diff options
29 files changed, 54 insertions, 4739 deletions
diff --git a/BoardConfig.mk b/BoardConfig.mk index f6ecfe5..cd6657d 100755 --- a/BoardConfig.mk +++ b/BoardConfig.mk @@ -29,8 +29,8 @@ TARGET_ARCH_VARIANT_CPU := cortex-a9 ARCH_ARM_HAVE_TLS_REGISTER := true TARGET_GLOBAL_CFLAGS += -mtune=cortex-a9 -mfpu=neon -mfloat-abi=softfp TARGET_GLOBAL_CPPFLAGS += -mtune=cortex-a9 -mfpu=neon -mfloat-abi=softfp -TARGET_BOARD_PLATFORM := smdkv310 -TARGET_BOOTLOADER_BOARD_NAME := GT-N7000 +TARGET_BOARD_PLATFORM := s5pc210 +TARGET_BOOTLOADER_BOARD_NAME := smdk4210 TARGET_BOARD_INFO_FILE := device/samsung/galaxynote/board-info.txt TARGET_NO_BOOTLOADER := true @@ -60,12 +60,12 @@ TARGET_RELEASETOOL_IMG_FROM_TARGET_SCRIPT := ./device/samsung/galaxynote/release # Graphics (Mali 400) BOARD_EGL_CFG := device/samsung/galaxynote/configs/egl.cfg USE_OPENGL_RENDERER := true + +# HWComposer BOARD_USES_HWCOMPOSER := true -BOARD_USES_LEGACY_EGL := true -COMMON_GLOBAL_CFLAGS += -DMISSING_EGL_EXTERNAL_IMAGE -DMISSING_EGL_PIXEL_FORMAT_YV12 -DMISSING_GRALLOC_BUFFERS +BOARD_USE_SECTVOUT := true # Audio -BOARD_USES_AUDIO_LEGACY := true BOARD_USE_YAMAHAPLAYER := true # Camera diff --git a/audio/Android.mk b/audio/Android.mk deleted file mode 100755 index 5026c54..0000000 --- a/audio/Android.mk +++ /dev/null @@ -1,103 +0,0 @@ -# -# Copyright (C) 2011 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. - -ifeq ($(BOARD_USES_AUDIO_LEGACY),true) -LOCAL_PATH := $(call my-dir) - -include $(CLEAR_VARS) - -LOCAL_MODULE := audio.primary.$(TARGET_BOARD_PLATFORM) -LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/hw -LOCAL_MODULE_TAGS := optional - -LOCAL_SHARED_LIBRARIES := \ - libcutils \ - libutils \ - libmedia \ - libhardware_legacy - -LOCAL_SHARED_LIBRARIES += libdl - -LOCAL_SHARED_LIBRARIES += libaudio - -ifeq ($(BOARD_FORCE_STATIC_A2DP),true) - LOCAL_SHARED_LIBRARIES += liba2dp -endif - - -LOCAL_STATIC_LIBRARIES := \ - libmedia_helper - -LOCAL_WHOLE_STATIC_LIBRARIES := \ - libaudiohw_legacy - -include $(BUILD_SHARED_LIBRARY) - -include $(CLEAR_VARS) - -LOCAL_SRC_FILES := \ - AudioPolicyManagerBase.cpp \ - AudioPolicyCompatClient.cpp \ - audio_policy_hal.cpp - -ifeq ($(AUDIO_POLICY_TEST),true) - LOCAL_CFLAGS += -DAUDIO_POLICY_TEST -endif - -ifeq ($(BOARD_HAVE_BLUETOOTH),true) - LOCAL_CFLAGS += -DWITH_A2DP -endif - -LOCAL_STATIC_LIBRARIES := libmedia_helper -LOCAL_MODULE := libaudiopolicy_legacy2 -LOCAL_MODULE_TAGS := optional - -include $(BUILD_STATIC_LIBRARY) - - -# The default audio policy, for now still implemented on top of legacy -# policy code -include $(CLEAR_VARS) - -LOCAL_SRC_FILES := \ - AudioPolicyManagerDefault.cpp - -LOCAL_SHARED_LIBRARIES := \ - libcutils \ - libutils \ - libmedia - -LOCAL_STATIC_LIBRARIES := \ - libmedia_helper - -LOCAL_WHOLE_STATIC_LIBRARIES := \ - libaudiopolicy_legacy2 - -ifeq ($(BOARD_USES_AUDIO_LEGACY),true) -LOCAL_SHARED_LIBRARIES += libaudiopolicy -endif - -LOCAL_C_INCLUDES := $(LOCAL_PATH) -LOCAL_MODULE := audio_policy.$(TARGET_BOARD_PLATFORM) -LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/hw -LOCAL_MODULE_TAGS := optional - -ifeq ($(BOARD_HAVE_BLUETOOTH),true) - LOCAL_CFLAGS += -DWITH_A2DP -endif - -include $(BUILD_SHARED_LIBRARY) - -endif ## AUDIOPOLICY diff --git a/audio/AudioPolicyCompatClient.cpp b/audio/AudioPolicyCompatClient.cpp deleted file mode 100755 index 08c31c5..0000000 --- a/audio/AudioPolicyCompatClient.cpp +++ /dev/null @@ -1,142 +0,0 @@ -/* - * Copyright (C) 2011 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 "AudioPolicyCompatClient" -//#define LOG_NDEBUG 0 - -#include <stdint.h> - -#include <hardware/hardware.h> -#include <system/audio.h> -#include <system/audio_policy.h> -#include <hardware/audio_policy.h> - -#include <hardware_legacy/AudioSystemLegacy.h> - -#include "AudioPolicyCompatClient.h" - -namespace android_audio_legacy { - -audio_io_handle_t AudioPolicyCompatClient::openOutput(uint32_t *pDevices, - uint32_t *pSamplingRate, - uint32_t *pFormat, - uint32_t *pChannels, - uint32_t *pLatencyMs, - AudioSystem::output_flags flags) -{ - return mServiceOps->open_output(mService, pDevices, pSamplingRate, pFormat, - pChannels, pLatencyMs, - (audio_policy_output_flags_t)flags); -} - -audio_io_handle_t AudioPolicyCompatClient::openDuplicateOutput(audio_io_handle_t output1, - audio_io_handle_t output2) -{ - return mServiceOps->open_duplicate_output(mService, output1, output2); -} - -status_t AudioPolicyCompatClient::closeOutput(audio_io_handle_t output) -{ - return mServiceOps->close_output(mService, output); -} - -status_t AudioPolicyCompatClient::suspendOutput(audio_io_handle_t output) -{ - return mServiceOps->suspend_output(mService, output); -} - -status_t AudioPolicyCompatClient::restoreOutput(audio_io_handle_t output) -{ - return mServiceOps->restore_output(mService, output); -} - -audio_io_handle_t AudioPolicyCompatClient::openInput(uint32_t *pDevices, - uint32_t *pSamplingRate, - uint32_t *pFormat, - uint32_t *pChannels, - uint32_t acoustics) -{ - return mServiceOps->open_input(mService, pDevices, pSamplingRate, pFormat, - pChannels, acoustics); -} - -status_t AudioPolicyCompatClient::closeInput(audio_io_handle_t input) -{ - return mServiceOps->close_input(mService, input); -} - -status_t AudioPolicyCompatClient::setStreamOutput(AudioSystem::stream_type stream, - audio_io_handle_t output) -{ - return mServiceOps->set_stream_output(mService, (audio_stream_type_t)stream, - output); -} - -status_t AudioPolicyCompatClient::moveEffects(int session, audio_io_handle_t srcOutput, - audio_io_handle_t dstOutput) -{ - return mServiceOps->move_effects(mService, session, srcOutput, dstOutput); -} - -String8 AudioPolicyCompatClient::getParameters(audio_io_handle_t ioHandle, const String8& keys) -{ - char *str; - String8 out_str8; - - str = mServiceOps->get_parameters(mService, ioHandle, keys.string()); - out_str8 = String8(str); - free(str); - - return out_str8; -} - -void AudioPolicyCompatClient::setParameters(audio_io_handle_t ioHandle, - const String8& keyValuePairs, - int delayMs) -{ - mServiceOps->set_parameters(mService, ioHandle, keyValuePairs.string(), - delayMs); -} - -status_t AudioPolicyCompatClient::setStreamVolume( - AudioSystem::stream_type stream, - float volume, - audio_io_handle_t output, - int delayMs) -{ - return mServiceOps->set_stream_volume(mService, (audio_stream_type_t)stream, - volume, output, delayMs); -} - -status_t AudioPolicyCompatClient::startTone(ToneGenerator::tone_type tone, - AudioSystem::stream_type stream) -{ - return mServiceOps->start_tone(mService, - AUDIO_POLICY_TONE_IN_CALL_NOTIFICATION, - (audio_stream_type_t)stream); -} - -status_t AudioPolicyCompatClient::stopTone() -{ - return mServiceOps->stop_tone(mService); -} - -status_t AudioPolicyCompatClient::setVoiceVolume(float volume, int delayMs) -{ - return mServiceOps->set_voice_volume(mService, volume, delayMs); -} - -}; // namespace android_audio_legacy diff --git a/audio/AudioPolicyCompatClient.h b/audio/AudioPolicyCompatClient.h deleted file mode 100755 index 4feee37..0000000 --- a/audio/AudioPolicyCompatClient.h +++ /dev/null @@ -1,79 +0,0 @@ -/* - * Copyright (C) 2011 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_AUDIOPOLICYCLIENTLEGACY_H -#define ANDROID_AUDIOPOLICYCLIENTLEGACY_H - -#include <system/audio.h> -#include <system/audio_policy.h> -#include <hardware/audio_policy.h> - -#include <hardware_legacy/AudioSystemLegacy.h> -#include <hardware_legacy/AudioPolicyInterface.h> - -/************************************/ -/* FOR BACKWARDS COMPATIBILITY ONLY */ -/************************************/ -namespace android_audio_legacy { - -class AudioPolicyCompatClient : public AudioPolicyClientInterface { -public: - AudioPolicyCompatClient(struct audio_policy_service_ops *serviceOps, - void *service) : - mServiceOps(serviceOps) , mService(service) {} - - virtual audio_io_handle_t openOutput(uint32_t *pDevices, - uint32_t *pSamplingRate, - uint32_t *pFormat, - uint32_t *pChannels, - uint32_t *pLatencyMs, - AudioSystem::output_flags flags); - virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, - audio_io_handle_t output2); - virtual status_t closeOutput(audio_io_handle_t output); - virtual status_t suspendOutput(audio_io_handle_t output); - virtual status_t restoreOutput(audio_io_handle_t output); - virtual audio_io_handle_t openInput(uint32_t *pDevices, - uint32_t *pSamplingRate, - uint32_t *pFormat, - uint32_t *pChannels, - uint32_t acoustics); - virtual status_t closeInput(audio_io_handle_t input); - virtual status_t setStreamOutput(AudioSystem::stream_type stream, audio_io_handle_t output); - virtual status_t moveEffects(int session, - audio_io_handle_t srcOutput, - audio_io_handle_t dstOutput); - - virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys); - virtual void setParameters(audio_io_handle_t ioHandle, - const String8& keyValuePairs, - int delayMs = 0); - virtual status_t setStreamVolume(AudioSystem::stream_type stream, - float volume, - audio_io_handle_t output, - int delayMs = 0); - virtual status_t startTone(ToneGenerator::tone_type tone, AudioSystem::stream_type stream); - virtual status_t stopTone(); - virtual status_t setVoiceVolume(float volume, int delayMs = 0); - -private: - struct audio_policy_service_ops* mServiceOps; - void* mService; -}; - -}; // namespace android_audio_legacy - -#endif // ANDROID_AUDIOPOLICYCLIENTLEGACY_H diff --git a/audio/AudioPolicyInterface.h b/audio/AudioPolicyInterface.h deleted file mode 100755 index 78f87da..0000000 --- a/audio/AudioPolicyInterface.h +++ /dev/null @@ -1,238 +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_AUDIOPOLICYINTERFACE_H -#define ANDROID_AUDIOPOLICYINTERFACE_H - -#include <media/AudioSystem.h> -#include <media/ToneGenerator.h> -#include <utils/String8.h> - -#include <hardware_legacy/AudioSystemLegacy.h> - -namespace android_audio_legacy { - using android::Vector; - using android::String8; - using android::ToneGenerator; - -// ---------------------------------------------------------------------------- - -// The AudioPolicyInterface and AudioPolicyClientInterface classes define the communication interfaces -// between the platform specific audio policy manager and Android generic audio policy manager. -// The platform specific audio policy manager must implement methods of the AudioPolicyInterface class. -// This implementation makes use of the AudioPolicyClientInterface to control the activity and -// configuration of audio input and output streams. -// -// The platform specific audio policy manager is in charge of the audio routing and volume control -// policies for a given platform. -// The main roles of this module are: -// - keep track of current system state (removable device connections, phone state, user requests...). -// System state changes and user actions are notified to audio policy manager with methods of the AudioPolicyInterface. -// - process getOutput() queries received when AudioTrack objects are created: Those queries -// return a handler on an output that has been selected, configured and opened by the audio policy manager and that -// must be used by the