + ALOGW("loadEffectConfig() no valid effects found in config %s", root->name); + delete desc; + return NULL; + } + return desc; +} + +status_t AudioPolicyEffects::loadInputEffectConfigurations(cnode *root, + const Vector & effects) +{ + cnode *node = config_find(root, PREPROCESSING_TAG); + if (node == NULL) { + return -ENOENT; + } + node = node->first_child; + while (node) { + audio_source_t source = inputSourceNameToEnum(node->name); + if (source == AUDIO_SOURCE_CNT) { + ALOGW("loadInputSources() invalid input source %s", node->name); + node = node->next; + continue; + } + ALOGV("loadInputSources() loading input source %s", node->name); + EffectDescVector *desc = loadEffectConfig(node, effects); + if (desc == NULL) { + node = node->next; + continue; + } + mInputSources.add(source, desc); + node = node->next; + } + return NO_ERROR; +} + +status_t AudioPolicyEffects::loadStreamEffectConfigurations(cnode *root, + const Vector & effects) +{ + cnode *node = config_find(root, OUTPUT_SESSION_PROCESSING_TAG); + if (node == NULL) { + return -ENOENT; + } + node = node->first_child; + while (node) { + audio_stream_type_t stream = streamNameToEnum(node->name); + if (stream == AUDIO_STREAM_PUBLIC_CNT) { + ALOGW("loadStreamEffectConfigurations() invalid output stream %s", node->name); + node = node->next; + continue; + } + ALOGV("loadStreamEffectConfigurations() loading output stream %s", node->name); + EffectDescVector *desc = loadEffectConfig(node, effects); + if (desc == NULL) { + node = node->next; + continue; + } + mOutputStreams.add(stream, desc); + node = node->next; + } + return NO_ERROR; +} + +AudioPolicyEffects::EffectDesc *AudioPolicyEffects::loadEffect(cnode *root) +{ + cnode *node = config_find(root, UUID_TAG); + if (node == NULL) { + return NULL; + } + effect_uuid_t uuid; + if (AudioEffect::stringToGuid(node->value, &uuid) != NO_ERROR) { + ALOGW("loadEffect() invalid uuid %s", node->value); + return NULL; + } + return new EffectDesc(root->name, uuid); +} + +status_t AudioPolicyEffects::loadEffects(cnode *root, Vector & effects) +{ + cnode *node = config_find(root, EFFECTS_TAG); + if (node == NULL) { + return -ENOENT; + } + node = node->first_child; + while (node) { + ALOGV("loadEffects() loading effect %s", node->name); + EffectDesc *effect = loadEffect(node); + if (effect == NULL) { + node = node->next; + continue; + } + effects.add(effect); + node = node->next; + } + return NO_ERROR; +} + +status_t AudioPolicyEffects::loadAudioEffectConfig(const char *path) +{ + cnode *root; + char *data; + + data = (char *)load_file(path, NULL); + if (data == NULL) { + return -ENODEV; + } + root = config_node("", ""); + config_load(root, data); + + Vector effects; + loadEffects(root, effects); + loadInputEffectConfigurations(root, effects); + loadStreamEffectConfigurations(root, effects); + + for (size_t i = 0; i < effects.size(); i++) { + delete effects[i]; + } + + config_free(root); + free(root); + free(data); + + return NO_ERROR; +} + + +}; // namespace android diff --git a/services/audiopolicy/service/AudioPolicyEffects.h b/services/audiopolicy/service/AudioPolicyEffects.h new file mode 100644 index 0000000..3dec437 --- /dev/null +++ b/services/audiopolicy/service/AudioPolicyEffects.h @@ -0,0 +1,196 @@ +/* + * Copyright (C) 2014 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef ANDROID_AUDIOPOLICYEFFECTS_H +#define ANDROID_AUDIOPOLICYEFFECTS_H + +#include +#include +#include +#include +#include +#include +#include +#include +#include + +namespace android { + +// ---------------------------------------------------------------------------- + +// AudioPolicyEffects class +// This class will manage all effects attached to input and output streams in +// AudioPolicyService as configured in audio_effects.conf. +class AudioPolicyEffects : public RefBase +{ + +public: + + // The constructor will parse audio_effects.conf + // First it will look whether vendor specific file exists, + // otherwise it will parse the system default file. + AudioPolicyEffects(); + virtual ~AudioPolicyEffects(); + + // NOTE: methods on AudioPolicyEffects should never be called with the AudioPolicyService + // main mutex (mLock) held as they will indirectly call back into AudioPolicyService when + // managing audio effects. + + // Return a list of effect descriptors for default input effects + // associated with audioSession + status_t queryDefaultInputEffects(int audioSession, + effect_descriptor_t *descriptors, + uint32_t *count); + + // Add all input effects associated with this input + // Effects are attached depending on the audio_source_t + status_t addInputEffects(audio_io_handle_t input, + audio_source_t inputSource, + int audioSession); + + // Add all input effects associated to this input + status_t releaseInputEffects(audio_io_handle_t input); + + + // Return a list of effect descriptors for default output effects + // associated with audioSession + status_t queryDefaultOutputSessionEffects(int audioSession, + effect_descriptor_t *descriptors, + uint32_t *count); + + // Add all output effects associated to this output + // Effects are attached depending on the audio_stream_type_t + status_t addOutputSessionEffects(audio_io_handle_t output, + audio_stream_type_t stream, + int audioSession); + + // release all output effects associated with this output stream and audiosession + status_t releaseOutputSessionEffects(audio_io_handle_t output, + audio_stream_type_t stream, + int audioSession); + +private: + + // class to store the description of an effects and its parameters + // as defined in audio_effects.conf + class EffectDesc { + public: + EffectDesc(const char *name, const effect_uuid_t& uuid) : + mName(strdup(name)), + mUuid(uuid) { } + EffectDesc(const EffectDesc& orig) : + mName(strdup(orig.mName)), + mUuid(orig.mUuid) { + // deep copy mParams + for (size_t k = 0; k < orig.mParams.size(); k++) { + effect_param_t *origParam = orig.mParams[k]; + // psize and vsize are rounded up to an int boundary for allocation + size_t origSize = sizeof(effect_param_t) + + ((origParam->psize + 3) & ~3) + + ((origParam->vsize + 3) & ~3); + effect_param_t *dupParam = (effect_param_t *) malloc(origSize); + memcpy(dupParam, origParam, origSize); + // This works because the param buffer allocation is also done by + // multiples of 4 bytes originally. In theory we should memcpy only + // the actual param size, that is without rounding vsize. + mParams.add(dupParam); + } + } + /*virtual*/ ~EffectDesc() { + free(mName); + for (size_t k = 0; k < mParams.size(); k++) { + free(mParams[k]); + } + } + char *mName; + effect_uuid_t mUuid; + Vector mParams; + }; + + // class to store voctor of EffectDesc + class EffectDescVector { + public: + EffectDescVector() {} + /*virtual*/ ~EffectDescVector() { + for (size_t j = 0; j < mEffects.size(); j++) { + delete mEffects[j]; + } + } + Vector mEffects; + }; + + // class to store voctor of AudioEffects + class EffectVector { + public: + EffectVector(int session) : mSessionId(session), mRefCount(0) {} + /*virtual*/ ~EffectVector() {} + + // Enable or disable all effects in effect vector + void setProcessorEnabled(bool enabled); + + const int mSessionId; + // AudioPolicyManager keeps mLock, no need for lock on reference count here + int mRefCount; + Vector< sp >mEffects; + }; + + + static const char * const kInputSourceNames[AUDIO_SOURCE_CNT -1]; + static audio_source_t inputSourceNameToEnum(const char *name); + + static const char *kStreamNames[AUDIO_STREAM_PUBLIC_CNT+1]; //+1 required as streams start from -1 + audio_stream_type_t streamNameToEnum(const char *name); + + // Parse audio_effects.conf + status_t loadAudioEffectConfig(const char *path); + + // Load all effects descriptors in configuration file + status_t loadEffects(cnode *root, Vector & effects); + EffectDesc *loadEffect(cnode *root); + + // Load all automatic effect configurations + status_t loadInputEffectConfigurations(cnode *root, const Vector & effects); + status_t loadStreamEffectConfigurations(cnode *root, const Vector & effects); + EffectDescVector *loadEffectConfig(cnode *root, const Vector & effects); + + // Load all automatic effect parameters + void loadEffectParameters(cnode *root, Vector & params); + effect_param_t *loadEffectParameter(cnode *root); + size_t readParamValue(cnode *node, + char *param, + size_t *curSize, + size_t *totSize); + size_t growParamSize(char *param, + size_t size, + size_t *curSize, + size_t *totSize); + + // protects access to mInputSources, mInputs, mOutputStreams, mOutputSessions + Mutex mLock; + // Automatic input effects are configured per audio_source_t + KeyedVector< audio_source_t, EffectDescVector* > mInputSources; + // Automatic input effects are unique for audio_io_handle_t + KeyedVector< audio_io_handle_t, EffectVector* > mInputs; + + // Automatic output effects are organized per audio_stream_type_t + KeyedVector< audio_stream_type_t, EffectDescVector* > mOutputStreams; + // Automatic output effects are unique for audiosession ID + KeyedVector< int32_t, EffectVector* > mOutputSessions; +}; + +}; // namespace android + +#endif // ANDROID_AUDIOPOLICYEFFECTS_H diff --git a/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp new file mode 100644 index 0000000..e9ff838 --- /dev/null +++ b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp @@ -0,0 +1,664 @@ +/* + * Copyright (C) 2009 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#define LOG_TAG "AudioPolicyIntefaceImpl" +//#define LOG_NDEBUG 0 + +#include +#include "AudioPolicyService.h" +#include "ServiceUtilities.h" + +namespace android { + + +// ---------------------------------------------------------------------------- + +status_t AudioPolicyService::setDeviceConnectionState(audio_devices_t device, + audio_policy_dev_state_t state, + const char *device_address, + const char *device_name) +{ + if (mAudioPolicyManager == NULL) { + return NO_INIT; + } + if (!settingsAllowed()) { + return PERMISSION_DENIED; + } + if (!audio_is_output_device(device) && !audio_is_input_device(device)) { + return BAD_VALUE; + } + if (state != AUDIO_POLICY_DEVICE_STATE_AVAILABLE && + state != AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) { + return BAD_VALUE; + } + + ALOGV("setDeviceConnectionState()"); + Mutex::Autolock _l(mLock); + return mAudioPolicyManager->setDeviceConnectionState(device, state, + device_address, device_name); +} + +audio_policy_dev_state_t AudioPolicyService::getDeviceConnectionState( + audio_devices_t device, + const char *device_address) +{ + if (mAudioPolicyManager == NULL) { + return AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE; + } + return mAudioPolicyManager->getDeviceConnectionState(device, + device_address); +} + +status_t AudioPolicyService::setPhoneState(audio_mode_t state) +{ + if (mAudioPolicyManager == NULL) { + return NO_INIT; + } + if (!settingsAllowed()) { + return PERMISSION_DENIED; + } + if (uint32_t(state) >= AUDIO_MODE_CNT) { + return BAD_VALUE; + } + + ALOGV("setPhoneState()"); + + // TODO: check if it is more appropriate to do it in platform specific policy manager + AudioSystem::setMode(state); + + Mutex::Autolock _l(mLock); + mAudioPolicyManager->setPhoneState(state); + mPhoneState = state; + return NO_ERROR; +} + +audio_mode_t AudioPolicyService::getPhoneState() +{ + Mutex::Autolock _l(mLock); + return mPhoneState; +} + +status_t AudioPolicyService::setForceUse(audio_policy_force_use_t