AudioTrack when registering to the AudioFlinger with the createTrack() method. -// When the AudioTrack object is released, a putOutput() query is received and the audio policy manager can decide -// to close or reconfigure the output depending on other streams using this output and current system state. -// - similarly process getInput() and putInput() queries received from AudioRecord objects and configure audio inputs. -// - process volume control requests: the stream volume is converted from an index value (received from UI) to a float value -// applicable to each output as a function of platform specific settings and current output route (destination device). It -// also make sure that streams are not muted if not allowed (e.g. camera shutter sound in some countries). -// -// The platform specific audio policy manager is provided as a shared library by platform vendors (as for libaudio.so) -// and is linked with libaudioflinger.so - - -// Audio Policy Manager Interface -class AudioPolicyInterface -{ - -public: - virtual ~AudioPolicyInterface() {} - // - // configuration functions - // - - // indicate a change in device connection status - virtual status_t setDeviceConnectionState(AudioSystem::audio_devices device, - AudioSystem::device_connection_state state, - const char *device_address) = 0; - // retreive a device connection status - virtual AudioSystem::device_connection_state getDeviceConnectionState(AudioSystem::audio_devices device, - const char *device_address) = 0; - // indicate a change in phone state. Valid phones states are defined by AudioSystem::audio_mode - virtual void setPhoneState(int state) = 0; - // indicate a change in ringer mode - virtual void setRingerMode(uint32_t mode, uint32_t mask) = 0; - // force using a specific device category for the specified usage - virtual void setForceUse(AudioSystem::force_use usage, AudioSystem::forced_config config) = 0; - // retreive current device category forced for a given usage - virtual AudioSystem::forced_config getForceUse(AudioSystem::force_use usage) = 0; - // set a system property (e.g. camera sound always audible) - virtual void setSystemProperty(const char* property, const char* value) = 0; - // check proper initialization - virtual status_t initCheck() = 0; - - // - // Audio routing query functions - // - - // request an output appriate for playback of the supplied stream type and parameters - virtual audio_io_handle_t getOutput(AudioSystem::stream_type stream, - uint32_t samplingRate = 0, - uint32_t format = AudioSystem::FORMAT_DEFAULT, - uint32_t channels = 0, - AudioSystem::output_flags flags = AudioSystem::OUTPUT_FLAG_INDIRECT) = 0; - // indicates to the audio policy manager that the output starts being used by corresponding stream. - virtual status_t startOutput(audio_io_handle_t output, - AudioSystem::stream_type stream, - int session = 0) = 0; - // indicates to the audio policy manager that the output stops being used by corresponding stream. - virtual status_t stopOutput(audio_io_handle_t output, - AudioSystem::stream_type stream, - int session = 0) = 0; - // releases the output. - virtual void releaseOutput(audio_io_handle_t output) = 0; - - // request an input appriate for record from the supplied device with supplied parameters. - virtual audio_io_handle_t getInput(int inputSource, - uint32_t samplingRate = 0, - uint32_t Format = AudioSystem::FORMAT_DEFAULT, - uint32_t channels = 0, - AudioSystem::audio_in_acoustics acoustics = (AudioSystem::audio_in_acoustics)0) = 0; - // indicates to the audio policy manager that the input starts being used. - virtual status_t startInput(audio_io_handle_t input) = 0; - // indicates to the audio policy manager that the input stops being used. - virtual status_t stopInput(audio_io_handle_t input) = 0; - // releases the input. - virtual void releaseInput(audio_io_handle_t input) = 0; - - // - // volume control functions - // - - // initialises stream volume conversion parameters by specifying volume index range. - virtual void initStreamVolume(AudioSystem::stream_type stream, - int indexMin, - int indexMax) = 0; - - // sets the new stream volume at a level corresponding to the supplied index - virtual status_t setStreamVolumeIndex(AudioSystem::stream_type stream, int index) = 0; - // retreive current volume index for the specified stream - virtual status_t getStreamVolumeIndex(AudioSystem::stream_type stream, int *index) = 0; - - // return the strategy corresponding to a given stream type - virtual uint32_t getStrategyForStream(AudioSystem::stream_type stream) = 0; - - // return the enabled output devices for the given stream type - virtual uint32_t getDevicesForStream(AudioSystem::stream_type stream) = 0; - - // Audio effect management - virtual audio_io_handle_t getOutputForEffect(effect_descriptor_t *desc) = 0; - virtual status_t registerEffect(effect_descriptor_t *desc, - audio_io_handle_t io, - uint32_t strategy, - int session, - int id) = 0; - virtual status_t unregisterEffect(int id) = 0; - - virtual bool isStreamActive(int stream, uint32_t inPastMs = 0) const = 0; - - //dump state - virtual status_t dump(int fd) = 0; -}; - - -// Audio Policy client Interface -class AudioPolicyClientInterface -{ -public: - virtual ~AudioPolicyClientInterface() {} - - // - // 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 audio_io_handle_t openOutput(uint32_t *pDevices, - uint32_t *pSamplingRate, - uint32_t *pFormat, - uint32_t *pChannels, - uint32_t *pLatencyMs, - AudioSystem::output_flags flags) = 0; - // 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) = 0; - // closes the output stream - virtual status_t closeOutput(audio_io_handle_t output) = 0; - // 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) = 0; - // restores a suspended output. - virtual status_t restoreOutput(audio_io_handle_t output) = 0; - - // - // Audio input Control functions - // - - // opens an audio input - virtual audio_io_handle_t openInput(uint32_t *pDevices, - uint32_t *pSamplingRate, - uint32_t *pFormat, - uint32_t *pChannels, - uint32_t acoustics) = 0; - // closes an audio input - virtual status_t closeInput(audio_io_handle_t input) = 0; - // - // 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(AudioSystem::stream_type stream, float volume, audio_io_handle_t output, int delayMs = 0) = 0; - - // reroute a given stream type to the specified output - virtual status_t setStreamOutput(AudioSystem::stream_type stream, audio_io_handle_t output) = 0; - - // 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) = 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) = 0; - - // 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(ToneGenerator::tone_type tone, AudioSystem::stream_type stream) = 0; - virtual status_t stopTone() = 0; - - // set down link audio volume. - virtual status_t setVoiceVolume(float volume, int delayMs = 0) = 0; - - // move effect to the specified output - virtual status_t moveEffects(int session, - audio_io_handle_t srcOutput, - audio_io_handle_t dstOutput) = 0; - -}; - -extern "C" AudioPolicyInterface* createAudioPolicyManager(AudioPolicyClientInterface *clientInterface); -extern "C" void destroyAudioPolicyManager(AudioPolicyInterface *interface); - - -}; // namespace android - -#endif // ANDROID_AUDIOPOLICYINTERFACE_H diff --git a/audio/AudioPolicyManagerBase.cpp b/audio/AudioPolicyManagerBase.cpp deleted file mode 100755 index ac31ad4..0000000 --- a/audio/AudioPolicyManagerBase.cpp +++ /dev/null @@ -1,2434 +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 "AudioPolicyManagerBase" -//#define LOG_NDEBUG 0 -#include <utils/Log.h> -#include <hardware_legacy/AudioPolicyManagerBase.h> -#include <hardware/audio_effect.h> -#include <math.h> - -namespace android_audio_legacy { - -// ---------------------------------------------------------------------------- -// AudioPolicyInterface implementation -// ---------------------------------------------------------------------------- - - -status_t AudioPolicyManagerBase::setDeviceConnectionState(AudioSystem::audio_devices device, - AudioSystem::device_connection_state state, - const char *device_address) -{ - - LOGV("setDeviceConnectionState() device: %x, state %d, address %s", device, state, device_address); - - // connect/disconnect only 1 device at a time - if (AudioSystem::popCount(device) != 1) return BAD_VALUE; - - if (strlen(device_address) >= MAX_DEVICE_ADDRESS_LEN) { - LOGE("setDeviceConnectionState() invalid address: %s", device_address); - return BAD_VALUE; - } - - // handle output devices - if (AudioSystem::isOutputDevice(device)) { - -#ifndef WITH_A2DP - if (AudioSystem::isA2dpDevice(device)) { - LOGE("setDeviceConnectionState() invalid device: %x", device); - return BAD_VALUE; - } -#endif - - switch (state) - { - // handle output device connection - case AudioSystem::DEVICE_STATE_AVAILABLE: - if (mAvailableOutputDevices & device) { - LOGW("setDeviceConnectionState() device already connected: %x", device); - return INVALID_OPERATION; - } - LOGV("setDeviceConnectionState() connecting device %x", device); - - // register new device as available - mAvailableOutputDevices |= device; - -#ifdef WITH_A2DP - // handle A2DP device connection - if (AudioSystem::isA2dpDevice(device)) { - status_t status = handleA2dpConnection(device, device_address); - if (status != NO_ERROR) { - mAvailableOutputDevices &= ~device; - return status; - } - } else -#endif - { - if (AudioSystem::isBluetoothScoDevice(device)) { - LOGV("setDeviceConnectionState() BT SCO device, address %s", device_address); - // keep track of SCO device address - mScoDeviceAddress = String8(device_address, MAX_DEVICE_ADDRESS_LEN); - } - } - break; - // handle output device disconnection - case AudioSystem::DEVICE_STATE_UNAVAILABLE: { - if (!(mAvailableOutputDevices & device)) { - LOGW("setDeviceConnectionState() device not connected: %x", device); - return INVALID_OPERATION; - } - - - LOGV("setDeviceConnectionState() disconnecting device %x", device); - // remove device from available output devices - mAvailableOutputDevices &= ~device; - -#ifdef WITH_A2DP - // handle A2DP device disconnection - if (AudioSystem::isA2dpDevice(device)) { - status_t status = handleA2dpDisconnection(device, device_address); - if (status != NO_ERROR) { - mAvailableOutputDevices |= device; - return status; - } - } else -#endif - { - if (AudioSystem::isBluetoothScoDevice(device)) { - mScoDeviceAddress = ""; - } - } - } break; - - default: - LOGE("setDeviceConnectionState() invalid state: %x", state); - return BAD_VALUE; - } - - // request routing change if necessary - uint32_t newDevice = getNewDevice(mHardwareOutput, false); -#ifdef WITH_A2DP - checkA2dpSuspend(); - checkOutputForAllStrategies(); - // A2DP outputs must be closed after checkOutputForAllStrategies() is executed - if (state == AudioSystem::DEVICE_STATE_UNAVAILABLE && AudioSystem::isA2dpDevice(device)) { - closeA2dpOutputs(); - } -#endif - updateDeviceForStrategy(); - setOutputDevice(mHardwareOutput, newDevice); - - if (device == AudioSystem::DEVICE_OUT_WIRED_HEADSET) { - device = AudioSystem::DEVICE_IN_WIRED_HEADSET; - } else if (device == AudioSystem::DEVICE_OUT_BLUETOOTH_SCO || - device == AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET || - device == AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT) { - device = AudioSystem::DEVICE_IN_BLUETOOTH_SCO_HEADSET; - } else { - return NO_ERROR; - } - } - // handle input devices - if (AudioSystem::isInputDevice(device)) { - - switch (state) - { - // handle input device connection - case AudioSystem::DEVICE_STATE_AVAILABLE: { - if (mAvailableInputDevices & device) { - LOGW("setDeviceConnectionState() device already connected: %d", device); - return INVALID_OPERATION; - } - mAvailableInputDevices |= device; - } - break; - - // handle input device disconnection - case AudioSystem::DEVICE_STATE_UNAVAILABLE: { - if (!(mAvailableInputDevices & device)) { - LOGW("setDeviceConnectionState() device not connected: %d", device); - return INVALID_OPERATION; - } - mAvailableInputDevices &= ~device; - } break; - - default: - LOGE("setDeviceConnectionState() invalid state: %x", state); - return BAD_VALUE; - } - - audio_io_handle_t activeInput = getActiveInput(); - if (activeInput != 0) { - AudioInputDescriptor *inputDesc = mInputs.valueFor(activeInput); - uint32_t newDevice = getDeviceForInputSource(inputDesc->mInputSource); - if (newDevice != inputDesc->mDevice) { - LOGV("setDeviceConnectionState() changing device from %x to %x for input %d", - inputDesc->mDevice, newDevice, activeInput); - inputDesc->mDevice = newDevice; - AudioParameter param = AudioParameter(); - param.addInt(String8(AudioParameter::keyRouting), (int)newDevice); - mpClientInterface->setParameters(activeInput, param.toString()); - } - } - - return NO_ERROR; - } - - LOGW("setDeviceConnectionState() invalid device: %x", device); - return BAD_VALUE; -} - -AudioSystem::device_connection_state AudioPolicyManagerBase::getDeviceConnectionState(AudioSystem::audio_devices device, - const char *device_address) -{ - AudioSystem::device_connection_state state = AudioSystem::DEVICE_STATE_UNAVAILABLE; - String8 address = String8(device_address); - if (AudioSystem::isOutputDevice(device)) { - if (device & mAvailableOutputDevices) { -#ifdef WITH_A2DP - if (AudioSystem::isA2dpDevice(device) && - address != "" && mA2dpDeviceAddress != address) { - return state; - } -#endif - if (AudioSystem::isBluetoothScoDevice(device) && - address != "" && mScoDeviceAddress != address) { - return state; - } - state = AudioSystem::DEVICE_STATE_AVAILABLE; - } - } else if (AudioSystem::isInputDevice(device)) { - if (device & mAvailableInputDevices) { - state = AudioSystem::DEVICE_STATE_AVAILABLE; - } - } - - return state; -} - -void AudioPolicyManagerBase::setPhoneState(int state) -{ - LOGV("setPhoneState() state %d", state); - uint32_t newDevice = 0; - if (state < 0 || state >= AudioSystem::NUM_MODES) { - LOGW("setPhoneState() invalid state %d", state); - return; - } - - if (state == mPhoneState ) { - LOGW("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()) { - LOGV("setPhoneState() in call state management: new state is %d", state); - for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) { - handleIncallSonification(stream, false, true); - } - } - - // 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)) { - LOGV(" Entering call in setPhoneState()"); - // force routing command to audio hardware when starting a call - // even if no device change is needed - force = true; - } else if (isStateInCall(oldState) && !isStateInCall(state)) { - LOGV(" Exiting call in setPhoneState()"); - // force routing command to audio hardware when exiting a call - // even if no device change is needed - force = true; - } else if (isStateInCall(state) && (state != oldState)) { - LOGV(" 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 - newDevice = getNewDevice(mHardwareOutput, false); -#ifdef WITH_A2DP - checkA2dpSuspend(); - checkOutputForAllStrategies(); -#endif - updateDeviceForStrategy(); - - AudioOutputDescriptor *hwOutputDesc = mOutputs.valueFor(mHardwareOutput); - - // force routing command to audio hardware when ending call - // even if no device change is needed - if (isStateInCall(oldState) && newDevice == 0) { - newDevice = hwOutputDesc->device(); - } - - // when changing from ring tone to in call mode, mute the ringing tone - // immediately and delay the route change to avoid sending the ring tone - // tail into the earpiece or headset. - int delayMs = 0; - if (isStateInCall(state) && oldState == AudioSystem::MODE_RINGTONE) { - // delay the device change command by twice the output latency to have some margin - // and be sure that audio buffers not yet affected by the mute are out when - // we actually apply the route change - delayMs = hwOutputDesc->mLatency*2; - setStreamMute(AudioSystem::RING, true, mHardwareOutput); - } - - // change routing is necessary - setOutputDevice(mHardwareOutput, newDevice, force, delayMs); - - // if entering in call state, handle special case of active streams - // pertaining to sonification strategy see handleIncallSonification() - if (isStateInCall(state)) { - LOGV("setPhoneState() in call state management: new state is %d", state); - // unmute the ringing tone after a sufficient delay if it was muted before - // setting output device above - if (oldState == AudioSystem::MODE_RINGTONE) { - setStreamMute(AudioSystem::RING, false, mHardwareOutput, MUTE_TIME_MS); - } - for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) { - handleIncallSonification(stream, true, true); - } - } - - // Flag that ringtone volume must be limited to music volume until we exit MODE_RINGTONE - if (state == AudioSystem::MODE_RINGTONE && - isStreamActive(AudioSystem::MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY)) { - mLimitRingtoneVolume = true; - } else { - mLimitRingtoneVolume = false; - } -} - -void AudioPolicyManagerBase::setRingerMode(uint32_t mode, uint32_t mask) -{ - LOGV("setRingerMode() mode %x, mask %x", mode, mask); - - mRingerMode = mode; -} - -void AudioPolicyManagerBase::setForceUse(AudioSystem::force_use usage, AudioSystem::forced_config config) -{ - LOGV("setForceUse() usage %d, config %d, mPhoneState %d", usage, config, mPhoneState); - - bool forceVolumeReeval = false; - switch(usage) { - case AudioSystem::FOR_COMMUNICATION: - if (config != AudioSystem::FORCE_SPEAKER && config != AudioSystem::FORCE_BT_SCO && - config != AudioSystem::FORCE_NONE) { - LOGW("setForceUse() invalid config %d for FOR_COMMUNICATION", config); - return; - } - forceVolumeReeval = true; - mForceUse[usage] = config; - break; - case AudioSystem::FOR_MEDIA: - if (config != AudioSystem::FORCE_HEADPHONES && config != AudioSystem::FORCE_BT_A2DP && - config != AudioSystem::FORCE_WIRED_ACCESSORY && - config != AudioSystem::FORCE_ANALOG_DOCK && - config != AudioSystem::FORCE_DIGITAL_DOCK && config != AudioSystem::FORCE_NONE) { - LOGW("setForceUse() invalid config %d for FOR_MEDIA", config); - return; - } - mForceUse[usage] = config; - break; - case AudioSystem::FOR_RECORD: - if (config != AudioSystem::FORCE_BT_SCO && config != AudioSystem::FORCE_WIRED_ACCESSORY && - config != AudioSystem::FORCE_NONE) { - LOGW("setForceUse() invalid config %d for FOR_RECORD", config); - return; - } - mForceUse[usage] = config; - break; - case AudioSystem::FOR_DOCK: - if (config != AudioSystem::FORCE_NONE && config != AudioSystem::FORCE_BT_CAR_DOCK && - config != AudioSystem::FORCE_BT_DESK_DOCK && - config != AudioSystem::FORCE_WIRED_ACCESSORY && - config != AudioSystem::FORCE_ANALOG_DOCK && - config != AudioSystem::FORCE_DIGITAL_DOCK) { - LOGW("setForceUse() invalid config %d for FOR_DOCK", config); - } - forceVolumeReeval = true; - mForceUse[usage] = config; - break; - default: - LOGW("setForceUse() invalid usage %d", usage); - break; - } - - // check for device and output changes triggered by new phone state - uint32_t newDevice = getNewDevice(mHardwareOutput, false); -#ifdef WITH_A2DP - checkA2dpSuspend(); - checkOutputForAllStrategies(); -#endif - updateDeviceForStrategy(); - setOutputDevice(mHardwareOutput, newDevice); - if (forceVolumeReeval) { - applyStreamVolumes(mHardwareOutput, newDevice, 0, true); - } - - audio_io_handle_t activeInput = getActiveInput(); - if (activeInput != 0) { - AudioInputDescriptor *inputDesc = mInputs.valueFor(activeInput); - newDevice = getDeviceForInputSource(inputDesc->mInputSource); - if (newDevice != inputDesc->mDevice) { - LOGV("setForceUse() changing device from %x to %x for input %d", - inputDesc->mDevice, newDevice, activeInput); - inputDesc->mDevice = newDevice; - AudioParameter param = AudioParameter(); - param.addInt(String8(AudioParameter::keyRouting), (int)newDevice); - mpClientInterface->setParameters(activeInput, param.toString()); - } - } - -} - -AudioSystem::forced_config AudioPolicyManagerBase::getForceUse(AudioSystem::force_use usage) -{ - return mForceUse[usage]; -} - -void AudioPolicyManagerBase::setSystemProperty(const char* property, const char* value) -{ - LOGV("setSystemProperty() property %s, value %s", property, value); - if (strcmp(property, "ro.camera.sound.forced") == 0) { - if (atoi(value)) { - LOGV("ENFORCED_AUDIBLE cannot be muted"); - mStreams[AudioSystem::ENFORCED_AUDIBLE].mCanBeMuted = false; - } else { - LOGV("ENFORCED_AUDIBLE can be muted"); - mStreams[AudioSystem::ENFORCED_AUDIBLE].mCanBeMuted = true; - } - } -} - -audio_io_handle_t AudioPolicyManagerBase::getOutput(AudioSystem::stream_type stream, - uint32_t samplingRate, - uint32_t format, - uint32_t channels, - AudioSystem::output_flags flags) -{ - audio_io_handle_t output = 0; - uint32_t latency = 0; - routing_strategy strategy = getStrategy((AudioSystem::stream_type)stream); - uint32_t device = getDeviceForStrategy(strategy); - LOGV("getOutput() stream %d, samplingRate %d, format %d, channels %x, flags %x", stream, samplingRate, format, channels, flags); - -#ifdef AUDIO_POLICY_TEST - if (mCurOutput != 0) { - LOGV("getOutput() test output mCurOutput %d, samplingRate %d, format %d, channels %x, mDirectOutput %d", - mCurOutput, mTestSamplingRate, mTestFormat, mTestChannels, mDirectOutput); - - if (mTestOutputs[mCurOutput] == 0) { - LOGV("getOutput() opening test output"); - AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(); - outputDesc->mDevice = mTestDevice; - outputDesc->mSamplingRate = mTestSamplingRate; - outputDesc->mFormat = mTestFormat; - outputDesc->mChannels = mTestChannels; - outputDesc->mLatency = mTestLatencyMs; - outputDesc->mFlags = (AudioSystem::output_flags)(mDirectOutput ? AudioSystem::OUTPUT_FLAG_DIRECT : 0); - outputDesc->mRefCount[stream] = 0; - mTestOutputs[mCurOutput] = mpClientInterface->openOutput(&outputDesc->mDevice, - &outputDesc->mSamplingRate, - &outputDesc->mFormat, - &outputDesc->mChannels, - &outputDesc->mLatency, - outputDesc->mFlags); - if (mTestOutputs[mCurOutput]) { - 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 - if (needsDirectOuput(stream, samplingRate, format, channels, flags, device)) { - - LOGV("getOutput() opening direct output device %x", device); - AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(); - outputDesc->mDevice = device; - outputDesc->mSamplingRate = samplingRate; - outputDesc->mFormat = format; - outputDesc->mChannels = channels; - outputDesc->mLatency = 0; - outputDesc->mFlags = (AudioSystem::output_flags)(flags | AudioSystem::OUTPUT_FLAG_DIRECT); - outputDesc->mRefCount[stream] = 0; - outputDesc->mStopTime[stream] = 0; - output = mpClientInterface->openOutput(&outputDesc->mDevice, - &outputDesc->mSamplingRate, - &outputDesc->mFormat, - &outputDesc->mChannels, - &outputDesc->mLatency, - outputDesc->mFlags); - - // only accept an output with the requeted parameters - if (output == 0 || - (samplingRate != 0 && samplingRate != outputDesc->mSamplingRate) || - (format != 0 && format != outputDesc->mFormat) || - (channels != 0 && channels != outputDesc->mChannels)) { - LOGV("getOutput() failed opening direct output: samplingRate %d, format %d, channels %d", - samplingRate, format, channels); - if (output != 0) { - mpClientInterface->closeOutput(output); - } - delete outputDesc; - return 0; - } - addOutput(output, outputDesc); - return output; - } - - if (channels != 0 && channels != AudioSystem::CHANNEL_OUT_MONO && - channels != AudioSystem::CHANNEL_OUT_STEREO) { - return 0; - } - // open a non direct output - - // get which output is suitable for the specified stream. The actual routing change will happen - // when startOutput() will be called - uint32_t a2dpDevice = device & AudioSystem::DEVICE_OUT_ALL_A2DP; - if (AudioSystem::popCount((AudioSystem::audio_devices)device) == 2) { -#ifdef WITH_A2DP - if (a2dpUsedForSonification() && a2dpDevice != 0) { - // if playing on 2 devices among which one is A2DP, use duplicated output - LOGV("getOutput() using duplicated output"); - LOGW_IF((mA2dpOutput == 0), "getOutput() A2DP device in multiple %x selected but A2DP output not opened", device); - output = mDuplicatedOutput; - } else -#endif - { - // if playing on 2 devices among which none is A2DP, use hardware output - output = mHardwareOutput; - } - LOGV("getOutput() using output %d for 2 devices %x", output, device); - } else { -#ifdef WITH_A2DP - if (a2dpDevice != 0) { - // if playing on A2DP device, use a2dp output - LOGW_IF((mA2dpOutput == 0), "getOutput() A2DP device %x selected but A2DP output not opened", device); - output = mA2dpOutput; - } else -#endif - { - // if playing on not A2DP device, use hardware output - output = mHardwareOutput; - } - } - - - LOGW_IF((output ==0), "getOutput() could not find output for stream %d, samplingRate %d, format %d, channels %x, flags %x", - stream, samplingRate, format, channels, flags); - - return output; -} - -status_t AudioPolicyManagerBase::startOutput(audio_io_handle_t output, - AudioSystem::stream_type stream, - int session) -{ - LOGV("startOutput() output %d, stream %d, session %d", output, stream, session); - ssize_t index = mOutputs.indexOfKey(output); - if (index < 0) { - LOGW("startOutput() unknow output %d", output); - return BAD_VALUE; - } - - AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index); - routing_strategy strategy = getStrategy((AudioSystem::stream_type)stream); - -#ifdef WITH_A2DP - if (mA2dpOutput != 0 && !a2dpUsedForSonification() && - (strategy == STRATEGY_SONIFICATION || strategy == STRATEGY_ENFORCED_AUDIBLE)) { - setStrategyMute(STRATEGY_MEDIA, true, mA2dpOutput); - } -#endif - - // incremenent 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 - // necassary for a correct control of hardware output routing by startOutput() and stopOutput() - outputDesc->changeRefCount(stream, 1); - - uint32_t prevDevice = outputDesc->mDevice; - setOutputDevice(output, getNewDevice(output)); - - // 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].mIndexCur, output, outputDesc->device()); - - // FIXME: need a delay to make sure that audio path switches to speaker before sound - // starts. Should be platform specific? - if (stream == AudioSystem::ENFORCED_AUDIBLE && - prevDevice != outputDesc->mDevice) { - usleep(outputDesc->mLatency*4*1000); - } - - return NO_ERROR; -} - -status_t AudioPolicyManagerBase::stopOutput(audio_io_handle_t output, - AudioSystem::stream_type stream, - int session) -{ - LOGV("stopOutput() output %d, stream %d, session %d", output, stream, session); - ssize_t index = mOutputs.indexOfKey(output); - if (index < 0) { - LOGW("stopOutput() unknow output %d", output); - return BAD_VALUE; - } - - AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index); - routing_strategy strategy = getStrategy((AudioSystem::stream_type)stream); - - // 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() - outputDesc->mStopTime[stream] = systemTime(); - - setOutputDevice(output, getNewDevice(output), false, outputDesc->mLatency*2); - -#ifdef WITH_A2DP - if (mA2dpOutput != 0 && !a2dpUsedForSonification() && - (strategy == STRATEGY_SONIFICATION || strategy == STRATEGY_ENFORCED_AUDIBLE)) { - setStrategyMute(STRATEGY_MEDIA, - false, - mA2dpOutput, - mOutputs.valueFor(mHardwareOutput)->mLatency*2); - } -#endif - if (output != mHardwareOutput) { - setOutputDevice(mHardwareOutput, getNewDevice(mHardwareOutput), true); - } - return NO_ERROR; - } else { - LOGW("stopOutput() refcount is already 0 for output %d", output); - return INVALID_OPERATION; - } -} - -void AudioPolicyManagerBase::releaseOutput(audio_io_handle_t output) -{ - LOGV("releaseOutput() %d", output); - ssize_t index = mOutputs.indexOfKey(output); - if (index < 0) { - LOGW("releaseOutput() releasing unknown output %d", output); - return; - } - -#ifdef AUDIO_POLICY_TEST - int testIndex = testOutputIndex(output); - if (testIndex != 0) { - AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index); - if (outputDesc->refCount() == 0) { - mpClientInterface->closeOutput(output); - delete mOutputs.valueAt(index); - mOutputs.removeItem(output); - mTestOutputs[testIndex] = 0; - } - return; - } -#endif //AUDIO_POLICY_TEST - - if (mOutputs.valueAt(index)->mFlags & AudioSystem::OUTPUT_FLAG_DIRECT) { - mpClientInterface->closeOutput(output); - delete mOutputs.valueAt(index); - mOutputs.removeItem(output); - } -} - -audio_io_handle_t AudioPolicyManagerBase::getInput(int inputSource, - uint32_t samplingRate, - uint32_t format, - uint32_t channels, - AudioSystem::audio_in_acoustics acoustics) -{ - audio_io_handle_t input = 0; - uint32_t device = getDeviceForInputSource(inputSource); - - LOGV("getInput() inputSource %d, samplingRate %d, format %d, channels %x, acoustics %x", inputSource, samplingRate, format, channels, acoustics); - - if (device == 0) { - return 0; - } - - // adapt channel selection to input source - switch(inputSource) { - case AUDIO_SOURCE_VOICE_UPLINK: - channels = AudioSystem::CHANNEL_IN_VOICE_UPLINK; - break; - case AUDIO_SOURCE_VOICE_DOWNLINK: - channels = AudioSystem::CHANNEL_IN_VOICE_DNLINK; - break; - case AUDIO_SOURCE_VOICE_CALL: - channels = (AudioSystem::CHANNEL_IN_VOICE_UPLINK | AudioSystem::CHANNEL_IN_VOICE_DNLINK); - break; - default: - break; - } - - AudioInputDescriptor *inputDesc = new AudioInputDescriptor(); - - inputDesc->mInputSource = inputSource; - inputDesc->mDevice = device; - inputDesc->mSamplingRate = samplingRate; - inputDesc->mFormat = format; - inputDesc->mChannels = channels; - inputDesc->mAcoustics = acoustics; - inputDesc->mRefCount = 0; - input = mpClientInterface->openInput(&inputDesc->mDevice, - &inputDesc->mSamplingRate, - &inputDesc->mFormat, - &inputDesc->mChannels, - inputDesc->mAcoustics); - - // only accept input with the exact requested set of parameters - if (input == 0 || - (samplingRate != inputDesc->mSamplingRate) || - (format != inputDesc->mFormat) || - (channels != inputDesc->mChannels)) { - LOGV("getInput() failed opening input: samplingRate %d, format %d, channels %d", - samplingRate, format, channels); - if (input != 0) { - mpClientInterface->closeInput(input); - } - delete inputDesc; - return 0; - } - mInputs.add(input, inputDesc); - return input; -} - -status_t AudioPolicyManagerBase::startInput(audio_io_handle_t input) -{ - LOGV("startInput() input %d", input); - ssize_t index = mInputs.indexOfKey(input); - if (index < 0) { - LOGW("startInput() unknow input %d", input); - return BAD_VALUE; - } - AudioInputDescriptor *inputDesc = mInputs.valueAt(index); - -#ifdef AUDIO_POLICY_TEST - if (mTestInput == 0) -#endif //AUDIO_POLICY_TEST - { - // refuse 2 active AudioRecord clients at the same time - if (getActiveInput() != 0) { - LOGW("startInput() input %d failed: other input already started", input); - return INVALID_OPERATION; - } - } - - AudioParameter param = AudioParameter(); - param.addInt(String8(AudioParameter::keyRouting), (int)inputDesc->mDevice); - - param.addInt(String8(AudioParameter::keyInputSource), (int)inputDesc->mInputSource); - LOGV("AudioPolicyManager::startInput() input source = %d", inputDesc->mInputSource); - - mpClientInterface->setParameters(input, param.toString()); - - inputDesc->mRefCount = 1; - return NO_ERROR; -} - -status_t AudioPolicyManagerBase::stopInput(audio_io_handle_t input) -{ - LOGV("stopInput() input %d", input); - ssize_t index = mInputs.indexOfKey(input); - if (index < 0) { - LOGW("stopInput() unknow input %d", input); - return BAD_VALUE; - } - AudioInputDescriptor *inputDesc = mInputs.valueAt(index); - - if (inputDesc->mRefCount == 0) { - LOGW("stopInput() input %d already stopped", input); - return INVALID_OPERATION; - } else { - AudioParameter param = AudioParameter(); - param.addInt(String8(AudioParameter::keyRouting), 0); - mpClientInterface->setParameters(input, param.toString()); - inputDesc->mRefCount = 0; - return NO_ERROR; - } -} - -void AudioPolicyManagerBase::releaseInput(audio_io_handle_t input) -{ - LOGV("releaseInput() %d", input); - ssize_t index = mInputs.indexOfKey(input); - if (index < 0) { - LOGW("releaseInput() releasing unknown input %d", input); - return; - } - mpClientInterface->closeInput(input); - delete mInputs.valueAt(index); - mInputs.removeItem(input); - LOGV("releaseInput() exit"); -} - -void AudioPolicyManagerBase::initStreamVolume(AudioSystem::stream_type stream, - int indexMin, - int indexMax) -{ - LOGV("initStreamVolume() stream %d, min %d, max %d", stream , indexMin, indexMax); - if (indexMin < 0 || indexMin >= indexMax) { - LOGW("initStreamVolume() invalid index limits for stream %d, min %d, max %d", stream , indexMin, indexMax); - return; - } - mStreams[stream].mIndexMin = indexMin; - mStreams[stream].mIndexMax = indexMax; -} - -status_t AudioPolicyManagerBase::setStreamVolumeIndex(AudioSystem::stream_type stream, int index) -{ - - if ((index < mStreams[stream].mIndexMin) || (index > mStreams[stream].mIndexMax)) { - return BAD_VALUE; - } - - // Force max volume if stream cannot be muted - if (!mStreams[stream].mCanBeMuted) index = mStreams[stream].mIndexMax; - - LOGV("setStreamVolumeIndex() stream %d, index %d", stream, index); - mStreams[stream].mIndexCur = index; - - // compute and apply stream volume on all outputs according to connected device - status_t status = NO_ERROR; - for (size_t i = 0; i < mOutputs.size(); i++) { - status_t volStatus = checkAndSetVolume(stream, index, mOutputs.keyAt(i), mOutputs.valueAt(i)->device()); - if (volStatus != NO_ERROR) { - status = volStatus; - } - } - return status; -} - -status_t