usage, + audio_policy_forced_cfg_t config) +{ + if (mAudioPolicyManager == NULL) { + return NO_INIT; + } + if (!settingsAllowed()) { + return PERMISSION_DENIED; + } + if (usage < 0 || usage >= AUDIO_POLICY_FORCE_USE_CNT) { + return BAD_VALUE; + } + if (config < 0 || config >= AUDIO_POLICY_FORCE_CFG_CNT) { + return BAD_VALUE; + } + ALOGV("setForceUse()"); + Mutex::Autolock _l(mLock); + mAudioPolicyManager->setForceUse(usage, config); + return NO_ERROR; +} + +audio_policy_forced_cfg_t AudioPolicyService::getForceUse(audio_policy_force_use_t usage) +{ + if (mAudioPolicyManager == NULL) { + return AUDIO_POLICY_FORCE_NONE; + } + if (usage < 0 || usage >= AUDIO_POLICY_FORCE_USE_CNT) { + return AUDIO_POLICY_FORCE_NONE; + } + return mAudioPolicyManager->getForceUse(usage); +} + +audio_io_handle_t AudioPolicyService::getOutput(audio_stream_type_t stream, + uint32_t samplingRate, + audio_format_t format, + audio_channel_mask_t channelMask, + audio_output_flags_t flags, + const audio_offload_info_t *offloadInfo) +{ + if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT) { + return AUDIO_IO_HANDLE_NONE; + } + if (mAudioPolicyManager == NULL) { + return AUDIO_IO_HANDLE_NONE; + } + ALOGV("getOutput()"); + Mutex::Autolock _l(mLock); + return mAudioPolicyManager->getOutput(stream, samplingRate, + format, channelMask, flags, offloadInfo); +} + +status_t AudioPolicyService::getOutputForAttr(const audio_attributes_t *attr, + audio_io_handle_t *output, + audio_session_t session, + audio_stream_type_t *stream, + uint32_t samplingRate, + audio_format_t format, + audio_channel_mask_t channelMask, + audio_output_flags_t flags, + const audio_offload_info_t *offloadInfo) +{ + if (mAudioPolicyManager == NULL) { + return NO_INIT; + } + ALOGV("getOutput()"); + Mutex::Autolock _l(mLock); + return mAudioPolicyManager->getOutputForAttr(attr, output, session, stream, samplingRate, + format, channelMask, flags, offloadInfo); +} + +status_t AudioPolicyService::startOutput(audio_io_handle_t output, + audio_stream_type_t stream, + audio_session_t session) +{ + if (uint32_t(stream) >= AUDIO_STREAM_CNT) { + return BAD_VALUE; + } + if (mAudioPolicyManager == NULL) { + return NO_INIT; + } + ALOGV("startOutput()"); + spaudioPolicyEffects; + { + Mutex::Autolock _l(mLock); + audioPolicyEffects = mAudioPolicyEffects; + } + if (audioPolicyEffects != 0) { + // create audio processors according to stream + status_t status = audioPolicyEffects->addOutputSessionEffects(output, stream, session); + if (status != NO_ERROR && status != ALREADY_EXISTS) { + ALOGW("Failed to add effects on session %d", session); + } + } + Mutex::Autolock _l(mLock); + return mAudioPolicyManager->startOutput(output, stream, session); +} + +status_t AudioPolicyService::stopOutput(audio_io_handle_t output, + audio_stream_type_t stream, + audio_session_t session) +{ + if (uint32_t(stream) >= AUDIO_STREAM_CNT) { + return BAD_VALUE; + } + if (mAudioPolicyManager == NULL) { + return NO_INIT; + } + ALOGV("stopOutput()"); + mOutputCommandThread->stopOutputCommand(output, stream, session); + return NO_ERROR; +} + +status_t AudioPolicyService::doStopOutput(audio_io_handle_t output, + audio_stream_type_t stream, + audio_session_t session) +{ + ALOGV("doStopOutput from tid %d", gettid()); + spaudioPolicyEffects; + { + Mutex::Autolock _l(mLock); + audioPolicyEffects = mAudioPolicyEffects; + } + if (audioPolicyEffects != 0) { + // release audio processors from the stream + status_t status = audioPolicyEffects->releaseOutputSessionEffects(output, stream, session); + if (status != NO_ERROR && status != ALREADY_EXISTS) { + ALOGW("Failed to release effects on session %d", session); + } + } + Mutex::Autolock _l(mLock); + return mAudioPolicyManager->stopOutput(output, stream, session); +} + +void AudioPolicyService::releaseOutput(audio_io_handle_t output, + audio_stream_type_t stream, + audio_session_t session) +{ + if (mAudioPolicyManager == NULL) { + return; + } + ALOGV("releaseOutput()"); + mOutputCommandThread->releaseOutputCommand(output, stream, session); +} + +void AudioPolicyService::doReleaseOutput(audio_io_handle_t output, + audio_stream_type_t stream, + audio_session_t session) +{ + ALOGV("doReleaseOutput from tid %d", gettid()); + Mutex::Autolock _l(mLock); + mAudioPolicyManager->releaseOutput(output, stream, session); +} + +status_t AudioPolicyService::getInputForAttr(const audio_attributes_t *attr, + audio_io_handle_t *input, + audio_session_t session, + uint32_t samplingRate, + audio_format_t format, + audio_channel_mask_t channelMask, + audio_input_flags_t flags) +{ + if (mAudioPolicyManager == NULL) { + return NO_INIT; + } + // already checked by client, but double-check in case the client wrapper is bypassed + if (attr->source >= AUDIO_SOURCE_CNT && attr->source != AUDIO_SOURCE_HOTWORD && + attr->source != AUDIO_SOURCE_FM_TUNER) { + return BAD_VALUE; + } + + if (((attr->source == AUDIO_SOURCE_HOTWORD) && !captureHotwordAllowed()) || + ((attr->source == AUDIO_SOURCE_FM_TUNER) && !captureFmTunerAllowed())) { + return BAD_VALUE; + } + spaudioPolicyEffects; + status_t status; + AudioPolicyInterface::input_type_t inputType; + { + Mutex::Autolock _l(mLock); + // the audio_in_acoustics_t parameter is ignored by get_input() + status = mAudioPolicyManager->getInputForAttr(attr, input, session, + samplingRate, format, channelMask, + flags, &inputType); + audioPolicyEffects = mAudioPolicyEffects; + + if (status == NO_ERROR) { + // enforce permission (if any) required for each type of input + switch (inputType) { + case AudioPolicyInterface::API_INPUT_LEGACY: + break; + case AudioPolicyInterface::API_INPUT_MIX_CAPTURE: + if (!captureAudioOutputAllowed()) { + ALOGE("getInputForAttr() permission denied: capture not allowed"); + status = PERMISSION_DENIED; + } + break; + case AudioPolicyInterface::API_INPUT_MIX_EXT_POLICY_REROUTE: + if (!modifyAudioRoutingAllowed()) { + ALOGE("getInputForAttr() permission denied: modify audio routing not allowed"); + status = PERMISSION_DENIED; + } + break; + case AudioPolicyInterface::API_INPUT_INVALID: + default: + LOG_ALWAYS_FATAL("getInputForAttr() encountered an invalid input type %d", + (int)inputType); + } + } + + if (status != NO_ERROR) { + if (status == PERMISSION_DENIED) { + mAudioPolicyManager->releaseInput(*input, session); + } + return status; + } + } + + if (audioPolicyEffects != 0) { + // create audio pre processors according to input source + status_t status = audioPolicyEffects->addInputEffects(*input, attr->source, session); + if (status != NO_ERROR && status != ALREADY_EXISTS) { + ALOGW("Failed to add effects on input %d", *input); + } + } + return NO_ERROR; +} + +status_t AudioPolicyService::startInput(audio_io_handle_t input, + audio_session_t session) +{ + if (mAudioPolicyManager == NULL) { + return NO_INIT; + } + Mutex::Autolock _l(mLock); + + return mAudioPolicyManager->startInput(input, session); +} + +status_t AudioPolicyService::stopInput(audio_io_handle_t input, + audio_session_t session) +{ + if (mAudioPolicyManager == NULL) { + return NO_INIT; + } + Mutex::Autolock _l(mLock); + + return mAudioPolicyManager->stopInput(input, session); +} + +void AudioPolicyService::releaseInput(audio_io_handle_t input, + audio_session_t session) +{ + if (mAudioPolicyManager == NULL) { + return; + } + spaudioPolicyEffects; + { + Mutex::Autolock _l(mLock); + mAudioPolicyManager->releaseInput(input, session); + audioPolicyEffects = mAudioPolicyEffects; + } + if (audioPolicyEffects != 0) { + // release audio processors from the input + status_t status = audioPolicyEffects->releaseInputEffects(input); + if(status != NO_ERROR) { + ALOGW("Failed to release effects on input %d", input); + } + } +} + +status_t AudioPolicyService::initStreamVolume(audio_stream_type_t stream, + int indexMin, + int indexMax) +{ + if (mAudioPolicyManager == NULL) { + return NO_INIT; + } + if (!settingsAllowed()) { + return PERMISSION_DENIED; + } + if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT) { + return BAD_VALUE; + } + Mutex::Autolock _l(mLock); + mAudioPolicyManager->initStreamVolume(stream, indexMin, indexMax); + return NO_ERROR; +} + +status_t AudioPolicyService::setStreamVolumeIndex(audio_stream_type_t stream, + int index, + audio_devices_t device) +{ + if (mAudioPolicyManager == NULL) { + return NO_INIT; + } + if (!settingsAllowed()) { + return PERMISSION_DENIED; + } + if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT) { + return BAD_VALUE; + } + Mutex::Autolock _l(mLock); + return mAudioPolicyManager->setStreamVolumeIndex(stream, + index, + device); +} + +status_t AudioPolicyService::getStreamVolumeIndex(audio_stream_type_t stream, + int *index, + audio_devices_t device) +{ + if (mAudioPolicyManager == NULL) { + return NO_INIT; + } + if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT) { + return BAD_VALUE; + } + Mutex::Autolock _l(mLock); + return mAudioPolicyManager->getStreamVolumeIndex(stream, + index, + device); +} + +uint32_t AudioPolicyService::getStrategyForStream(audio_stream_type_t stream) +{ + if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT) { + return 0; + } + if (mAudioPolicyManager == NULL) { + return 0; + } + return mAudioPolicyManager->getStrategyForStream(stream); +} + +//audio policy: use audio_device_t appropriately + +audio_devices_t AudioPolicyService::getDevicesForStream(audio_stream_type_t stream) +{ + if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT) { + return AUDIO_DEVICE_NONE; + } + if (mAudioPolicyManager == NULL) { + return AUDIO_DEVICE_NONE; + } + return mAudioPolicyManager->getDevicesForStream(stream); +} + +audio_io_handle_t AudioPolicyService::getOutputForEffect(const effect_descriptor_t *desc) +{ + // FIXME change return type to status_t, and return NO_INIT here + if (mAudioPolicyManager == NULL) { + return 0; + } + Mutex::Autolock _l(mLock); + return mAudioPolicyManager->getOutputForEffect(desc); +} + +status_t AudioPolicyService::registerEffect(const effect_descriptor_t *desc, + audio_io_handle_t io, + uint32_t strategy, + int session, + int id) +{ + if (mAudioPolicyManager == NULL) { + return NO_INIT; + } + return mAudioPolicyManager->registerEffect(desc, io, strategy, session, id); +} + +status_t AudioPolicyService::unregisterEffect(int id) +{ + if (mAudioPolicyManager == NULL) { + return NO_INIT; + } + return mAudioPolicyManager->unregisterEffect(id); +} + +status_t AudioPolicyService::setEffectEnabled(int id, bool enabled) +{ + if (mAudioPolicyManager == NULL) { + return NO_INIT; + } + return mAudioPolicyManager->setEffectEnabled(id, enabled); +} + +bool AudioPolicyService::isStreamActive(audio_stream_type_t stream, uint32_t inPastMs) const +{ + if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT) { + return false; + } + if (mAudioPolicyManager == NULL) { + return false; + } + Mutex::Autolock _l(mLock); + return mAudioPolicyManager->isStreamActive(stream, inPastMs); +} + +bool AudioPolicyService::isStreamActiveRemotely(audio_stream_type_t stream, uint32_t inPastMs) const +{ + if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT) { + return false; + } + if (mAudioPolicyManager == NULL) { + return false; + } + Mutex::Autolock _l(mLock); + return mAudioPolicyManager->isStreamActiveRemotely(stream, inPastMs); +} + +bool AudioPolicyService::isSourceActive(audio_source_t source) const +{ + if (mAudioPolicyManager == NULL) { + return false; + } + Mutex::Autolock _l(mLock); + return mAudioPolicyManager->isSourceActive(source); +} + +status_t AudioPolicyService::queryDefaultPreProcessing(int audioSession, + effect_descriptor_t *descriptors, + uint32_t *count) +{ + if (mAudioPolicyManager == NULL) { + *count = 0; + return NO_INIT; + } + spaudioPolicyEffects; + { + Mutex::Autolock _l(mLock); + audioPolicyEffects = mAudioPolicyEffects; + } + if (audioPolicyEffects == 0) { + *count = 0; + return NO_INIT; + } + return audioPolicyEffects->queryDefaultInputEffects(audioSession, descriptors, count); +} + +bool AudioPolicyService::isOffloadSupported(const audio_offload_info_t& info) +{ + if (mAudioPolicyManager == NULL) { + ALOGV("mAudioPolicyManager == NULL"); + return false; + } + + return mAudioPolicyManager->isOffloadSupported(info); +} + +status_t AudioPolicyService::listAudioPorts(audio_port_role_t role, + audio_port_type_t type, + unsigned int *num_ports, + struct audio_port *ports, + unsigned int *generation) +{ + Mutex::Autolock _l(mLock); + if(!modifyAudioRoutingAllowed()) { + return PERMISSION_DENIED; + } + if (mAudioPolicyManager == NULL) { + return NO_INIT; + } + + return mAudioPolicyManager->listAudioPorts(role, type, num_ports, ports, generation); +} + +status_t AudioPolicyService::getAudioPort(struct audio_port *port) +{ + Mutex::Autolock _l(mLock); + if(!modifyAudioRoutingAllowed()) { + return PERMISSION_DENIED; + } + if (mAudioPolicyManager == NULL) { + return NO_INIT; + } + + return mAudioPolicyManager->getAudioPort(port); +} + +status_t AudioPolicyService::createAudioPatch(const struct audio_patch *patch, + audio_patch_handle_t *handle) +{ + Mutex::Autolock _l(mLock); + if(!modifyAudioRoutingAllowed()) { + return PERMISSION_DENIED; + } + if (mAudioPolicyManager == NULL) { + return NO_INIT; + } + return mAudioPolicyManager->createAudioPatch(patch, handle, + IPCThreadState::self()->getCallingUid()); +} + +status_t AudioPolicyService::releaseAudioPatch(audio_patch_handle_t handle) +{ + Mutex::Autolock _l(mLock); + if(!modifyAudioRoutingAllowed()) { + return PERMISSION_DENIED; + } + if (mAudioPolicyManager == NULL) { + return NO_INIT; + } + + return mAudioPolicyManager->releaseAudioPatch(handle, + IPCThreadState::self()->getCallingUid()); +} + +status_t AudioPolicyService::listAudioPatches(unsigned int *num_patches, + struct audio_patch *patches, + unsigned int *generation) +{ + Mutex::Autolock _l(mLock); + if(!modifyAudioRoutingAllowed()) { + return PERMISSION_DENIED; + } + if (mAudioPolicyManager == NULL) { + return NO_INIT; + } + + return mAudioPolicyManager->listAudioPatches(num_patches, patches, generation); +} + +status_t AudioPolicyService::setAudioPortConfig(const struct audio_port_config *config) +{ + Mutex::Autolock _l(mLock); + if(!modifyAudioRoutingAllowed()) { + return PERMISSION_DENIED; + } + if (mAudioPolicyManager == NULL) { + return NO_INIT; + } + + return mAudioPolicyManager->setAudioPortConfig(config); +} + +status_t AudioPolicyService::acquireSoundTriggerSession(audio_session_t *session, + audio_io_handle_t *ioHandle, + audio_devices_t *device) +{ + if (mAudioPolicyManager == NULL) { + return NO_INIT; + } + + return mAudioPolicyManager->acquireSoundTriggerSession(session, ioHandle, device); +} + +status_t AudioPolicyService::releaseSoundTriggerSession(audio_session_t session) +{ + if (mAudioPolicyManager == NULL) { + return NO_INIT; + } + + return mAudioPolicyManager->releaseSoundTriggerSession(session); +} + +status_t AudioPolicyService::registerPolicyMixes(Vector mixes, bool registration) +{ + Mutex::Autolock _l(mLock); + if(!modifyAudioRoutingAllowed()) { + return PERMISSION_DENIED; + } + if (mAudioPolicyManager == NULL) { + return NO_INIT; + } + if (registration) { + return mAudioPolicyManager->registerPolicyMixes(mixes); + } else { + return mAudioPolicyManager->unregisterPolicyMixes(mixes); + } +} + +}; // namespace android diff --git a/services/audiopolicy/service/AudioPolicyInterfaceImplLegacy.cpp b/services/audiopolicy/service/AudioPolicyInterfaceImplLegacy.cpp new file mode 100644 index 0000000..5a91192 --- /dev/null +++ b/services/audiopolicy/service/AudioPolicyInterfaceImplLegacy.cpp @@ -0,0 +1,607 @@ +/* + * Copyright (C) 2009 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#define LOG_TAG "AudioPolicyService" +//#define LOG_NDEBUG 0 + +#include +#include "AudioPolicyService.h" +#include "ServiceUtilities.h" + +#include +#include +#include +#include + +namespace android { + + +// ---------------------------------------------------------------------------- + +status_t AudioPolicyService::setDeviceConnectionState(audio_devices_t device, + audio_policy_dev_state_t state, + const char *device_address, + const char *device_name __unused) +{ + if (mpAudioPolicy == NULL) { + return NO_INIT; + } + if (!settingsAllowed()) { + return PERMISSION_DENIED; + } + if (!audio_is_output_device(device) && !audio_is_input_device(device)) { + return BAD_VALUE; + } + if (state != AUDIO_POLICY_DEVICE_STATE_AVAILABLE && + state != AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) { + return BAD_VALUE; + } + + ALOGV("setDeviceConnectionState()"); + Mutex::Autolock _l(mLock); + return mpAudioPolicy->set_device_connection_state(mpAudioPolicy, device, + state, device_address); +} + +audio_policy_dev_state_t AudioPolicyService::getDeviceConnectionState( + audio_devices_t device, + const char *device_address) +{ + if (mpAudioPolicy == NULL) { + return AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE; + } + return mpAudioPolicy->get_device_connection_state(mpAudioPolicy, device, + device_address); +} + +status_t AudioPolicyService::setPhoneState(audio_mode_t state) +{ + if (mpAudioPolicy == NULL) { + return NO_INIT; + } + if (!settingsAllowed()) { + return PERMISSION_DENIED; + } + if (uint32_t(state) >= AUDIO_MODE_CNT) { + return BAD_VALUE; + } + + ALOGV("setPhoneState()"); + + // TODO: check if it is more appropriate to do it in platform specific policy manager + AudioSystem::setMode(state); + + Mutex::Autolock _l(mLock); + mpAudioPolicy->set_phone_state(mpAudioPolicy, state); + mPhoneState = state; + return NO_ERROR; +} + +audio_mode_t AudioPolicyService::getPhoneState() +{ + Mutex::Autolock _l(mLock); + return mPhoneState; +} + +status_t AudioPolicyService::setForceUse(audio_policy_force_use_t usage, + audio_policy_forced_cfg_t config) +{ + if (mpAudioPolicy == NULL) { + return NO_INIT; + } + if (!settingsAllowed()) { + return PERMISSION_DENIED; + } + if (usage < 0 || usage >= AUDIO_POLICY_FORCE_USE_CNT) { + return BAD_VALUE; + } + if (config < 0 || config >= AUDIO_POLICY_FORCE_CFG_CNT) { + return BAD_VALUE; + } + ALOGV("setForceUse()"); + Mutex::Autolock _l(mLock); + mpAudioPolicy->set_force_use(mpAudioPolicy, usage, config); + return NO_ERROR; +} + +audio_policy_forced_cfg_t AudioPolicyService::getForceUse(audio_policy_force_use_t usage) +{ + if (mpAudioPolicy == NULL) { + return AUDIO_POLICY_FORCE_NONE; + } + if (usage < 0 || usage >= AUDIO_POLICY_FORCE_USE_CNT) { + return AUDIO_POLICY_FORCE_NONE; + } + return mpAudioPolicy->get_force_use(mpAudioPolicy, usage); +} + +audio_io_handle_t AudioPolicyService::getOutput(audio_stream_type_t stream, + uint32_t samplingRate, + audio_format_t format, + audio_channel_mask_t channelMask, + audio_output_flags_t flags, + const audio_offload_info_t *offloadInfo) +{ + if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT) { + return AUDIO_IO_HANDLE_NONE; + } + if (mpAudioPolicy == NULL) { + return AUDIO_IO_HANDLE_NONE; + } + ALOGV("getOutput()"); + Mutex::Autolock _l(mLock); + return mpAudioPolicy->get_output(mpAudioPolicy, stream, samplingRate, + format, channelMask, flags, offloadInfo); +} + +status_t AudioPolicyService::startOutput(audio_io_handle_t output, + audio_stream_type_t stream, + audio_session_t session) +{ + if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT) { + return BAD_VALUE; + } + if (mpAudioPolicy == NULL) { + return NO_INIT; + } + ALOGV("startOutput()"); + // create audio processors according to stream + spaudioPolicyEffects; + { + Mutex::Autolock _l(mLock); + audioPolicyEffects = mAudioPolicyEffects; + } + if (audioPolicyEffects != 0) { + status_t status = audioPolicyEffects->addOutputSessionEffects(output, stream, session); + if (status != NO_ERROR && status != ALREADY_EXISTS) { + ALOGW("Failed to add effects on session %d", session); + } + } + + Mutex::Autolock _l(mLock); + return mpAudioPolicy->start_output(mpAudioPolicy, output, stream, session); +} + +status_t AudioPolicyService::stopOutput(audio_io_handle_t output, + audio_stream_type_t stream, + audio_session_t session) +{ + if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT) { + return BAD_VALUE; + } + if (mpAudioPolicy == NULL) { + return NO_INIT; + } + ALOGV("stopOutput()"); + mOutputCommandThread->stopOutputCommand(output, stream, session); + return NO_ERROR; +} + +status_t AudioPolicyService::doStopOutput(audio_io_handle_t output, + audio_stream_type_t stream, + audio_session_t session) +{ + ALOGV("doStopOutput from tid %d", gettid()); + // release audio processors from the stream + spaudioPolicyEffects; + { + Mutex::Autolock _l(mLock); + audioPolicyEffects = mAudioPolicyEffects; + } + if (audioPolicyEffects != 0) { + status_t status = audioPolicyEffects->releaseOutputSessionEffects(output, stream, session); + if (status != NO_ERROR && status != ALREADY_EXISTS) { + ALOGW("Failed to release effects on session %d", session); + } + } + Mutex::Autolock _l(mLock); + return mpAudioPolicy->stop_output(mpAudioPolicy, output, stream, session); +} + +void AudioPolicyService::releaseOutput(audio_io_handle_t output, + audio_stream_type_t stream, + audio_session_t session) +{ + if (mpAudioPolicy == NULL) { + return; + } + ALOGV("releaseOutput()"); + mOutputCommandThread->releaseOutputCommand(output, stream, session); +} + +void AudioPolicyService::doReleaseOutput(audio_io_handle_t output, + audio_stream_type_t stream __unused, + audio_session_t session __unused) +{ + ALOGV("doReleaseOutput from tid %d", gettid()); + Mutex::Autolock _l(mLock); + mpAudioPolicy->release_output(mpAudioPolicy, output); +} + +status_t AudioPolicyService::getInputForAttr(const audio_attributes_t *attr, + audio_io_handle_t *input, + audio_session_t session, + uint32_t samplingRate, + audio_format_t format, + audio_channel_mask_t channelMask, + audio_input_flags_t flags __unused) +{ + if (mpAudioPolicy == NULL) { + return NO_INIT; + } + + audio_source_t inputSource = attr->source; + + // already checked by client, but double-check in case the client wrapper is bypassed + if (inputSource >= AUDIO_SOURCE_CNT && inputSource != AUDIO_SOURCE_HOTWORD && + inputSource != AUDIO_SOURCE_FM_TUNER) { + return BAD_VALUE; + } + + if (inputSource == AUDIO_SOURCE_DEFAULT) { + inputSource = AUDIO_SOURCE_MIC; + } + + if (((inputSource == AUDIO_SOURCE_HOTWORD) && !captureHotwordAllowed()) || + ((inputSource == AUDIO_SOURCE_FM_TUNER) && !captureFmTunerAllowed())) { + return BAD_VALUE; + } + + spaudioPolicyEffects; + { + Mutex::Autolock _l(mLock); + // the audio_in_acoustics_t parameter is ignored by get_input() + *input = mpAudioPolicy->get_input(mpAudioPolicy, inputSource, samplingRate, + format, channelMask, (audio_in_acoustics_t) 0); + audioPolicyEffects = mAudioPolicyEffects; + } + if (*input == AUDIO_IO_HANDLE_NONE) { + return INVALID_OPERATION; + } + + if (audioPolicyEffects != 0) { + // create audio pre processors according to input source + status_t status = audioPolicyEffects->addInputEffects(*input, inputSource, session); + if (status != NO_ERROR && status != ALREADY_EXISTS) { + ALOGW("Failed to add effects on input %d", input); + } + } + return NO_ERROR; +} + +status_t AudioPolicyService::startInput(audio_io_handle_t input, + audio_session_t session __unused) +{ + if (mpAudioPolicy == NULL) { + return NO_INIT; + } + Mutex::Autolock _l(mLock); + + return mpAudioPolicy->start_input(mpAudioPolicy, input); +} + +status_t AudioPolicyService::stopInput(audio_io_handle_t input, + audio_session_t session __unused) +{ + if (mpAudioPolicy == NULL) { + return NO_INIT; + } + Mutex::Autolock _l(mLock); + + return mpAudioPolicy->stop_input(mpAudioPolicy, input); +} + +void AudioPolicyService::releaseInput(audio_io_handle_t input, + audio_session_t session __unused) +{ + if (mpAudioPolicy == NULL) { + return; + } + + spaudioPolicyEffects; + { + Mutex::Autolock _l(mLock); + mpAudioPolicy->release_input(mpAudioPolicy, input); + audioPolicyEffects = mAudioPolicyEffects; + } + if (audioPolicyEffects != 0) { + // release audio processors from the input + status_t status = audioPolicyEffects->releaseInputEffects(input); + if(status != NO_ERROR) { + ALOGW("Failed to release effects on input %d", input); + } + } +} + +status_t AudioPolicyService::initStreamVolume(audio_stream_type_t stream, + int indexMin, + int indexMax) +{ + if (mpAudioPolicy == NULL) { + return NO_INIT; + } + if (!settingsAllowed()) { + return PERMISSION_DENIED; + } + if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT) { + return BAD_VALUE; + } + Mutex::Autolock _l(mLock); + mpAudioPolicy->init_stream_volume(mpAudioPolicy, stream, indexMin, indexMax); + return NO_ERROR; +} + +status_t AudioPolicyService::setStreamVolumeIndex(audio_stream_type_t stream, + int index, + audio_devices_t device) +{ + if (mpAudioPolicy == NULL) { + return NO_INIT; + } + if (!settingsAllowed()) { + return PERMISSION_DENIED; + } + if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT) { + return BAD_VALUE; + } + Mutex::Autolock _l(mLock); + if (mpAudioPolicy->set_stream_volume_index_for_device) { + return mpAudioPolicy->set_stream_volume_index_for_device(mpAudioPolicy, + stream, + index, + device); + } else { + return mpAudioPolicy->set_stream_volume_index(mpAudioPolicy, stream, index); + } +} + +status_t AudioPolicyService::getStreamVolumeIndex(audio_stream_type_t stream, + int *index, + audio_devices_t device) +{ + if (mpAudioPolicy == NULL) { + return NO_INIT; + } + if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT) { + return BAD_VALUE; + } + Mutex::Autolock _l(mLock); + if (mpAudioPolicy->get_stream_volume_index_for_device) { + return mpAudioPolicy->get_stream_volume_index_for_device(mpAudioPolicy, + stream, + index, + device); + } else { + return mpAudioPolicy->get_stream_volume_index(mpAudioPolicy, stream, index); + } +} + +uint32_t AudioPolicyService::getStrategyForStream(audio_stream_type_t stream) +{ + if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT) { + return 0; + } + if (mpAudioPolicy == NULL) { + return 0; + } + return mpAudioPolicy->get_strategy_for_stream(mpAudioPolicy, stream); +} + +//audio policy: use audio_device_t appropriately + +audio_devices_t AudioPolicyService::getDevicesForStream(audio_stream_type_t stream) +{ + if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT) { + return AUDIO_DEVICE_NONE; + } + if (mpAudioPolicy == NULL) { + return AUDIO_DEVICE_NONE; + } + return mpAudioPolicy->get_devices_for_stream(mpAudioPolicy, stream); +} + +audio_io_handle_t AudioPolicyService::getOutputForEffect(const effect_descriptor_t *desc) +{ + // FIXME change return type to status_t, and return NO_INIT here + if (mpAudioPolicy == NULL) { + return 0; + } + Mutex::Autolock _l(mLock); + return mpAudioPolicy->get_output_for_effect(mpAudioPolicy, desc); +} + +status_t AudioPolicyService::registerEffect(const effect_descriptor_t *desc, + audio_io_handle_t io, + uint32_t strategy, + int session, + int id) +{ + if (mpAudioPolicy == NULL) { + return NO_INIT; + } + return mpAudioPolicy->register_effect(mpAudioPolicy, desc, io, strategy, session, id); +} + +status_t AudioPolicyService::unregisterEffect(int id) +{ + if (mpAudioPolicy == NULL) { + return NO_INIT; + } + return mpAudioPolicy->unregister_effect(mpAudioPolicy, id); +} + +status_t AudioPolicyService::setEffectEnabled(int id, bool enabled) +{ + if (mpAudioPolicy == NULL) { + return NO_INIT; + } + return mpAudioPolicy->set_effect_enabled(mpAudioPolicy, id, enabled); +} + +bool AudioPolicyService::isStreamActive(audio_stream_type_t stream, uint32_t inPastMs) const +{ + if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT) { + return false; + } + if (mpAudioPolicy == NULL) { + return false; + } + Mutex::Autolock _l(mLock); + return mpAudioPolicy->is_stream_active(mpAudioPolicy, stream, inPastMs); +} + +bool AudioPolicyService::isStreamActiveRemotely(audio_stream_type_t stream, uint32_t inPastMs) const +{ + if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT) { + return false; + } + if (mpAudioPolicy == NULL) { + return false; + } + Mutex::Autolock _l(mLock); + return mpAudioPolicy->is_stream_active_remotely(mpAudioPolicy, stream, inPastMs); +} + +bool AudioPolicyService::isSourceActive(audio_source_t source) const +{ + if (mpAudioPolicy == NULL) { + return false; + } + if (mpAudioPolicy->is_source_active == 0) { + return false; + } + Mutex::Autolock _l(mLock); + return mpAudioPolicy->is_source_active(mpAudioPolicy, source); +} + +status_t AudioPolicyService::queryDefaultPreProcessing(int audioSession, + effect_descriptor_t *descriptors, + uint32_t *count) +{ + if (mpAudioPolicy == NULL) { + *count = 0; + return NO_INIT; + } + spaudioPolicyEffects; + { + Mutex::Autolock _l(mLock); + audioPolicyEffects = mAudioPolicyEffects; + } + if (audioPolicyEffects == 0) { + *count = 0; + return NO_INIT; + } + return audioPolicyEffects->queryDefaultInputEffects(audioSession, descriptors, count); +} + +bool AudioPolicyService::isOffloadSupported(const audio_offload_info_t& info) +{ + if (mpAudioPolicy == NULL) { + ALOGV("mpAudioPolicy == NULL"); + return false; + } + + if (mpAudioPolicy->is_offload_supported == NULL) { + ALOGV("HAL does not implement is_offload_supported"); + return false; + } + + return mpAudioPolicy->is_offload_supported(mpAudioPolicy, &info); +} + +status_t AudioPolicyService::listAudioPorts(audio_port_role_t role __unused, + audio_port_type_t type __unused, + unsigned int *num_ports, + struct audio_port *ports __unused, + unsigned int *generation __unused) +{ + *num_ports = 0; + return INVALID_OPERATION; +} + +status_t AudioPolicyService::getAudioPort(struct audio_port *port __unused) +{ + return INVALID_OPERATION; +} + +status_t AudioPolicyService::createAudioPatch(const struct audio_patch *patch __unused, + audio_patch_handle_t *handle __unused) +{ + return INVALID_OPERATION; +} + +status_t AudioPolicyService::releaseAudioPatch(audio_patch_handle_t handle __unused) +{ + return INVALID_OPERATION; +} + +status_t AudioPolicyService::listAudioPatches(unsigned int *num_patches, + struct audio_patch *patches __unused, + unsigned int *generation __unused) +{ + *num_patches = 0; + return INVALID_OPERATION; +} + +status_t AudioPolicyService::setAudioPortConfig(const struct audio_port_config *config __unused) +{ + return INVALID_OPERATION; +} + +status_t AudioPolicyService::getOutputForAttr(const audio_attributes_t *attr, + audio_io_handle_t *output, + audio_session_t session __unused, + audio_stream_type_t *stream, + uint32_t samplingRate, + audio_format_t format, + audio_channel_mask_t channelMask, + audio_output_flags_t flags, + const audio_offload_info_t *offloadInfo) +{ + if (attr != NULL) { + *stream = audio_attributes_to_stream_type(attr); + } else { + if (*stream == AUDIO_STREAM_DEFAULT) { + return BAD_VALUE; + } + } + *output = getOutput(*stream, samplingRate, format, channelMask, + flags, offloadInfo); + if (*output == AUDIO_IO_HANDLE_NONE) { + return INVALID_OPERATION; + } + return NO_ERROR; +} + +status_t AudioPolicyService::acquireSoundTriggerSession(audio_session_t *session __unused, + audio_io_handle_t *ioHandle __unused, + audio_devices_t *device __unused) +{ + return INVALID_OPERATION; +} + +status_t AudioPolicyService::releaseSoundTriggerSession(audio_session_t session __unused) +{ + return INVALID_OPERATION; +} + +status_t AudioPolicyService::registerPolicyMixes(Vector mixes __unused, + bool registration __unused) +{ + return INVALID_OPERATION; +} + +}; // namespace android diff --git a/services/audiopolicy/service/AudioPolicyService.cpp b/services/audiopolicy/service/AudioPolicyService.cpp new file mode 100644 index 0000000..eb9116d --- /dev/null +++ b/services/audiopolicy/service/AudioPolicyService.cpp @@ -0,0 +1,1068 @@ +/* + * Copyright (C) 2009 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#define LOG_TAG "AudioPolicyService" +//#define LOG_NDEBUG 0 + +#include "Configuration.h" +#undef __STRICT_ANSI__ +#define __STDINT_LIMITS +#define __STDC_LIMIT_MACROS +#include + +#include +#include +#include +#include +#include +#include +#include +#include "AudioPolicyService.h" +#include "ServiceUtilities.h" +#include +#include +#include +#include + +#include +#include +#include +#include + +namespace android { + +static const char kDeadlockedString[] = "AudioPolicyService may be deadlocked\n"; +static const char kCmdDeadlockedString[] = "AudioPolicyService command thread may be deadlocked\n"; + +static const int kDumpLockRetries = 50; +static const int kDumpLockSleepUs = 20000; + +static const nsecs_t kAudioCommandTimeoutNs = seconds(3); // 3 seconds + +namespace { + extern struct audio_policy_service_ops aps_ops; +}; + +// ---------------------------------------------------------------------------- + +AudioPolicyService::AudioPolicyService() + : BnAudioPolicyService(), mpAudioPolicyDev(NULL), mpAudioPolicy(NULL), + mAudioPolicyManager(NULL), mAudioPolicyClient(NULL), mPhoneState(AUDIO_MODE_INVALID) +{ +} + +void AudioPolicyService::onFirstRef() +{ + char value[PROPERTY_VALUE_MAX]; + const struct hw_module_t *module; + int forced_val; + int rc; + + { + Mutex::Autolock _l(mLock); + + // start tone playback thread + mTonePlaybackThread = new AudioCommandThread(String8("ApmTone"), this); + // start audio commands thread + mAudioCommandThread = new AudioCommandThread(String8("ApmAudio"), this); + // start output activity command thread + mOutputCommandThread = new AudioCommandThread(String8("ApmOutput"), this); + +#ifdef USE_LEGACY_AUDIO_POLICY + ALOGI("AudioPolicyService CSTOR in legacy mode"); + + /* instantiate the audio policy manager */ + rc = hw_get_module(AUDIO_POLICY_HARDWARE_MODULE_ID, &module); + if (rc) { + return; + } + rc = audio_policy_dev_open(module, &mpAudioPolicyDev); + ALOGE_IF(rc, "couldn't open audio policy device (%s)", strerror(-rc)); + if (rc) { + return; + } + + rc = mpAudioPolicyDev->create_audio_policy(mpAudioPolicyDev, &aps_ops, this, + &mpAudioPolicy); + ALOGE_IF(rc, "couldn't create audio policy (%s)", strerror(-rc)); + if (rc) { + return; + } + + rc = mpAudioPolicy->init_check(mpAudioPolicy); + ALOGE_IF(rc, "couldn't init_check the audio policy (%s)", strerror(-rc)); + if (rc) { + return; + } + ALOGI("Loaded audio policy from %s (%s)", module->name, module->id); +#else + ALOGI("AudioPolicyService CSTOR in new mode"); + + mAudioPolicyClient = new AudioPolicyClient(this); + mAudioPolicyManager = createAudioPolicyManager(mAudioPolicyClient); +#endif + } + // load audio processing modules + spaudioPolicyEffects = new AudioPolicyEffects(); + { + Mutex::Autolock _l(mLock); + mAudioPolicyEffects = audioPolicyEffects; + } +} + +AudioPolicyService::~AudioPolicyService() +{ + mTonePlaybackThread->exit(); + mAudioCommandThread->exit(); + mOutputCommandThread->exit(); + +#ifdef USE_LEGACY_AUDIO_POLICY + if (mpAudioPolicy != NULL && mpAudioPolicyDev != NULL) { + mpAudioPolicyDev->destroy_audio_policy(mpAudioPolicyDev, mpAudioPolicy); + } + if (mpAudioPolicyDev != NULL) { + audio_policy_dev_close(mpAudioPolicyDev); + } +#else + destroyAudioPolicyManager(mAudioPolicyManager); + delete mAudioPolicyClient; +#endif + + mNotificationClients.clear(); + mAudioPolicyEffects.clear(); +} + +// A notification client is always registered by AudioSystem when the client process +// connects to AudioPolicyService. +void AudioPolicyService::registerClient(const sp& client) +{ + + Mutex::Autolock _l(mNotificationClientsLock); + + uid_t uid = IPCThreadState::self()->getCallingUid(); + if (mNotificationClients.indexOfKey(uid) < 0) { + sp notificationClient = new NotificationClient(this, + client, + uid); + ALOGV("registerClient() client %p, uid %d", client.get(), uid); + + mNotificationClients.add(uid, notificationClient); + + sp binder = IInterface::asBinder(client); + binder->linkToDeath(notificationClient); + } +} + +// removeNotificationClient() is called when the client process dies. +void AudioPolicyService::removeNotificationClient(uid_t uid) +{ + { + Mutex::Autolock _l(mNotificationClientsLock); + mNotificationClients.removeItem(uid); + } +#ifndef USE_LEGACY_AUDIO_POLICY + { + Mutex::Autolock _l(mLock); + if (mAudioPolicyManager) { + mAudioPolicyManager->clearAudioPatches(uid); + } + } +#endif +} + +void AudioPolicyService::onAudioPortListUpdate() +{ + mOutputCommandThread->updateAudioPortListCommand(); +} + +void AudioPolicyService::doOnAudioPortListUpdate() +{ + Mutex::Autolock _l(mNotificationClientsLock); + for (size_t i = 0; i < mNotificationClients.size(); i++) { + mNotificationClients.valueAt(i)->onAudioPortListUpdate(); + } +} + +void AudioPolicyService::onAudioPatchListUpdate() +{ + mOutputCommandThread->updateAudioPatchListCommand(); +} + +status_t AudioPolicyService::clientCreateAudioPatch(const struct audio_patch *patch, + audio_patch_handle_t *handle, + int delayMs) +{ + return mAudioCommandThread->createAudioPatchCommand(patch, handle, delayMs); +} + +status_t AudioPolicyService::clientReleaseAudioPatch(audio_patch_handle_t handle, + int delayMs) +{ + return mAudioCommandThread->releaseAudioPatchCommand(handle, delayMs); +} + +void AudioPolicyService::doOnAudioPatchListUpdate() +{ + Mutex::Autolock _l(mNotificationClientsLock); + for (size_t i = 0; i < mNotificationClients.size(); i++) { + mNotificationClients.valueAt(i)->onAudioPatchListUpdate(); + } +} + +status_t AudioPolicyService::clientSetAudioPortConfig(const struct audio_port_config *config, + int delayMs) +{ + return mAudioCommandThread->setAudioPortConfigCommand(config, delayMs); +} + +AudioPolicyService::NotificationClient::NotificationClient(const sp& service, + const sp& client, + uid_t uid) + : mService(service), mUid(uid), mAudioPolicyServiceClient(client) +{ +} + +AudioPolicyService::NotificationClient::~NotificationClient() +{ +} + +void AudioPolicyService::NotificationClient::binderDied(const wp& who __unused) +{ + sp keep(this); + sp service = mService.promote(); + if (service != 0) { + service->removeNotificationClient(mUid); + } +} + +void AudioPolicyService::NotificationClient::onAudioPortListUpdate() +{ + if (mAudioPolicyServiceClient != 0) { + mAudioPolicyServiceClient->onAudioPortListUpdate(); + } +} + +void AudioPolicyService::NotificationClient::onAudioPatchListUpdate() +{ + if (mAudioPolicyServiceClient != 0) { + mAudioPolicyServiceClient->onAudioPatchListUpdate(); + } +} + +void AudioPolicyService::binderDied(const wp& who) { + ALOGW("binderDied() %p, calling pid %d", who.unsafe_get(), + IPCThreadState::self()->getCallingPid()); +} + +static bool tryLock(Mutex& mutex) +{ + bool locked = false; + for (int i = 0; i < kDumpLockRetries; ++i) { + if (mutex.tryLock() == NO_ERROR) { + locked = true; + break; + } + usleep(kDumpLockSleepUs); + } + return locked; +} + +status_t AudioPolicyService::dumpInternals(int fd) +{ + const size_t SIZE = 256; + char buffer[SIZE]; + String8 result; + +#ifdef USE_LEGACY_AUDIO_POLICY + snprintf(buffer, SIZE, "PolicyManager Interface: %p\n", mpAudioPolicy); +#else + snprintf(buffer, SIZE, "AudioPolicyManager: %p\n", mAudioPolicyManager); +#endif + result.append(buffer); + snprintf(buffer, SIZE, "Command Thread: %p\n", mAudioCommandThread.get()); + result.append(buffer); + snprintf(buffer, SIZE, "Tones Thread: %p\n", mTonePlaybackThread.get()); + result.append(buffer); + + write(fd, result.string(), result.size()); + return NO_ERROR; +} + +status_t AudioPolicyService::dump(int fd, const Vector& args __unused) +{ + if (!dumpAllowed()) { + dumpPermissionDenial(fd); + } else { + bool locked = tryLock(mLock); + if (!locked) { + String8 result(kDeadlockedString); + write(fd, result.string(), result.size()); + } + + dumpInternals(fd); + if (mAudioCommandThread != 0) { + mAudioCommandThread->dump(fd); + } + if (mTonePlaybackThread != 0) { + mTonePlaybackThread->dump(fd); + } + +#ifdef USE_LEGACY_AUDIO_POLICY + if (mpAudioPolicy) { + mpAudioPolicy->dump(mpAudioPolicy, fd); + } +#else + if (mAudioPolicyManager) { + mAudioPolicyManager->dump(fd); + } +#endif + + if (locked) mLock.unlock(); + } + return NO_ERROR; +} + +status_t AudioPolicyService::dumpPermissionDenial(int fd) +{ + const size_t SIZE = 256; + char buffer[SIZE]; + String8 result; + snprintf(buffer, SIZE, "Permission Denial: " + "can't dump AudioPolicyService from pid=%d, uid=%d\n", + IPCThreadState::self()->getCallingPid(), + IPCThreadState::self()->getCallingUid()); + result.append(buffer); + write(fd, result.string(), result.size()); + return NO_ERROR; +} + +status_t AudioPolicyService::onTransact( + uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) +{ + return BnAudioPolicyService::onTransact(code, data, reply, flags); +} + + +// ----------- AudioPolicyService::AudioCommandThread implementation ---------- + +AudioPolicyService::AudioCommandThread::AudioCommandThread(String8 name, + const wp& service) + : Thread(false), mName(name), mService(service) +{ + mpToneGenerator = NULL; +} + + +AudioPolicyService::AudioCommandThread::~AudioCommandThread() +{ + if (!mAudioCommands.isEmpty()) { + release_wake_lock(mName.string()); + } + mAudioCommands.clear(); + delete mpToneGenerator; +} + +void AudioPolicyService::AudioCommandThread::onFirstRef() +{ + run(mName.string(), ANDROID_PRIORITY_AUDIO); +} + +bool AudioPolicyService::AudioCommandThread::threadLoop() +{ + nsecs_t waitTime = INT64_MAX; + + mLock.lock(); + while (!exitPending()) + { + sp svc; + while (!mAudioCommands.isEmpty() && !exitPending()) { + nsecs_t curTime = systemTime(); + // commands are sorted by increasing time stamp: execute them from index 0 and up + if (mAudioCommands[0]->mTime <= curTime) { + sp command = mAudioCommands[0]; + mAudioCommands.removeAt(0); + mLastCommand = command; + + switch (command->mCommand) { + case START_TONE: { + mLock.unlock(); + ToneData *data = (ToneData *)command->mParam.get(); + ALOGV("AudioCommandThread() processing start tone %d on stream %d", + data->mType, data->mStream); + delete mpToneGenerator; + mpToneGenerator = new ToneGenerator(data->mStream, 1.0); + mpToneGenerator->startTone(data->mType); + mLock.lock(); + }break; + case STOP_TONE: { + mLock.unlock(); + ALOGV("AudioCommandThread() processing stop tone"); + if (mpToneGenerator != NULL) { + mpToneGenerator->stopTone(); + delete mpToneGenerator; + mpToneGenerator = NULL; + } + mLock.lock(); + }break; + case SET_VOLUME: { + VolumeData *data = (VolumeData *)command->mParam.get(); + ALOGV("AudioCommandThread() processing set volume stream %d, \ + volume %f, output %d", data->mStream, data->mVolume, data->mIO); + command->mStatus = AudioSystem::setStreamVolume(data->mStream, + data->mVolume, + data->mIO); + }break; + case SET_PARAMETERS: { + ParametersData *data = (ParametersData *)command->mParam.get(); + ALOGV("AudioCommandThread() processing set parameters string %s, io %d", + data->mKeyValuePairs.string(), data->mIO); + command->mStatus = AudioSystem::setParameters(data->mIO, data->mKeyValuePairs); + }break; + case SET_VOICE_VOLUME: { + VoiceVolumeData *data = (VoiceVolumeData *)command->mParam.get(); + ALOGV("AudioCommandThread() processing set voice volume volume %f", + data->mVolume); + command->mStatus = AudioSystem::setVoiceVolume(data->mVolume); + }break; + case STOP_OUTPUT: { + StopOutputData *data = (StopOutputData *)command->mParam.get(); + ALOGV("AudioCommandThread() processing stop output %d", + data->mIO); + svc = mService.promote(); + if (svc == 0) { + break; + } + mLock.unlock(); + svc->doStopOutput(data->mIO, data->mStream, data->mSession); + mLock.lock(); + }break; + case RELEASE_OUTPUT: { + ReleaseOutputData *data = (ReleaseOutputData *)command->mParam.get(); + ALOGV("AudioCommandThread() processing release output %d", + data->mIO); + svc = mService.promote(); + if (svc == 0) { + break; + } + mLock.unlock(); + svc->doReleaseOutput(data->mIO, data->mStream, data->mSession); + mLock.lock(); + }break; + case CREATE_AUDIO_PATCH: { + CreateAudioPatchData *data = (CreateAudioPatchData *)command->mParam.get(); + ALOGV("AudioCommandThread() processing create audio patch"); + sp af = AudioSystem::get_audio_flinger(); + if (af == 0) { + command->mStatus = PERMISSION_DENIED; + } else { + command->mStatus = af->createAudioPatch(&data->mPatch, &data->mHandle); + } + } break; + case RELEASE_AUDIO_PATCH: { + ReleaseAudioPatchData *data = (ReleaseAudioPatchData *)command->mParam.get(); + ALOGV("AudioCommandThread() processing release audio patch"); + sp af = AudioSystem::get_audio_flinger(); + if (af == 0) { + command->mStatus = PERMISSION_DENIED; + } else { + command->mStatus = af->releaseAudioPatch(data->mHandle); + } + } break; + case UPDATE_AUDIOPORT_LIST: { + ALOGV("AudioCommandThread() processing update audio port list"); + svc = mService.promote(); + if (svc == 0) { + break; + } + mLock.unlock(); + svc->doOnAudioPortListUpdate(); + mLock.lock(); + }break; + case UPDATE_AUDIOPATCH_LIST: { + ALOGV("AudioCommandThread() processing update audio patch list"); + svc = mService.promote(); + if (svc == 0) { + break; + } + mLock.unlock(); + svc->doOnAudioPatchListUpdate(); + mLock.lock(); + }break; + case SET_AUDIOPORT_CONFIG: { + SetAudioPortConfigData *data = (SetAudioPortConfigData *)command->mParam.get(); + ALOGV("AudioCommandThread() processing set port config"); + sp af = AudioSystem::get_audio_flinger(); + if (af == 0) { + command->mStatus = PERMISSION_DENIED; + } else { + command->mStatus = af->setAudioPortConfig(&data->mConfig); + } + } break; + default: + ALOGW("AudioCommandThread() unknown command %d", command->mCommand); + } + { + Mutex::Autolock _l(command->mLock); + if (command->mWaitStatus) { + command->mWaitStatus = false; + command->mCond.signal(); + } + } + waitTime = INT64_MAX; + } else { + waitTime = mAudioCommands[0]->mTime - curTime; + break; + } + } + // release mLock before releasing strong reference on the service as + // AudioPolicyService destructor calls AudioCommandThread::exit() which acquires mLock. + mLock.unlock(); + svc.clear(); + mLock.lock(); + if (!exitPending() && mAudioCommands.isEmpty()) { + // release delayed commands wake lock + release_wake_lock(mName.string()); + ALOGV("AudioCommandThread() going to sleep"); + mWaitWorkCV.waitRelative(mLock, waitTime); + ALOGV("AudioCommandThread() waking up"); + } + } + // release delayed commands wake lock before quitting + if (!mAudioCommands.isEmpty()) { + release_wake_lock(mName.string()); + } + mLock.unlock(); + return false; +} + +status_t AudioPolicyService::AudioCommandThread::dump(int fd) +{ + const size_t SIZE = 256; + char buffer[SIZE]; + String8 result; + + snprintf(buffer, SIZE, "AudioCommandThread %p Dump\n", this); + result.append(buffer); + write(fd, result.string(), result.size()); + + bool locked = tryLock(mLock); + if (!locked) { + String8 result2(kCmdDeadlockedString); + write(fd, result2.string(), result2.size()); + } + + snprintf(buffer, SIZE, "- Commands:\n"); + result = String8(buffer); + result.append(" Command Time Wait pParam\n"); + for (size_t i = 0; i < mAudioCommands.size(); i++) { + mAudioCommands[i]->dump(buffer, SIZE); + result.append(buffer); + } + result.append(" Last Command\n"); + if (mLastCommand != 0) { + mLastCommand->dump(buffer, SIZE); + result.append(buffer); + } else { + result.append(" none\n"); + } + + write(fd, result.string(), result.size()); + + if (locked) mLock.unlock(); + + return NO_ERROR; +} + +void AudioPolicyService::AudioCommandThread::startToneCommand(ToneGenerator::tone_type type, + audio_stream_type_t stream) +{ + sp command = new AudioCommand(); + command->mCommand = START_TONE; + sp data = new ToneData(); + data->mType = type; + data->mStream = stream; + command->mParam = data; + ALOGV("AudioCommandThread() adding tone start type %d, stream %d", type, stream); + sendCommand(command); +} + +void AudioPolicyService::AudioCommandThread::stopToneCommand() +{ + sp command = new AudioCommand(); + command->mCommand = STOP_TONE; + ALOGV("AudioCommandThread() adding tone stop"); + sendCommand(command); +} + +status_t AudioPolicyService::AudioCommandThread::volumeCommand(audio_stream_type_t stream, + float volume, + audio_io_handle_t output, + int delayMs) +{ + sp command = new AudioCommand(); + command->mCommand = SET_VOLUME; + sp data = new VolumeData(); + data->mStream = stream; + data->mVolume = volume; + data->mIO = output; + command->mParam = data; + command->mWaitStatus = true; + ALOGV("AudioCommandThread() adding set volume stream %d, volume %f, output %d", + stream, volume, output); + return sendCommand(command, delayMs); +} + +status_t AudioPolicyService::AudioCommandThread::parametersCommand(audio_io_handle_t ioHandle, + const char *keyValuePairs, + int delayMs) +{ + sp command = new AudioCommand(); + command->mCommand = SET_PARAMETERS; + sp data = new ParametersData(); + data->mIO = ioHandle; + data->mKeyValuePairs = String8(keyValuePairs); + command->mParam = data; + command->mWaitStatus = true; + ALOGV("AudioCommandThread() adding set parameter string %s, io %d ,delay %d", + keyValuePairs, ioHandle, delayMs); + return sendCommand(command, delayMs); +} + +status_t AudioPolicyService::AudioCommandThread::voiceVolumeCommand(float volume, int delayMs) +{ + sp command = new AudioCommand(); + command->mCommand = SET_VOICE_VOLUME; + sp data = new VoiceVolumeData(); + data->mVolume = volume; + command->mParam = data; + command->mWaitStatus = true; + ALOGV("AudioCommandThread() adding set voice volume volume %f", volume); + return sendCommand(command, delayMs); +} + +void AudioPolicyService::AudioCommandThread::stopOutputCommand(audio_io_handle_t output, + audio_stream_type_t stream, + audio_session_t session) +{ + sp command = new AudioCommand(); + command->mCommand = STOP_OUTPUT; + sp data = new StopOutputData(); + data->mIO = output; + data->mStream = stream; + data->mSession = session; + command->mParam = data; + ALOGV("AudioCommandThread() adding stop output %d", output); + sendCommand(command); +} + +void AudioPolicyService::AudioCommandThread::releaseOutputCommand(audio_io_handle_t output, + audio_stream_type_t stream, + audio_session_t session) +{ + sp command = new AudioCommand(); + command->mCommand = RELEASE_OUTPUT; + sp data = new ReleaseOutputData(); + data->mIO = output; + data->mStream = stream; + data->mSession = session; + command->mParam = data; + ALOGV("AudioCommandThread() adding release output %d", output); + sendCommand(command); +} + +status_t AudioPolicyService::AudioCommandThread::createAudioPatchCommand( + const struct audio_patch *patch, + audio_patch_handle_t *handle, + int delayMs) +{ + status_t status = NO_ERROR; + + sp command = new AudioCommand(); + command->mCommand = CREATE_AUDIO_PATCH; + CreateAudioPatchData *data = new CreateAudioPatchData(); + data->mPatch = *patch; + data->mHandle = *handle; + command->mParam = data; + command->mWaitStatus = true; + ALOGV("AudioCommandThread() adding create patch delay %d", delayMs); + status = sendCommand(command, delayMs); + if (status == NO_ERROR) { + *handle = data->mHandle; + } + return status; +} + +status_t AudioPolicyService::AudioCommandThread::releaseAudioPatchCommand(audio_patch_handle_t handle, + int delayMs) +{ + sp command = new AudioCommand(); + command->mCommand = RELEASE_AUDIO_PATCH; + ReleaseAudioPatchData *data = new ReleaseAudioPatchData(); + data->mHandle = handle; + command->mParam = data; + command->mWaitStatus = true; + ALOGV("AudioCommandThread() adding release patch delay %d", delayMs); + return sendCommand(command, delayMs); +} + +void AudioPolicyService::AudioCommandThread::updateAudioPortListCommand() +{ + sp command = new AudioCommand(); + command->mCommand = UPDATE_AUDIOPORT_LIST; + ALOGV("AudioCommandThread() adding update audio port list"); + sendCommand(command); +} + +void AudioPolicyService::AudioCommandThread::updateAudioPatchListCommand() +{ + spcommand = new AudioCommand(); + command->mCommand = UPDATE_AUDIOPATCH_LIST; + ALOGV("AudioCommandThread() adding update audio patch list"); + sendCommand(command); +} + +status_t AudioPolicyService::AudioCommandThread::setAudioPortConfigCommand( + const struct audio_port_config *config, int delayMs) +{ + sp command = new AudioCommand(); + command->mCommand = SET_AUDIOPORT_CONFIG; + SetAudioPortConfigData *data = new SetAudioPortConfigData(); + data->mConfig = *config; + command->mParam = data; + command->mWaitStatus = true; + ALOGV("AudioCommandThread() adding set port config delay %d", delayMs); + return sendCommand(command, delayMs); +} + +status_t AudioPolicyService::AudioCommandThread::sendCommand(sp& command, int delayMs) +{ + { + Mutex::Autolock _l(mLock); + insertCommand_l(command, delayMs); + mWaitWorkCV.signal(); + } + Mutex::Autolock _l(command->mLock); + while (command->mWaitStatus) { + nsecs_t timeOutNs = kAudioCommandTimeoutNs + milliseconds(delayMs); + if (command->mCond.waitRelative(command->mLock, timeOutNs) != NO_ERROR) { + command->mStatus = TIMED_OUT; + command->mWaitStatus = false; + } + } + return command->mStatus; +} + +// insertCommand_l() must be called with mLock held +void AudioPolicyService::AudioCommandThread::insertCommand_l(sp& command, int delayMs) +{ + ssize_t i; // not size_t because i will count down to -1 + Vector < sp > removedCommands; + command->mTime = systemTime() + milliseconds(delayMs); + + // acquire wake lock to make sure delayed commands are processed + if (mAudioCommands.isEmpty()) { + acquire_wake_lock(PARTIAL_WAKE_LOCK, mName.string()); + } + + // check same pending commands with later time stamps and eliminate them + for (i = mAudioCommands.size()-1; i >= 0; i--) { + sp command2 = mAudioCommands[i]; + // commands are sorted by increasing time stamp: no need to scan the rest of mAudioCommands + if (command2->mTime <= command->mTime) break; + + // create audio patch or release audio patch commands are equivalent + // with regard to filtering + if ((command->mCommand == CREATE_AUDIO_PATCH) || + (command->mCommand == RELEASE_AUDIO_PATCH)) { + if ((command2->mCommand != CREATE_AUDIO_PATCH) && + (command2->mCommand != RELEASE_AUDIO_PATCH)) { + continue; + } + } else if (command2->mCommand != command->mCommand) continue; + + switch (command->mCommand) { + case SET_PARAMETERS: { + ParametersData *data = (ParametersData *)command->mParam.get(); + ParametersData *data2 = (ParametersData *)command2->mParam.get(); + if (data->mIO != data2->mIO) break; + ALOGV("Comparing parameter command %s to new command %s", + data2->mKeyValuePairs.string(), data->mKeyValuePairs.string()); + AudioParameter param = AudioParameter(data->mKeyValuePairs); + AudioParameter param2 = AudioParameter(data2->mKeyValuePairs); + for (size_t j = 0; j < param.size(); j++) { + String8 key; + String8 value; + param.getAt(j, key, value); + for (size_t k = 0; k < param2.size(); k++) { + String8 key2; + String8 value2; + param2.getAt(k, key2, value2); + if (key2 == key) { + param2.remove(key2); + ALOGV("Filtering out parameter %s", key2.string()); + break; + } + } + } + // if all keys have been filtered out, remove the command. + // otherwise, update the key value pairs + if (param2.size() == 0) { + removedCommands.add(command2); + } else { + data2->mKeyValuePairs = param2.toString(); + } + command->mTime = command2->mTime; + // force delayMs to non 0 so that code below does not request to wait for + // command status as the command is now delayed + delayMs = 1; + } break; + + case SET_VOLUME: { + VolumeData *data = (VolumeData *)command->mParam.get(); + VolumeData *data2 = (VolumeData *)command2->mParam.get(); + if (data->mIO != data2->mIO) break; + if (data->mStream != data2->mStream) break; + ALOGV("Filtering out volume command on output %d for stream %d", + data->mIO, data->mStream); + removedCommands.add(command2); + command->mTime = command2->mTime; + // force delayMs to non 0 so that code below does not request to wait for + // command status as the command is now delayed + delayMs = 1; + } break; + + case CREATE_AUDIO_PATCH: + case RELEASE_AUDIO_PATCH: { + audio_patch_handle_t handle; + struct audio_patch patch; + if (command->mCommand == CREATE_AUDIO_PATCH) { + handle = ((CreateAudioPatchData *)command->mParam.get())->mHandle; + patch = ((CreateAudioPatchData *)command->mParam.get())->mPatch; + } else { + handle = ((ReleaseAudioPatchData *)command->mParam.get())->mHandle; + } + audio_patch_handle_t handle2; + struct audio_patch patch2; + if (command2->mCommand == CREATE_AUDIO_PATCH) { + handle2 = ((CreateAudioPatchData *)command2->mParam.get())->mHandle; + patch2 = ((CreateAudioPatchData *)command2->mParam.get())->mPatch; + } else { + handle2 = ((ReleaseAudioPatchData *)command2->mParam.get())->mHandle; + } + if (handle != handle2) break; + /* Filter CREATE_AUDIO_PATCH commands only when they are issued for + same output. */ + if( (command->mCommand == CREATE_AUDIO_PATCH) && + (command2->mCommand == CREATE_AUDIO_PATCH) ) { + bool isOutputDiff = false; + if (patch.num_sources == patch2.num_sources) { + for (unsigned count = 0; count < patch.num_sources; count++) { + if (patch.sources[count].id != patch2.sources[count].id) { + isOutputDiff = true; + break; + } + } + if (isOutputDiff) + break; + } + } + ALOGV("Filtering out %s audio patch command for handle %d", + (command->mCommand == CREATE_AUDIO_PATCH) ? "create" : "release", handle); + removedCommands.add(command2); + command->mTime = command2->mTime; + // force delayMs to non 0 so that code below does not request to wait for + // command status as the command is now delayed + delayMs = 1; + } break; + + case START_TONE: + case STOP_TONE: + default: + break; + } + } + + // remove filtered commands + for (size_t j = 0; j < removedCommands.size(); j++) { + // removed commands always have time stamps greater than current command + for (size_t k = i + 1; k < mAudioCommands.size(); k++) { + if (mAudioCommands[k].get() == removedCommands[j].get()) { + ALOGV("suppressing command: %d", mAudioCommands[k]->mCommand); + mAudioCommands.removeAt(k); + break; + } + } + } + removedCommands.clear(); + + // Disable wait for status if delay is not 0. + // Except for create audio patch command because the returned patch handle + // is needed by audio policy manager + if (delayMs != 0 && command->mCommand != CREATE_AUDIO_PATCH) { + command->mWaitStatus = false; + } + + // insert command at the right place according to its time stamp + ALOGV("inserting command: %d at index %zd, num commands %zu", + command->mCommand, i+1, mAudioCommands.size()); + mAudioCommands.insertAt(command, i + 1); +} + +void AudioPolicyService::AudioCommandThread::exit() +{ + ALOGV("AudioCommandThread::exit"); + { + AutoMutex _l(mLock); + requestExit(); + mWaitWorkCV.signal(); + } + requestExitAndWait(); +} + +void AudioPolicyService::AudioCommandThread::AudioCommand::dump(char* buffer, size_t size) +{ + snprintf(buffer, size, " %02d %06d.%03d %01u %p\n", + mCommand, + (int)ns2s(mTime), + (int)ns2ms(mTime)%1000, + mWaitStatus, + mParam.get()); +} + +/******* helpers for the service_ops callbacks defined below *********/ +void AudioPolicyService::setParameters(audio_io_handle_t ioHandle, + const char *keyValuePairs, + int delayMs) +{ + mAudioCommandThread->parametersCommand(ioHandle, keyValuePairs, + delayMs); +} + +int AudioPolicyService::setStreamVolume(audio_stream_type_t stream, + float volume, + audio_io_handle_t output, + int delayMs) +{ + return (int)mAudioCommandThread->volumeCommand(stream, volume, + output, delayMs); +} + +int AudioPolicyService::startTone(audio_policy_tone_t tone, + audio_stream_type_t stream) +{ + if (tone != AUDIO_POLICY_TONE_IN_CALL_NOTIFICATION) { + ALOGE("startTone: illegal tone requested (%d)", tone); + } + if (stream != AUDIO_STREAM_VOICE_CALL) { + ALOGE("startTone: illegal stream (%d) requested for tone %d", stream, + tone); + } + mTonePlaybackThread->startToneCommand(ToneGenerator::TONE_SUP_CALL_WAITING, + AUDIO_STREAM_VOICE_CALL); + return 0; +} + +int AudioPolicyService::stopTone() +{ + mTonePlaybackThread->stopToneCommand(); + return 0; +} + +int AudioPolicyService::setVoiceVolume(float volume, int delayMs) +{ + return (int)mAudioCommandThread->voiceVolumeCommand(volume, delayMs); +} + +extern "C" { +audio_module_handle_t aps_load_hw_module(void *service __unused, + const char *name); +audio_io_handle_t aps_open_output(void *service __unused, + audio_devices_t *pDevices, + uint32_t *pSamplingRate, + audio_format_t *pFormat, + audio_channel_mask_t *pChannelMask, + uint32_t *pLatencyMs, + audio_output_flags_t flags); + +audio_io_handle_t aps_open_output_on_module(void *service __unused, + audio_module_handle_t module, + audio_devices_t *pDevices, + uint32_t *pSamplingRate, + audio_format_t *pFormat, + audio_channel_mask_t *pChannelMask, + uint32_t *pLatencyMs, + audio_output_flags_t flags, + const audio_offload_info_t *offloadInfo); +audio_io_handle_t aps_open_dup_output(void *service __unused, + audio_io_handle_t output1, + audio_io_handle_t output2); +int aps_close_output(void *service __unused, audio_io_handle_t output); +int aps_suspend_output(void *service __unused, audio_io_handle_t output); +int aps_restore_output(void *service __unused, audio_io_handle_t output); +audio_io_handle_t aps_open_input(void *service __unused, + audio_devices_t *pDevices, + uint32_t *pSamplingRate, + audio_format_t *pFormat, + audio_channel_mask_t *pChannelMask, + audio_in_acoustics_t acoustics __unused); +audio_io_handle_t aps_open_input_on_module(void *service __unused, + audio_module_handle_t module, + audio_devices_t *pDevices, + uint32_t *pSamplingRate, + audio_format_t *pFormat, + audio_channel_mask_t *pChannelMask); +int aps_close_input(void *service __unused, audio_io_handle_t input); +int aps_invalidate_stream(void *service __unused, audio_stream_type_t stream); +int aps_move_effects(void *service __unused, int session, + audio_io_handle_t src_output, + audio_io_handle_t dst_output); +char * aps_get_parameters(void *service __unused, audio_io_handle_t io_handle, + const char *keys); +void aps_set_parameters(void *service, audio_io_handle_t io_handle, + const char *kv_pairs, int delay_ms); +int aps_set_stream_volume(void *service, audio_stream_type_t stream, + float volume, audio_io_handle_t output, + int delay_ms); +int aps_start_tone(void *service, audio_policy_tone_t tone, + audio_stream_type_t stream); +int aps_stop_tone(void *service); +int aps_set_voice_volume(void *service, float volume, int delay_ms); +}; + +namespace { + struct audio_policy_service_ops aps_ops = { + .open_output = aps_open_output, + .open_duplicate_output = aps_open_dup_output, + .close_output = aps_close_output, + .suspend_output = aps_suspend_output, + .restore_output = aps_restore_output, + .open_input = aps_open_input, + .close_input = aps_close_input, + .set_stream_volume = aps_set_stream_volume, + .invalidate_stream = aps_invalidate_stream, + .set_parameters = aps_set_parameters, + .get_parameters = aps_get_parameters, + .start_tone = aps_start_tone, + .stop_tone = aps_stop_tone, + .set_voice_volume = aps_set_voice_volume, + .move_effects = aps_move_effects, + .load_hw_module = aps_load_hw_module, + .open_output_on_module = aps_open_output_on_module, + .open_input_on_module = aps_open_input_on_module, + }; +}; // namespace + +}; // namespace android diff --git a/services/audiopolicy/service/AudioPolicyService.h b/services/audiopolicy/service/AudioPolicyService.h new file mode 100644 index 0000000..0378384 --- /dev/null +++ b/services/audiopolicy/service/AudioPolicyService.h @@ -0,0 +1,524 @@ +/* + * Copyright (C) 2009 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef ANDROID_AUDIOPOLICYSERVICE_H +#define ANDROID_AUDIOPOLICYSERVICE_H + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#ifdef USE_LEGACY_AUDIO_POLICY +#include +#endif +#include "AudioPolicyEffects.h" +#include "managerdefault/AudioPolicyManager.h" + + +namespace android { + +// ---------------------------------------------------------------------------- + +class AudioPolicyService : + public BinderService, + public BnAudioPolicyService, + public IBinder::DeathRecipient +{ + friend class BinderService; + +public: + // for BinderService + static const char *getServiceName() ANDROID_API { return "media.audio_policy"; } + + virtual status_t dump(int fd, const Vector& args); + + // + // BnAudioPolicyService (see AudioPolicyInterface for method descriptions) + // + + virtual status_t setDeviceConnectionState(audio_devices_t device, + audio_policy_dev_state_t state, + const char *device_address, + const char *device_name); + virtual audio_policy_dev_state_t getDeviceConnectionState( + audio_devices_t device, + const char *device_address); + virtual status_t setPhoneState(audio_mode_t state); + virtual status_t setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config); + virtual audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage); + virtual audio_io_handle_t getOutput(audio_stream_type_t stream, + uint32_t samplingRate = 0, + audio_format_t format = AUDIO_FORMAT_DEFAULT, + audio_channel_mask_t channelMask = 0, + audio_output_flags_t flags = + AUDIO_OUTPUT_FLAG_NONE, + const audio_offload_info_t *offloadInfo = NULL); + virtual status_t getOutputForAttr(const audio_attributes_t *attr, + audio_io_handle_t *output, + audio_session_t session, + audio_stream_type_t *stream, + uint32_t samplingRate = 0, + audio_format_t format = AUDIO_FORMAT_DEFAULT, + audio_channel_mask_t channelMask = 0, + audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, + const audio_offload_info_t *offloadInfo = NULL); + virtual status_t startOutput(audio_io_handle_t output, + audio_stream_type_t stream, + audio_session_t session); + virtual status_t stopOutput(audio_io_handle_t output, + audio_stream_type_t stream, + audio_session_t session); + virtual void releaseOutput(audio_io_handle_t output, + audio_stream_type_t stream, + audio_session_t session); + virtual status_t getInputForAttr(const audio_attributes_t *attr, + audio_io_handle_t *input, + audio_session_t session, + uint32_t samplingRate, + audio_format_t format, + audio_channel_mask_t channelMask, + audio_input_flags_t flags); + virtual status_t startInput(audio_io_handle_t input, + audio_session_t session); + virtual status_t stopInput(audio_io_handle_t input, + audio_session_t session); + virtual void releaseInput(audio_io_handle_t input, + audio_session_t session); + virtual status_t initStreamVolume(audio_stream_type_t stream, + int indexMin, + int indexMax); + virtual status_t setStreamVolumeIndex(audio_stream_type_t stream, + int index, + audio_devices_t device); + virtual status_t getStreamVolumeIndex(audio_stream_type_t stream, + int *index, + audio_devices_t device); + + virtual uint32_t getStrategyForStream(audio_stream_type_t stream); + virtual audio_devices_t getDevicesForStream(audio_stream_type_t stream); + + virtual audio_io_handle_t getOutputForEffect(const effect_descriptor_t *desc); + virtual status_t registerEffect(const effect_descriptor_t *desc, + audio_io_handle_t io, + uint32_t strategy, + int session, + int id); + virtual status_t unregisterEffect(int id); + virtual status_t setEffectEnabled(int id, bool enabled); + virtual bool isStreamActive(audio_stream_type_t stream, uint32_t inPastMs = 0) const; + virtual bool isStreamActiveRemotely(audio_stream_type_t stream, uint32_t inPastMs = 0) const; + virtual bool isSourceActive(audio_source_t source) const; + + virtual status_t queryDefaultPreProcessing(int audioSession, + effect_descriptor_t *descriptors, + uint32_t *count); + virtual status_t onTransact( + uint32_t code, + const Parcel& data, + Parcel* reply, + uint32_t flags); + + // IBinder::DeathRecipient + virtual void binderDied(const wp& who); + + // RefBase + virtual void onFirstRef(); + + // + // Helpers for the struct audio_policy_service_ops implementation. + // This is used by the audio policy manager for certain operations that + // are implemented by the policy service. + // + virtual void setParameters(audio_io_handle_t ioHandle, + const char *keyValuePairs, + int delayMs); + + virtual status_t setStreamVolume(audio_stream_type_t stream, + float volume, + audio_io_handle_t output, + int delayMs = 0); + virtual status_t startTone(audio_policy_tone_t tone, audio_stream_type_t stream); + virtual status_t stopTone(); + virtual status_t setVoiceVolume(float volume, int delayMs = 0); + virtual bool isOffloadSupported(const audio_offload_info_t &config); + + virtual status_t listAudioPorts(audio_port_role_t role, + audio_port_type_t type, + unsigned int *num_ports, + struct audio_port *ports, + unsigned int *generation); + virtual status_t getAudioPort(struct audio_port *port); + virtual status_t createAudioPatch(const struct audio_patch *patch, + audio_patch_handle_t *handle); + virtual status_t releaseAudioPatch(audio_patch_handle_t handle); + virtual status_t listAudioPatches(unsigned int *num_patches, + struct audio_patch *patches, + unsigned int *generation); + virtual status_t setAudioPortConfig(const struct audio_port_config *config); + + virtual void registerClient(const sp& client); + + virtual status_t acquireSoundTriggerSession(audio_session_t *session, + audio_io_handle_t *ioHandle, + audio_devices_t *device); + + virtual status_t releaseSoundTriggerSession(audio_session_t session); + + virtual audio_mode_t getPhoneState(); + + virtual status_t registerPolicyMixes(Vector mixes, bool registration); + + status_t doStopOutput(audio_io_handle_t output, + audio_stream_type_t stream, + audio_session_t session); + void doReleaseOutput(audio_io_handle_t output, + audio_stream_type_t stream, + audio_session_t session); + + status_t clientCreateAudioPatch(const struct audio_patch *patch, + audio_patch_handle_t *handle, + int delayMs); + status_t clientReleaseAudioPatch(audio_patch_handle_t handle, + int delayMs); + virtual status_t clientSetAudioPortConfig(const struct audio_port_config *config, + int delayMs); + + void removeNotificationClient(uid_t uid); + void onAudioPortListUpdate(); + void doOnAudioPortListUpdate(); + void onAudioPatchListUpdate(); + void doOnAudioPatchListUpdate(); + +private: + AudioPolicyService() ANDROID_API; + virtual ~AudioPolicyService(); + + status_t dumpInternals(int fd); + + // Thread used for tone playback and to send audio config commands to audio flinger + // For tone playback, using a separate thread is necessary to avoid deadlock with mLock because + // startTone() and stopTone() are normally called with mLock locked and requesting a tone start + // or stop will cause calls to AudioPolicyService and an attempt to lock mLock. + // For audio config commands, it is necessary because audio flinger requires that the calling + // process (user) has permission to modify audio settings. + class AudioCommandThread : public Thread { + class AudioCommand; + public: + + // commands for tone AudioCommand + enum { + START_TONE, + STOP_TONE, + SET_VOLUME, + SET_PARAMETERS, + SET_VOICE_VOLUME, + STOP_OUTPUT, + RELEASE_OUTPUT, + CREATE_AUDIO_PATCH, + RELEASE_AUDIO_PATCH, + UPDATE_AUDIOPORT_LIST, + UPDATE_AUDIOPATCH_LIST, + SET_AUDIOPORT_CONFIG, + }; + + AudioCommandThread (String8 name, const wp& service); + virtual ~AudioCommandThread(); + + status_t dump(int fd); + + // Thread virtuals + virtual void onFirstRef(); + virtual bool threadLoop(); + + void exit(); + void startToneCommand(ToneGenerator::tone_type type, + audio_stream_type_t stream); + void stopToneCommand(); + status_t volumeCommand(audio_stream_type_t stream, float volume, + audio_io_handle_t output, int delayMs = 0); + status_t parametersCommand(audio_io_handle_t ioHandle, + const char *keyValuePairs, int delayMs = 0); + status_t voiceVolumeCommand(float volume, int delayMs = 0); + void stopOutputCommand(audio_io_handle_t output, + audio_stream_type_t stream, + audio_session_t session); + void releaseOutputCommand(audio_io_handle_t output, + audio_stream_type_t stream, + audio_session_t session); + status_t sendCommand(sp& command, int delayMs = 0); + void insertCommand_l(sp& command, int delayMs = 0); + status_t createAudioPatchCommand(const struct audio_patch *patch, + audio_patch_handle_t *handle, + int delayMs); + status_t releaseAudioPatchCommand(audio_patch_handle_t handle, + int delayMs); + void updateAudioPortListCommand(); + void updateAudioPatchListCommand(); + status_t setAudioPortConfigCommand(const struct audio_port_config *config, + int delayMs); + void insertCommand_l(AudioCommand *command, int delayMs = 0); + + private: + class AudioCommandData; + + // descriptor for requested tone playback event + class AudioCommand: public RefBase { + + public: + AudioCommand() + : mCommand(-1), mStatus(NO_ERROR), mWaitStatus(false) {} + + void dump(char* buffer, size_t size); + + int mCommand; // START_TONE, STOP_TONE ... + nsecs_t mTime; // time stamp + Mutex mLock; // mutex associated to mCond + Condition mCond; // condition for status return + status_t mStatus; // command status + bool mWaitStatus; // true if caller is waiting for status + sp mParam; // command specific parameter data + }; + + class AudioCommandData: public RefBase { + public: + virtual ~AudioCommandData() {} + protected: + AudioCommandData() {} + }; + + class ToneData : public AudioCommandData { + public: + ToneGenerator::tone_type mType; // tone type (START_TONE only) + audio_stream_type_t mStream; // stream type (START_TONE only) + }; + + class VolumeData : public AudioCommandData { + public: + audio_stream_type_t mStream; + float mVolume; + audio_io_handle_t mIO; + }; + + class ParametersData : public AudioCommandData { + public: + audio_io_handle_t mIO; + String8 mKeyValuePairs; + }; + + class VoiceVolumeData : public AudioCommandData { + public: + float mVolume; + }; + + class StopOutputData : public AudioCommandData { + public: + audio_io_handle_t mIO; + audio_stream_type_t mStream; + audio_session_t mSession; + }; + + class ReleaseOutputData : public AudioCommandData { + public: + audio_io_handle_t mIO; + audio_stream_type_t mStream; + audio_session_t mSession; + }; + + class CreateAudioPatchData : public AudioCommandData { + public: + struct audio_patch mPatch; + audio_patch_handle_t mHandle; + }; + + class ReleaseAudioPatchData : public AudioCommandData { + public: + audio_patch_handle_t mHandle; + }; + + class SetAudioPortConfigData : public AudioCommandData { + public: + struct audio_port_config mConfig; + }; + + Mutex mLock; + Condition mWaitWorkCV; + Vector < sp > mAudioCommands; // list of pending commands + ToneGenerator *mpToneGenerator; // the tone generator + sp mLastCommand; // last processed command (used by dump) + String8 mName; // string used by wake lock fo delayed commands + wp mService; + }; + + class AudioPolicyClient : public AudioPolicyClientInterface + { + public: + AudioPolicyClient(AudioPolicyService *service) : mAudioPolicyService(service) {} + virtual ~AudioPolicyClient() {} + + // + // Audio HW module functions + // + + // loads a HW module. + virtual audio_module_handle_t loadHwModule(const char *name); + + // + // Audio output Control functions + // + + // opens an audio output with the requested parameters. The parameter values can indicate to use the default values + // in case the audio policy manager has no specific requirements for the output being opened. + // When the function returns, the parameter values reflect the actual values used by the audio hardware output stream. + // The audio policy manager can check if the proposed parameters are suitable or not and act accordingly. + virtual status_t openOutput(audio_module_handle_t module, + audio_io_handle_t *output, + audio_config_t *config, + audio_devices_t *devices, + const String8& address, + uint32_t *latencyMs, + audio_output_flags_t flags); + // creates a special output that is duplicated to the two outputs passed as arguments. The duplication is performed by + // a special mixer thread in the AudioFlinger. + virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, audio_io_handle_t output2); + // closes the output stream + virtual status_t closeOutput(audio_io_handle_t output); + // suspends the output. When an output is suspended, the corresponding audio hardware output stream is placed in + // standby and the AudioTracks attached to the mixer thread are still processed but the output mix is discarded. + virtual status_t suspendOutput(audio_io_handle_t output); + // restores a suspended output. + virtual status_t restoreOutput(audio_io_handle_t output); + + // + // Audio input Control functions + // + + // opens an audio input + virtual audio_io_handle_t openInput(audio_module_handle_t module, + audio_io_handle_t *input, + audio_config_t *config, + audio_devices_t *devices, + const String8& address, + audio_source_t source, + audio_input_flags_t flags); + // closes an audio input + virtual status_t closeInput(audio_io_handle_t input); + // + // misc control functions + // + + // set a stream volume for a particular output. For the same user setting, a given stream type can have different volumes + // for each output (destination device) it is attached to. + virtual status_t setStreamVolume(audio_stream_type_t stream, float volume, audio_io_handle_t output, int delayMs = 0); + + // invalidate a stream type, causing a reroute to an unspecified new output + virtual status_t invalidateStream(audio_stream_type_t stream); + + // function enabling to send proprietary informations directly from audio policy manager to audio hardware interface. + virtual void setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs, int delayMs = 0); + // function enabling to receive proprietary informations directly from audio hardware interface to audio policy manager. + virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys); + + // request the playback of a tone on the specified stream: used for instance to replace notification sounds when playing + // over a telephony device during a phone call. + virtual status_t startTone(audio_policy_tone_t tone, audio_stream_type_t stream); + virtual status_t stopTone(); + + // set down link audio volume. + virtual status_t setVoiceVolume(float volume, int delayMs = 0); + + // move effect to the specified output + virtual status_t moveEffects(int session, + audio_io_handle_t srcOutput, + audio_io_handle_t dstOutput); + + /* Create a patch between several source and sink ports */ + virtual status_t createAudioPatch(const struct audio_patch *patch, + audio_patch_handle_t *handle, + int delayMs); + + /* Release a patch */ + virtual status_t releaseAudioPatch(audio_patch_handle_t handle, + int delayMs); + + /* Set audio port configuration */ + virtual status_t setAudioPortConfig(const struct audio_port_config *config, int delayMs); + + virtual void onAudioPortListUpdate(); + virtual void onAudioPatchListUpdate(); + + virtual audio_unique_id_t newAudioUniqueId(); + + private: + AudioPolicyService *mAudioPolicyService; + }; + + // --- Notification Client --- + class NotificationClient : public IBinder::DeathRecipient { + public: + NotificationClient(const sp& service, + const sp& client, + uid_t uid); + virtual ~NotificationClient(); + + void onAudioPortListUpdate(); + void onAudioPatchListUpdate(); + + // IBinder::DeathRecipient + virtual void binderDied(const wp& who); + + private: + NotificationClient(const NotificationClient&); + NotificationClient& operator = (const NotificationClient&); + + const wp mService; + const uid_t mUid; + const sp mAudioPolicyServiceClient; + }; + + // Internal dump utilities. + status_t dumpPermissionDenial(int fd); + + + mutable Mutex mLock; // prevents concurrent access to AudioPolicy manager functions changing + // device connection state or routing + sp mAudioCommandThread; // audio commands thread + sp mTonePlaybackThread; // tone playback thread + sp mOutputCommandThread; // process stop and release output + struct audio_policy_device *mpAudioPolicyDev; + struct audio_policy *mpAudioPolicy; + AudioPolicyInterface *mAudioPolicyManager; + AudioPolicyClient *mAudioPolicyClient; + + DefaultKeyedVector< uid_t, sp > mNotificationClients; + Mutex mNotificationClientsLock; // protects mNotificationClients + // Manage all effects configured in audio_effects.conf + sp mAudioPolicyEffects; + audio_mode_t mPhoneState; +}; + +}; // namespace android + +#endif // ANDROID_AUDIOPOLICYSERVICE_H -- cgit v1.1