AudioPolicyManagerBase::getStreamVolumeIndex(AudioSystem::stream_type stream, int *index) -{ - if (index == 0) { - return BAD_VALUE; - } - LOGV("getStreamVolumeIndex() stream %d", stream); - *index = mStreams[stream].mIndexCur; - return NO_ERROR; -} - -audio_io_handle_t AudioPolicyManagerBase::getOutputForEffect(effect_descriptor_t *desc) -{ - LOGV("getOutputForEffect()"); - // apply simple rule where global effects are attached to the same output as MUSIC streams - return getOutput(AudioSystem::MUSIC); -} - -status_t AudioPolicyManagerBase::registerEffect(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) { - LOGW("registerEffect() unknown io %d", io); - return INVALID_OPERATION; - } - } - - if (mTotalEffectsMemory + desc->memoryUsage > getMaxEffectsMemory()) { - LOGW("registerEffect() memory limit exceeded for Fx %s, Memory %d KB", - desc->name, desc->memoryUsage); - return INVALID_OPERATION; - } - mTotalEffectsMemory += desc->memoryUsage; - LOGV("registerEffect() effect %s, io %d, strategy %d session %d id %d", - desc->name, io, strategy, session, id); - LOGV("registerEffect() memory %d, total memory %d", desc->memoryUsage, mTotalEffectsMemory); - - EffectDescriptor *pDesc = new EffectDescriptor(); - memcpy (&pDesc->mDesc, desc, sizeof(effect_descriptor_t)); - pDesc->mIo = io; - pDesc->mStrategy = (routing_strategy)strategy; - pDesc->mSession = session; - pDesc->mEnabled = false; - - mEffects.add(id, pDesc); - - return NO_ERROR; -} - -status_t AudioPolicyManagerBase::unregisterEffect(int id) -{ - ssize_t index = mEffects.indexOfKey(id); - if (index < 0) { - LOGW("unregisterEffect() unknown effect ID %d", id); - return INVALID_OPERATION; - } - - EffectDescriptor *pDesc = mEffects.valueAt(index); - - setEffectEnabled(pDesc, false); - - if (mTotalEffectsMemory < pDesc->mDesc.memoryUsage) { - LOGW("unregisterEffect() memory %d too big for total %d", - pDesc->mDesc.memoryUsage, mTotalEffectsMemory); - pDesc->mDesc.memoryUsage = mTotalEffectsMemory; - } - mTotalEffectsMemory -= pDesc->mDesc.memoryUsage; - LOGV("unregisterEffect() effect %s, ID %d, memory %d total memory %d", - pDesc->mDesc.name, id, pDesc->mDesc.memoryUsage, mTotalEffectsMemory); - - mEffects.removeItem(id); - delete pDesc; - - return NO_ERROR; -} - -status_t AudioPolicyManagerBase::setEffectEnabled(int id, bool enabled) -{ - ssize_t index = mEffects.indexOfKey(id); - if (index < 0) { - LOGW("unregisterEffect() unknown effect ID %d", id); - return INVALID_OPERATION; - } - - return setEffectEnabled(mEffects.valueAt(index), enabled); -} - -status_t AudioPolicyManagerBase::setEffectEnabled(EffectDescriptor *pDesc, bool enabled) -{ - if (enabled == pDesc->mEnabled) { - LOGV("setEffectEnabled(%s) effect already %s", - enabled?"true":"false", enabled?"enabled":"disabled"); - return INVALID_OPERATION; - } - - if (enabled) { - if (mTotalEffectsCpuLoad + pDesc->mDesc.cpuLoad > getMaxEffectsCpuLoad()) { - LOGW("setEffectEnabled(true) CPU Load limit exceeded for Fx %s, CPU %f MIPS", - pDesc->mDesc.name, (float)pDesc->mDesc.cpuLoad/10); - return INVALID_OPERATION; - } - mTotalEffectsCpuLoad += pDesc->mDesc.cpuLoad; - LOGV("setEffectEnabled(true) total CPU %d", mTotalEffectsCpuLoad); - } else { - if (mTotalEffectsCpuLoad < pDesc->mDesc.cpuLoad) { - LOGW("setEffectEnabled(false) CPU load %d too high for total %d", - pDesc->mDesc.cpuLoad, mTotalEffectsCpuLoad); - pDesc->mDesc.cpuLoad = mTotalEffectsCpuLoad; - } - mTotalEffectsCpuLoad -= pDesc->mDesc.cpuLoad; - LOGV("setEffectEnabled(false) total CPU %d", mTotalEffectsCpuLoad); - } - pDesc->mEnabled = enabled; - return NO_ERROR; -} - -bool AudioPolicyManagerBase::isStreamActive(int stream, uint32_t inPastMs) const -{ - nsecs_t sysTime = systemTime(); - for (size_t i = 0; i < mOutputs.size(); i++) { - if (mOutputs.valueAt(i)->mRefCount[stream] != 0 || - ns2ms(sysTime - mOutputs.valueAt(i)->mStopTime[stream]) < inPastMs) { - return true; - } - } - return false; -} - - -status_t AudioPolicyManagerBase::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, " Hardware Output: %d\n", mHardwareOutput); - result.append(buffer); -#ifdef WITH_A2DP - snprintf(buffer, SIZE, " A2DP Output: %d\n", mA2dpOutput); - result.append(buffer); - snprintf(buffer, SIZE, " Duplicated Output: %d\n", mDuplicatedOutput); - result.append(buffer); - snprintf(buffer, SIZE, " A2DP device address: %s\n", mA2dpDeviceAddress.string()); - result.append(buffer); -#endif - snprintf(buffer, SIZE, " SCO device address: %s\n", mScoDeviceAddress.string()); - result.append(buffer); - snprintf(buffer, SIZE, " Output devices: %08x\n", mAvailableOutputDevices); - result.append(buffer); - snprintf(buffer, SIZE, " Input devices: %08x\n", mAvailableInputDevices); - result.append(buffer); - snprintf(buffer, SIZE, " Phone state: %d\n", mPhoneState); - result.append(buffer); - snprintf(buffer, SIZE, " Ringer mode: %d\n", mRingerMode); - result.append(buffer); - snprintf(buffer, SIZE, " Force use for communications %d\n", mForceUse[AudioSystem::FOR_COMMUNICATION]); - result.append(buffer); - snprintf(buffer, SIZE, " Force use for media %d\n", mForceUse[AudioSystem::FOR_MEDIA]); - result.append(buffer); - snprintf(buffer, SIZE, " Force use for record %d\n", mForceUse[AudioSystem::FOR_RECORD]); - result.append(buffer); - snprintf(buffer, SIZE, " Force use for dock %d\n", mForceUse[AudioSystem::FOR_DOCK]); - result.append(buffer); - write(fd, result.string(), result.size()); - - 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 Index Min Index Max Index Cur Can be muted\n"); - write(fd, buffer, strlen(buffer)); - for (size_t i = 0; i < AudioSystem::NUM_STREAM_TYPES; i++) { - snprintf(buffer, SIZE, " %02d", i); - mStreams[i].dump(buffer + 3, SIZE); - write(fd, buffer, strlen(buffer)); - } - - 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); - } - - - return NO_ERROR; -} - -// ---------------------------------------------------------------------------- -// AudioPolicyManagerBase -// ---------------------------------------------------------------------------- - -AudioPolicyManagerBase::AudioPolicyManagerBase(AudioPolicyClientInterface *clientInterface) - : -#ifdef AUDIO_POLICY_TEST - Thread(false), -#endif //AUDIO_POLICY_TEST - mPhoneState(AudioSystem::MODE_NORMAL), mRingerMode(0), - mLimitRingtoneVolume(false), mLastVoiceVolume(-1.0f), - mTotalEffectsCpuLoad(0), mTotalEffectsMemory(0), - mA2dpSuspended(false) -{ - mpClientInterface = clientInterface; - - for (int i = 0; i < AudioSystem::NUM_FORCE_USE; i++) { - mForceUse[i] = AudioSystem::FORCE_NONE; - } - - initializeVolumeCurves(); - - // devices available by default are speaker, ear piece and microphone - mAvailableOutputDevices = AudioSystem::DEVICE_OUT_EARPIECE | - AudioSystem::DEVICE_OUT_SPEAKER; - mAvailableInputDevices = AudioSystem::DEVICE_IN_BUILTIN_MIC; - -#ifdef WITH_A2DP - mA2dpOutput = 0; - mDuplicatedOutput = 0; - mA2dpDeviceAddress = String8(""); -#endif - mScoDeviceAddress = String8(""); - - // open hardware output - AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(); - outputDesc->mDevice = (uint32_t)AudioSystem::DEVICE_OUT_SPEAKER; - mHardwareOutput = mpClientInterface->openOutput(&outputDesc->mDevice, - &outputDesc->mSamplingRate, - &outputDesc->mFormat, - &outputDesc->mChannels, - &outputDesc->mLatency, - outputDesc->mFlags); - - if (mHardwareOutput == 0) { - LOGE("Failed to initialize hardware output stream, samplingRate: %d, format %d, channels %d", - outputDesc->mSamplingRate, outputDesc->mFormat, outputDesc->mChannels); - } else { - addOutput(mHardwareOutput, outputDesc); - setOutputDevice(mHardwareOutput, (uint32_t)AudioSystem::DEVICE_OUT_SPEAKER, true); - //TODO: configure audio effect output stage here - } - - updateDeviceForStrategy(); -#ifdef AUDIO_POLICY_TEST - if (mHardwareOutput != 0) { - AudioParameter outputCmd = AudioParameter(); - outputCmd.addInt(String8("set_id"), 0); - mpClientInterface->setParameters(mHardwareOutput, outputCmd.toString()); - - mTestDevice = AudioSystem::DEVICE_OUT_SPEAKER; - mTestSamplingRate = 44100; - mTestFormat = AudioSystem::PCM_16_BIT; - mTestChannels = AudioSystem::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 -} - -AudioPolicyManagerBase::~AudioPolicyManagerBase() -{ -#ifdef AUDIO_POLICY_TEST - exit(); -#endif //AUDIO_POLICY_TEST - for (size_t i = 0; i < mOutputs.size(); i++) { - mpClientInterface->closeOutput(mOutputs.keyAt(i)); - delete mOutputs.valueAt(i); - } - mOutputs.clear(); - for (size_t i = 0; i < mInputs.size(); i++) { - mpClientInterface->closeInput(mInputs.keyAt(i)); - delete mInputs.valueAt(i); - } - mInputs.clear(); -} - -status_t AudioPolicyManagerBase::initCheck() -{ - return (mHardwareOutput == 0) ? NO_INIT : NO_ERROR; -} - -#ifdef AUDIO_POLICY_TEST -bool AudioPolicyManagerBase::threadLoop() -{ - LOGV("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) { - LOGV("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 = AudioSystem::INVALID_FORMAT; - if (value == "PCM 16 bits") { - format = AudioSystem::PCM_16_BIT; - } else if (value == "PCM 8 bits") { - format = AudioSystem::PCM_8_BIT; - } else if (value == "Compressed MP3") { - format = AudioSystem::MP3; - } - if (format != AudioSystem::INVALID_FORMAT) { - 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 = AudioSystem::CHANNEL_OUT_STEREO; - } else if (value == "Channels Mono") { - channels = AudioSystem::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")); - - mpClientInterface->closeOutput(mHardwareOutput); - delete mOutputs.valueFor(mHardwareOutput); - mOutputs.removeItem(mHardwareOutput); - - AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(); - outputDesc->mDevice = (uint32_t)AudioSystem::DEVICE_OUT_SPEAKER; - mHardwareOutput = mpClientInterface->openOutput(&outputDesc->mDevice, - &outputDesc->mSamplingRate, - &outputDesc->mFormat, - &outputDesc->mChannels, - &outputDesc->mLatency, - outputDesc->mFlags); - if (mHardwareOutput == 0) { - LOGE("Failed to reopen hardware output stream, samplingRate: %d, format %d, channels %d", - outputDesc->mSamplingRate, outputDesc->mFormat, outputDesc->mChannels); - } else { - AudioParameter outputCmd = AudioParameter(); - outputCmd.addInt(String8("set_id"), 0); - mpClientInterface->setParameters(mHardwareOutput, outputCmd.toString()); - addOutput(mHardwareOutput, outputDesc); - } - } - - - mpClientInterface->setParameters(0, String8("test_cmd_policy=")); - } - } - return false; -} - -void AudioPolicyManagerBase::exit() -{ - { - AutoMutex _l(mLock); - requestExit(); - mWaitWorkCV.signal(); - } - requestExitAndWait(); -} - -int AudioPolicyManagerBase::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 AudioPolicyManagerBase::addOutput(audio_io_handle_t id, AudioOutputDescriptor *outputDesc) -{ - outputDesc->mId = id; - mOutputs.add(id, outputDesc); -} - - -#ifdef WITH_A2DP -status_t AudioPolicyManagerBase::handleA2dpConnection(AudioSystem::audio_devices device, - const char *device_address) -{ - // when an A2DP device is connected, open an A2DP and a duplicated output - LOGV("opening A2DP output for device %s", device_address); - AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(); - outputDesc->mDevice = device; - mA2dpOutput = mpClientInterface->openOutput(&outputDesc->mDevice, - &outputDesc->mSamplingRate, - &outputDesc->mFormat, - &outputDesc->mChannels, - &outputDesc->mLatency, - outputDesc->mFlags); - if (mA2dpOutput) { - // add A2DP output descriptor - addOutput(mA2dpOutput, outputDesc); - - //TODO: configure audio effect output stage here - - // set initial stream volume for A2DP device - applyStreamVolumes(mA2dpOutput, device); - if (a2dpUsedForSonification()) { - mDuplicatedOutput = mpClientInterface->openDuplicateOutput(mA2dpOutput, mHardwareOutput); - } - if (mDuplicatedOutput != 0 || - !a2dpUsedForSonification()) { - // If both A2DP and duplicated outputs are open, send device address to A2DP hardware - // interface - AudioParameter param; - param.add(String8("a2dp_sink_address"), String8(device_address)); - mpClientInterface->setParameters(mA2dpOutput, param.toString()); - mA2dpDeviceAddress = String8(device_address, MAX_DEVICE_ADDRESS_LEN); - - if (a2dpUsedForSonification()) { - // add duplicated output descriptor - AudioOutputDescriptor *dupOutputDesc = new AudioOutputDescriptor(); - dupOutputDesc->mOutput1 = mOutputs.valueFor(mHardwareOutput); - dupOutputDesc->mOutput2 = mOutputs.valueFor(mA2dpOutput); - dupOutputDesc->mSamplingRate = outputDesc->mSamplingRate; - dupOutputDesc->mFormat = outputDesc->mFormat; - dupOutputDesc->mChannels = outputDesc->mChannels; - dupOutputDesc->mLatency = outputDesc->mLatency; - addOutput(mDuplicatedOutput, dupOutputDesc); - applyStreamVolumes(mDuplicatedOutput, device); - } - } else { - LOGW("getOutput() could not open duplicated output for %d and %d", - mHardwareOutput, mA2dpOutput); - mpClientInterface->closeOutput(mA2dpOutput); - mOutputs.removeItem(mA2dpOutput); - mA2dpOutput = 0; - delete outputDesc; - return NO_INIT; - } - } else { - LOGW("setDeviceConnectionState() could not open A2DP output for device %x", device); - delete outputDesc; - return NO_INIT; - } - AudioOutputDescriptor *hwOutputDesc = mOutputs.valueFor(mHardwareOutput); - - if (!a2dpUsedForSonification()) { - // mute music on A2DP output if a notification or ringtone is playing - uint32_t refCount = hwOutputDesc->strategyRefCount(STRATEGY_SONIFICATION); - refCount += hwOutputDesc->strategyRefCount(STRATEGY_ENFORCED_AUDIBLE); - for (uint32_t i = 0; i < refCount; i++) { - setStrategyMute(STRATEGY_MEDIA, true, mA2dpOutput); - } - } - - mA2dpSuspended = false; - - return NO_ERROR; -} - -status_t AudioPolicyManagerBase::handleA2dpDisconnection(AudioSystem::audio_devices device, - const char *device_address) -{ - if (mA2dpOutput == 0) { - LOGW("setDeviceConnectionState() disconnecting A2DP and no A2DP output!"); - return INVALID_OPERATION; - } - - if (mA2dpDeviceAddress != device_address) { - LOGW("setDeviceConnectionState() disconnecting unknow A2DP sink address %s", device_address); - return INVALID_OPERATION; - } - - // mute media strategy to avoid outputting sound on hardware output while music stream - // is switched from A2DP output and before music is paused by music application - setStrategyMute(STRATEGY_MEDIA, true, mHardwareOutput); - setStrategyMute(STRATEGY_MEDIA, false, mHardwareOutput, MUTE_TIME_MS); - - if (!a2dpUsedForSonification()) { - // unmute music on A2DP output if a notification or ringtone is playing - uint32_t refCount = mOutputs.valueFor(mHardwareOutput)->strategyRefCount(STRATEGY_SONIFICATION); - refCount += mOutputs.valueFor(mHardwareOutput)->strategyRefCount(STRATEGY_ENFORCED_AUDIBLE); - for (uint32_t i = 0; i < refCount; i++) { - setStrategyMute(STRATEGY_MEDIA, false, mA2dpOutput); - } - } - mA2dpDeviceAddress = ""; - mA2dpSuspended = false; - return NO_ERROR; -} - -void AudioPolicyManagerBase::closeA2dpOutputs() -{ - - LOGV("setDeviceConnectionState() closing A2DP and duplicated output!"); - - if (mDuplicatedOutput != 0) { - AudioOutputDescriptor *dupOutputDesc = mOutputs.valueFor(mDuplicatedOutput); - AudioOutputDescriptor *hwOutputDesc = mOutputs.valueFor(mHardwareOutput); - // As all active tracks on duplicated output will be deleted, - // and as they were also referenced on hardware output, the reference - // count for their stream type must be adjusted accordingly on - // hardware output. - for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) { - int refCount = dupOutputDesc->mRefCount[i]; - hwOutputDesc->changeRefCount((AudioSystem::stream_type)i,-refCount); - } - - mpClientInterface->closeOutput(mDuplicatedOutput); - delete mOutputs.valueFor(mDuplicatedOutput); - mOutputs.removeItem(mDuplicatedOutput); - mDuplicatedOutput = 0; - } - if (mA2dpOutput != 0) { - AudioParameter param; - param.add(String8("closing"), String8("true")); - mpClientInterface->setParameters(mA2dpOutput, param.toString()); - - mpClientInterface->closeOutput(mA2dpOutput); - delete mOutputs.valueFor(mA2dpOutput); - mOutputs.removeItem(mA2dpOutput); - mA2dpOutput = 0; - } -} - -void AudioPolicyManagerBase::checkOutputForStrategy(routing_strategy strategy) -{ - uint32_t prevDevice = getDeviceForStrategy(strategy); - uint32_t curDevice = getDeviceForStrategy(strategy, false); - bool a2dpWasUsed = AudioSystem::isA2dpDevice((AudioSystem::audio_devices)(prevDevice & ~AudioSystem::DEVICE_OUT_SPEAKER)); - bool a2dpIsUsed = AudioSystem::isA2dpDevice((AudioSystem::audio_devices)(curDevice & ~AudioSystem::DEVICE_OUT_SPEAKER)); - audio_io_handle_t srcOutput = 0; - audio_io_handle_t dstOutput = 0; - - if (a2dpWasUsed && !a2dpIsUsed) { - bool dupUsed = a2dpUsedForSonification() && a2dpWasUsed && (AudioSystem::popCount(prevDevice) == 2); - dstOutput = mHardwareOutput; - if (dupUsed) { - LOGV("checkOutputForStrategy() moving strategy %d from duplicated", strategy); - srcOutput = mDuplicatedOutput; - } else { - LOGV("checkOutputForStrategy() moving strategy %d from a2dp", strategy); - srcOutput = mA2dpOutput; - } - } - if (a2dpIsUsed && !a2dpWasUsed) { - bool dupUsed = a2dpUsedForSonification() && a2dpIsUsed && (AudioSystem::popCount(curDevice) == 2); - srcOutput = mHardwareOutput; - if (dupUsed) { - LOGV("checkOutputForStrategy() moving strategy %d to duplicated", strategy); - dstOutput = mDuplicatedOutput; - } else { - LOGV("checkOutputForStrategy() moving strategy %d to a2dp", strategy); - dstOutput = mA2dpOutput; - } - } - - if (srcOutput != 0 && dstOutput != 0) { - // Move effects associated to this strategy from previous output to new output - for (size_t i = 0; i < mEffects.size(); i++) { - EffectDescriptor *desc = mEffects.valueAt(i); - if (desc->mSession != AudioSystem::SESSION_OUTPUT_STAGE && - desc->mStrategy == strategy && - desc->mIo == srcOutput) { - LOGV("checkOutputForStrategy() moving effect %d to output %d", mEffects.keyAt(i), dstOutput); - mpClientInterface->moveEffects(desc->mSession, srcOutput, dstOutput); - desc->mIo = dstOutput; - } - } - // Move tracks associated to this strategy from previous output to new output - for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) { - if (getStrategy((AudioSystem::stream_type)i) == strategy) { - mpClientInterface->setStreamOutput((AudioSystem::stream_type)i, dstOutput); - } - } - } -} - -void AudioPolicyManagerBase::checkOutputForAllStrategies() -{ - checkOutputForStrategy(STRATEGY_ENFORCED_AUDIBLE); - checkOutputForStrategy(STRATEGY_PHONE); - checkOutputForStrategy(STRATEGY_SONIFICATION); - checkOutputForStrategy(STRATEGY_MEDIA); - checkOutputForStrategy(STRATEGY_DTMF); -} - -void AudioPolicyManagerBase::checkA2dpSuspend() -{ - // 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 (mA2dpOutput == 0) { - return; - } - - if (mA2dpSuspended) { - if (((mScoDeviceAddress == "") || - ((mForceUse[AudioSystem::FOR_COMMUNICATION] != AudioSystem::FORCE_BT_SCO) && - (mForceUse[AudioSystem::FOR_RECORD] != AudioSystem::FORCE_BT_SCO))) && - ((mPhoneState != AudioSystem::MODE_IN_CALL) && - (mPhoneState != AudioSystem::MODE_RINGTONE))) { - - mpClientInterface->restoreOutput(mA2dpOutput); - mA2dpSuspended = false; - } - } else { - if (((mScoDeviceAddress != "") && - ((mForceUse[AudioSystem::FOR_COMMUNICATION] == AudioSystem::FORCE_BT_SCO) || - (mForceUse[AudioSystem::FOR_RECORD] == AudioSystem::FORCE_BT_SCO))) || - ((mPhoneState == AudioSystem::MODE_IN_CALL) || - (mPhoneState == AudioSystem::MODE_RINGTONE))) { - - mpClientInterface->suspendOutput(mA2dpOutput); - mA2dpSuspended = true; - } - } -} - - -#endif - -uint32_t AudioPolicyManagerBase::getNewDevice(audio_io_handle_t output, bool fromCache) -{ - uint32_t device = 0; - - AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output); - // check the following by order of priority to request a routing change if necessary: - // 1: the strategy enforced audible is active 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 sonification is active on the output: - // use device for strategy sonification - // 4: the strategy media is active on the output: - // use device for strategy media - // 5: the strategy DTMF is active on the output: - // use device for strategy DTMF - if (outputDesc->isUsedByStrategy(STRATEGY_ENFORCED_AUDIBLE)) { - device = getDeviceForStrategy(STRATEGY_ENFORCED_AUDIBLE, fromCache); - } else if (isInCall() || - outputDesc->isUsedByStrategy(STRATEGY_PHONE)) { - device = getDeviceForStrategy(STRATEGY_PHONE, fromCache); - } else if (outputDesc->isUsedByStrategy(STRATEGY_SONIFICATION)) { - device = getDeviceForStrategy(STRATEGY_SONIFICATION, fromCache); - } else if (outputDesc->isUsedByStrategy(STRATEGY_MEDIA)) { - device = getDeviceForStrategy(STRATEGY_MEDIA, fromCache); - } else if (outputDesc->isUsedByStrategy(STRATEGY_DTMF)) { - device = getDeviceForStrategy(STRATEGY_DTMF, fromCache); - } - - LOGV("getNewDevice() selected device %x", device); - return device; -} - -uint32_t AudioPolicyManagerBase::getStrategyForStream(AudioSystem::stream_type stream) { - return (uint32_t)getStrategy(stream); -} - -uint32_t AudioPolicyManagerBase::getDevicesForStream(AudioSystem::stream_type stream) { - uint32_t devices; - // By checking the range of stream before calling getStrategy, we avoid - // getStrategy's behavior for invalid streams. getStrategy would do a LOGE - // and then return STRATEGY_MEDIA, but we want to return the empty set. - if (stream < (AudioSystem::stream_type) 0 || stream >= AudioSystem::NUM_STREAM_TYPES) { - devices = 0; - } else { - AudioPolicyManagerBase::routing_strategy strategy = getStrategy(stream); - devices = getDeviceForStrategy(strategy, true); - } - return devices; -} - -AudioPolicyManagerBase::routing_strategy AudioPolicyManagerBase::getStrategy( - AudioSystem::stream_type stream) { - // stream to strategy mapping - switch (stream) { - case AudioSystem::VOICE_CALL: - case AudioSystem::BLUETOOTH_SCO: - return STRATEGY_PHONE; - case AudioSystem::RING: - case AudioSystem::NOTIFICATION: - case AudioSystem::ALARM: - return STRATEGY_SONIFICATION; - case AudioSystem::DTMF: - return STRATEGY_DTMF; - default: - LOGE("unknown stream type"); - case AudioSystem::SYSTEM: - // NOTE: SYSTEM stream uses MEDIA strategy because muting music and switching outputs - // while key clicks are played produces a poor result - case AudioSystem::TTS: - case AudioSystem::MUSIC: - return STRATEGY_MEDIA; - case AudioSystem::ENFORCED_AUDIBLE: - return STRATEGY_ENFORCED_AUDIBLE; - } -} - -uint32_t AudioPolicyManagerBase::getDeviceForStrategy(routing_strategy strategy, bool fromCache) -{ - uint32_t device = 0; - - if (fromCache) { - LOGV("getDeviceForStrategy() from cache strategy %d, device %x", strategy, mDeviceForStrategy[strategy]); - return mDeviceForStrategy[strategy]; - } - - switch (strategy) { - case STRATEGY_DTMF: - if (!isInCall()) { - // when off call, DTMF strategy follows the same rules as MEDIA strategy - device = getDeviceForStrategy(STRATEGY_MEDIA, false); - break; - } - // when in call, DTMF and PHONE strategies follow the same rules - // FALL THROUGH - - case STRATEGY_PHONE: - // for phone strategy, we first consider the forced use and then the available devices by order - // of priority - switch (mForceUse[AudioSystem::FOR_COMMUNICATION]) { - case AudioSystem::FORCE_BT_SCO: - if (!isInCall() || strategy != STRATEGY_DTMF) { - device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT; - if (device) break; - } - device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET; - if (device) break; - device = mAvailableOutputDevices & AudioSystem::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 - device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_WIRED_HEADPHONE; - if (device) break; - device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_WIRED_HEADSET; - if (device) break; -#ifdef WITH_A2DP - // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to A2DP - if (!isInCall() && !mA2dpSuspended) { - device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP; - if (device) break; - device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES; - if (device) break; - } -#endif - device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_DGTL_DOCK_HEADSET; - if (device) break; - device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_AUX_DIGITAL; - if (device) break; - device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_ANLG_DOCK_HEADSET; - if (device) break; - device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_EARPIECE; - if (device == 0) { - LOGE("getDeviceForStrategy() earpiece device not found"); - } - break; - - case AudioSystem::FORCE_SPEAKER: -#ifdef WITH_A2DP - // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to - // A2DP speaker when forcing to speaker output - if (!isInCall() && !mA2dpSuspended) { - device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER; - if (device) break; - } -#endif - device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_DGTL_DOCK_HEADSET; - if (device) break; - device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_AUX_DIGITAL; - if (device) break; - device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_ANLG_DOCK_HEADSET; - if (device) break; - device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_SPEAKER; - if (device == 0) { - LOGE("getDeviceForStrategy() speaker device not found"); - } - 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); - 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 - - device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_SPEAKER; - if (device == 0) { - LOGE("getDeviceForStrategy() speaker device not found"); - } - // The second device used for sonification is the same as the device used by media strategy - // FALL THROUGH - - case STRATEGY_MEDIA: { - uint32_t device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_WIRED_HEADPHONE; - if (device2 == 0) { - device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_WIRED_HEADSET; - } -#ifdef WITH_A2DP - if ((mA2dpOutput != 0) && !mA2dpSuspended && - (strategy == STRATEGY_MEDIA || a2dpUsedForSonification())) { - if (device2 == 0) { - device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP; - } - if (device2 == 0) { - device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES; - } - if (device2 == 0) { - device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER; - } - } -#endif - if (device2 == 0) { - device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_DGTL_DOCK_HEADSET; - } - if (device2 == 0) { - device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_AUX_DIGITAL; - } - if (device2 == 0) { - device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_ANLG_DOCK_HEADSET; - } - if (device2 == 0) { - device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_SPEAKER; - } - - // device is DEVICE_OUT_SPEAKER if we come from case STRATEGY_SONIFICATION or - // STRATEGY_ENFORCED_AUDIBLE, 0 otherwise - device |= device2; - if (device == 0) { - LOGE("getDeviceForStrategy() speaker device not found"); - } - } break; - - default: - LOGW("getDeviceForStrategy() unknown strategy: %d", strategy); - break; - } - - LOGV("getDeviceForStrategy() strategy %d, device %x", strategy, device); - return device; -} - -void AudioPolicyManagerBase::updateDeviceForStrategy() -{ - for (int i = 0; i < NUM_STRATEGIES; i++) { - mDeviceForStrategy[i] = getDeviceForStrategy((routing_strategy)i, false); - } -} - -void AudioPolicyManagerBase::setOutputDevice(audio_io_handle_t output, uint32_t device, bool force, int delayMs) -{ - LOGV("setOutputDevice() output %d device %x delayMs %d", output, device, delayMs); - AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output); - - - if (outputDesc->isDuplicated()) { - setOutputDevice(outputDesc->mOutput1->mId, device, force, delayMs); - setOutputDevice(outputDesc->mOutput2->mId, device, force, delayMs); - return; - } -#ifdef WITH_A2DP - // filter devices according to output selected - if (output == mA2dpOutput) { - device &= AudioSystem::DEVICE_OUT_ALL_A2DP; - } else { - device &= ~AudioSystem::DEVICE_OUT_ALL_A2DP; - } -#endif - - uint32_t prevDevice = (uint32_t)outputDesc->device(); - // Do not change the routing if: - // - the requestede device is 0 - // - the requested device is the same as current device and force is not specified. - // Doing this check here allows the caller to call setOutputDevice() without conditions - if ((device == 0 || device == prevDevice) && !force) { - LOGV("setOutputDevice() setting same device %x or null device for output %d", device, output); - return; - } - - outputDesc->mDevice = device; - // mute media streams if both speaker and headset are selected - if (output == mHardwareOutput && AudioSystem::popCount(device) == 2) { - setStrategyMute(STRATEGY_MEDIA, true, output); - // wait for the PCM output buffers to empty before proceeding with the rest of the command - // FIXME: increased delay due to larger buffers used for low power audio mode. - // remove when low power audio is controlled by policy manager. - usleep(outputDesc->mLatency*8*1000); - } - - // do the routing - AudioParameter param = AudioParameter(); - param.addInt(String8(AudioParameter::keyRouting), (int)device); - mpClientInterface->setParameters(mHardwareOutput, param.toString(), delayMs); - // update stream volumes according to new device - applyStreamVolumes(output, device, delayMs); - - // if changing from a combined headset + speaker route, unmute media streams - if (output == mHardwareOutput && AudioSystem::popCount(prevDevice) == 2) { - setStrategyMute(STRATEGY_MEDIA, false, output, delayMs); - } -} - -uint32_t AudioPolicyManagerBase::getDeviceForInputSource(int inputSource) -{ - uint32_t device; - - switch(inputSource) { - case AUDIO_SOURCE_DEFAULT: - case AUDIO_SOURCE_MIC: - case AUDIO_SOURCE_VOICE_RECOGNITION: - case AUDIO_SOURCE_VOICE_COMMUNICATION: - if (mForceUse[AudioSystem::FOR_RECORD] == AudioSystem::FORCE_BT_SCO && - mAvailableInputDevices & AudioSystem::DEVICE_IN_BLUETOOTH_SCO_HEADSET) { - device = AudioSystem::DEVICE_IN_BLUETOOTH_SCO_HEADSET; - } else if (mAvailableInputDevices & AudioSystem::DEVICE_IN_WIRED_HEADSET) { - device = AudioSystem::DEVICE_IN_WIRED_HEADSET; - } else { - device = AudioSystem::DEVICE_IN_BUILTIN_MIC; - } - break; - case AUDIO_SOURCE_CAMCORDER: - if (hasBackMicrophone()) { - device = AudioSystem::DEVICE_IN_BACK_MIC; - } else { - device = AudioSystem::DEVICE_IN_BUILTIN_MIC; - } - break; - case AUDIO_SOURCE_VOICE_UPLINK: - case AUDIO_SOURCE_VOICE_DOWNLINK: - case AUDIO_SOURCE_VOICE_CALL: - device = AudioSystem::DEVICE_IN_VOICE_CALL; - break; - default: - LOGW("getDeviceForInputSource() invalid input source %d", inputSource); - device = 0; - break; - } - LOGV("getDeviceForInputSource()input source %d, device %08x", inputSource, device); - return device; -} - -audio_io_handle_t AudioPolicyManagerBase::getActiveInput() -{ - for (size_t i = 0; i < mInputs.size(); i++) { - if (mInputs.valueAt(i)->mRefCount > 0) { - return mInputs.keyAt(i); - } - } - return 0; -} - - -AudioPolicyManagerBase::device_category AudioPolicyManagerBase::getDeviceCategory(uint32_t device) -{ - if (device == 0) { - // this happens when forcing a route update and no track is active on an output. - // In this case the returned category is not important. - return DEVICE_CATEGORY_SPEAKER; - } - - if (AudioSystem::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. - if (device & AUDIO_DEVICE_OUT_SPEAKER) - return DEVICE_CATEGORY_SPEAKER; - - device &= AUDIO_DEVICE_OUT_ALL_A2DP; - } - - LOGW_IF(AudioSystem::popCount(device) != 1, - "getDeviceCategory() invalid device combination: %08x", - device); - - switch(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_SPEAKER: - case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT: - case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER: - default: - return DEVICE_CATEGORY_SPEAKER; - } -} - -float AudioPolicyManagerBase::volIndexToAmpl(uint32_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 ) - - LOGV("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 AudioPolicyManagerBase::VolumeCurvePoint - AudioPolicyManagerBase::sDefaultVolumeCurve[AudioPolicyManagerBase::VOLCNT] = { - {1, -49.5f}, {33, -33.5f}, {66, -17.0f}, {100, 0.0f} -}; - -const AudioPolicyManagerBase::VolumeCurvePoint - AudioPolicyManagerBase::sDefaultMediaVolumeCurve[AudioPolicyManagerBase::VOLCNT] = { - {1, -58.0f}, {20, -40.0f}, {60, -17.0f}, {100, 0.0f} -}; - -const AudioPolicyManagerBase::VolumeCurvePoint - AudioPolicyManagerBase::sSpeakerMediaVolumeCurve[AudioPolicyManagerBase::VOLCNT] = { - {1, -56.0f}, {20, -34.0f}, {60, -11.0f}, {100, 0.0f} -}; - -const AudioPolicyManagerBase::VolumeCurvePoint - AudioPolicyManagerBase::sSpeakerSonificationVolumeCurve[AudioPolicyManagerBase::VOLCNT] = { - {1, -29.7f}, {33, -20.1f}, {66, -10.2f}, {100, 0.0f} -}; - - -const AudioPolicyManagerBase::VolumeCurvePoint - *AudioPolicyManagerBase::sVolumeProfiles[AudioPolicyManagerBase::NUM_STRATEGIES] - [AudioPolicyManagerBase::DEVICE_CATEGORY_CNT] = { - { // STRATEGY_MEDIA - sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_HEADSET - sSpeakerMediaVolumeCurve, // DEVICE_CATEGORY_SPEAKER - sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EARPIECE - }, - { // STRATEGY_PHONE - sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET - sDefaultVolumeCurve, // DEVICE_CATEGORY_SPEAKER - sDefaultVolumeCurve // DEVICE_CATEGORY_EARPIECE - }, - { // STRATEGY_SONIFICATION - sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET - sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER - sDefaultVolumeCurve // DEVICE_CATEGORY_EARPIECE - }, - { // STRATEGY_DTMF - sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET - sDefaultVolumeCurve, // DEVICE_CATEGORY_SPEAKER - sDefaultVolumeCurve // DEVICE_CATEGORY_EARPIECE - }, - { // STRATEGY_ENFORCED_AUDIBLE - sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET - sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER - sDefaultVolumeCurve // DEVICE_CATEGORY_EARPIECE - }, -}; - -void AudioPolicyManagerBase::initializeVolumeCurves() -{ - for (int i = 0; i < AudioSystem::NUM_STREAM_TYPES; i++) { - for (int j = 0; j < DEVICE_CATEGORY_CNT; j++) { - mStreams[i].mVolumeCurve[j] = - sVolumeProfiles[getStrategy((AudioSystem::stream_type)i)][j]; - } - } -} - -float AudioPolicyManagerBase::computeVolume(int stream, int index, audio_io_handle_t output, uint32_t device) -{ - float volume = 1.0; - AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output); - StreamDescriptor &streamDesc = mStreams[stream]; - - if (device == 0) { - device = outputDesc->device(); - } - - // if volume is not 0 (not muted), force media volume to max on digital output - if (stream == AudioSystem::MUSIC && - index != mStreams[stream].mIndexMin && - (device == AudioSystem::DEVICE_OUT_AUX_DIGITAL || - device == AudioSystem::DEVICE_OUT_DGTL_DOCK_HEADSET)) { - return 1.0; - } - - 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. - if ((device & - (AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP | - AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES | - AudioSystem::DEVICE_OUT_WIRED_HEADSET | - AudioSystem::DEVICE_OUT_WIRED_HEADPHONE)) && - ((getStrategy((AudioSystem::stream_type)stream) == STRATEGY_SONIFICATION) || - (stream == AudioSystem::SYSTEM)) && - 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 (outputDesc->mRefCount[AudioSystem::MUSIC] || mLimitRingtoneVolume) { - float musicVol = computeVolume(AudioSystem::MUSIC, mStreams[AudioSystem::MUSIC].mIndexCur, output, device); - float minVol = (musicVol > SONIFICATION_HEADSET_VOLUME_MIN) ? musicVol : SONIFICATION_HEADSET_VOLUME_MIN; - if (volume > minVol) { - volume = minVol; - LOGV("computeVolume limiting volume to %f musicVol %f", minVol, musicVol); - } - } - } - - return volume; -} - -status_t AudioPolicyManagerBase::checkAndSetVolume(int stream, int index, audio_io_handle_t output, uint32_t device, int delayMs, bool force) -{ - - // do not change actual stream volume if the stream is muted - if (mOutputs.valueFor(output)->mMuteCount[stream] != 0) { - LOGV("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 == AudioSystem::VOICE_CALL && mForceUse[AudioSystem::FOR_COMMUNICATION] == AudioSystem::FORCE_BT_SCO) || - (stream == AudioSystem::BLUETOOTH_SCO && mForceUse[AudioSystem::FOR_COMMUNICATION] != AudioSystem::FORCE_BT_SCO)) { - LOGV("checkAndSetVolume() cannot set stream %d volume with force use = %d for comm", - stream, mForceUse[AudioSystem::FOR_COMMUNICATION]); - return INVALID_OPERATION; - } - - float volume = computeVolume(stream, index, output, device); - // 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; - LOGV("setStreamVolume() for output %d stream %d, volume %f, delay %d", output, stream, volume, delayMs); - if (stream == AudioSystem::VOICE_CALL || - stream == AudioSystem::DTMF || - stream == AudioSystem::BLUETOOTH_SCO) { - // offset value to reflect actual hardware volume that never reaches 0 - // 1% corresponds roughly to first step in VOICE_CALL stream volume setting (see AudioService.java) - volume = 0.01 + 0.99 * volume; - // Force VOICE_CALL to track BLUETOOTH_SCO stream volume when bluetooth audio is - // enabled - if (stream == AudioSystem::BLUETOOTH_SCO) { - mpClientInterface->setStreamVolume(AudioSystem::VOICE_CALL, volume, output, delayMs); - } - } - - mpClientInterface->setStreamVolume((AudioSystem::stream_type)stream, volume, output, delayMs); - } - - if (stream == AudioSystem::VOICE_CALL || - stream == AudioSystem::BLUETOOTH_SCO) { - float voiceVolume; - // Force voice volume to max for bluetooth SCO as volume is managed by the headset - if (stream == AudioSystem::VOICE_CALL) { - voiceVolume = (float)index/(float)mStreams[stream].mIndexMax; - } else { - voiceVolume = 1.0; - } - - if (voiceVolume != mLastVoiceVolume && output == mHardwareOutput) { - mpClientInterface->setVoiceVolume(voiceVolume, delayMs); - mLastVoiceVolume = voiceVolume; - } - } - - return NO_ERROR; -} - -void AudioPolicyManagerBase::applyStreamVolumes(audio_io_handle_t output, uint32_t device, int delayMs, bool force) -{ - LOGV("applyStreamVolumes() for output %d and device %x", output, device); - - for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) { - checkAndSetVolume(stream, mStreams[stream].mIndexCur, output, device, delayMs, force); - } -} - -void AudioPolicyManagerBase::setStrategyMute(routing_strategy strategy, bool on, audio_io_handle_t output, int delayMs) -{ - LOGV("setStrategyMute() strategy %d, mute %d, output %d", strategy, on, output); - for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) { - if (getStrategy((AudioSystem::stream_type)stream) == strategy) { - setStreamMute(stream, on, output, delayMs); - } - } -} - -void AudioPolicyManagerBase::setStreamMute(int stream, bool on, audio_io_handle_t output, int delayMs) -{ - StreamDescriptor &streamDesc = mStreams[stream]; - AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output); - - LOGV("setStreamMute() stream %d, mute %d, output %d, mMuteCount %d", stream, on, output, outputDesc->mMuteCount[stream]); - - if (on) { - if (outputDesc->mMuteCount[stream] == 0) { - if (streamDesc.mCanBeMuted) { - checkAndSetVolume(stream, 0, output, outputDesc->device(), delayMs); - } - } - // increment mMuteCount after calling checkAndSetVolume() so that volume change is not ignored - outputDesc->mMuteCount[stream]++; - } else { - if (outputDesc->mMuteCount[stream] == 0) { - LOGW("setStreamMute() unmuting non muted stream!"); - return; - } - if (--outputDesc->mMuteCount[stream] == 0) { - checkAndSetVolume(stream, streamDesc.mIndexCur, output, outputDesc->device(), delayMs); - } - } -} - -void AudioPolicyManagerBase::handleIncallSonification(int 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 - - if (getStrategy((AudioSystem::stream_type)stream) == STRATEGY_SONIFICATION) { - AudioOutputDescriptor *outputDesc = mOutputs.valueFor(mHardwareOutput); - LOGV("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 (AudioSystem::isLowVisibility((AudioSystem::stream_type)stream)) { - LOGV("handleIncallSonification() low visibility, muteCount %d", muteCount); - for (int i = 0; i < muteCount; i++) { - setStreamMute(stream, starting, mHardwareOutput); - } - } else { - LOGV("handleIncallSonification() high visibility"); - if (outputDesc->device() & getDeviceForStrategy(STRATEGY_PHONE)) { - LOGV("handleIncallSonification() high visibility muted, muteCount %d", muteCount); - for (int i = 0; i < muteCount; i++) { - setStreamMute(stream, starting, mHardwareOutput); - } - } - if (starting) { - mpClientInterface->startTone(ToneGenerator::TONE_SUP_CALL_WAITING, AudioSystem::VOICE_CALL); - } else { - mpClientInterface->stopTone(); - } - } - } - } -} - -bool AudioPolicyManagerBase::isInCall() -{ - return isStateInCall(mPhoneState); -} - -bool AudioPolicyManagerBase::isStateInCall(int state) { - return ((state == AudioSystem::MODE_IN_CALL) || - (state == AudioSystem::MODE_IN_COMMUNICATION)); -} - -bool AudioPolicyManagerBase::needsDirectOuput(AudioSystem::stream_type stream, - uint32_t samplingRate, - uint32_t format, - uint32_t channels, - AudioSystem::output_flags flags, - uint32_t device) -{ - return ((flags & AudioSystem::OUTPUT_FLAG_DIRECT) || - (format !=0 && !AudioSystem::isLinearPCM(format))); -} - -uint32_t AudioPolicyManagerBase::getMaxEffectsCpuLoad() -{ - return MAX_EFFECTS_CPU_LOAD; -} - -uint32_t AudioPolicyManagerBase::getMaxEffectsMemory() -{ - return MAX_EFFECTS_MEMORY; -} - -// --- AudioOutputDescriptor class implementation - -AudioPolicyManagerBase::AudioOutputDescriptor::AudioOutputDescriptor() - : mId(0), mSamplingRate(0), mFormat(0), mChannels(0), mLatency(0), - mFlags((AudioSystem::output_flags)0), mDevice(0), mOutput1(0), mOutput2(0) -{ - // clear usage count for all stream types - for (int i = 0; i < AudioSystem::NUM_STREAM_TYPES; i++) { - mRefCount[i] = 0; - mCurVolume[i] = -1.0; - mMuteCount[i] = 0; - mStopTime[i] = 0; - } -} - -uint32_t AudioPolicyManagerBase::AudioOutputDescriptor::device() -{ - uint32_t device = 0; - if (isDuplicated()) { - device = mOutput1->mDevice | mOutput2->mDevice; - } else { - device = mDevice; - } - return device; -} - -void AudioPolicyManagerBase::AudioOutputDescriptor::changeRefCount(AudioSystem::stream_type 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) { - LOGW("changeRefCount() invalid delta %d for stream %d, refCount %d", delta, stream, mRefCount[stream]); - mRefCount[stream] = 0; - return; - } - mRefCount[stream] += delta; - LOGV("changeRefCount() stream %d, count %d", stream, mRefCount[stream]); -} - -uint32_t AudioPolicyManagerBase::AudioOutputDescriptor::refCount() -{ - uint32_t refcount = 0; - for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) { - refcount += mRefCount[i]; - } - return refcount; -} - -uint32_t AudioPolicyManagerBase::AudioOutputDescriptor::strategyRefCount(routing_strategy strategy) -{ - uint32_t refCount = 0; - for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) { - if (getStrategy((AudioSystem::stream_type)i) == strategy) { - refCount += mRefCount[i]; - } - } - return refCount; -} - -status_t AudioPolicyManagerBase::AudioOutputDescriptor::dump(int fd) -{ - const size_t SIZE = 256; - char buffer[SIZE]; - String8 result; - - 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", mChannels); - 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 < AudioSystem::NUM_STREAM_TYPES; 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 - -AudioPolicyManagerBase::AudioInputDescriptor::AudioInputDescriptor() - : mSamplingRate(0), mFormat(0), mChannels(0), - mAcoustics((AudioSystem::audio_in_acoustics)0), mDevice(0), mRefCount(0), - mInputSource(0) -{ -} - -status_t AudioPolicyManagerBase::AudioInputDescriptor::dump(int fd) -{ - const size_t SIZE = 256; - char buffer[SIZE]; - String8 result; - - 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", mChannels); - result.append(buffer); - snprintf(buffer, SIZE, " Acoustics %08x\n", mAcoustics); - result.append(buffer); - snprintf(buffer, SIZE, " Devices %08x\n", mDevice); - result.append(buffer); - snprintf(buffer, SIZE, " Ref Count %d\n", mRefCount); - result.append(buffer); - write(fd, result.string(), result.size()); - - return NO_ERROR; -} - -// --- StreamDescriptor class implementation - -void AudioPolicyManagerBase::StreamDescriptor::dump(char* buffer, size_t size) -{ - snprintf(buffer, size, " %02d %02d %02d %d\n", - mIndexMin, - mIndexMax, - mIndexCur, - mCanBeMuted); -} - -// --- EffectDescriptor class implementation - -status_t AudioPolicyManagerBase::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; -} - - - -}; // namespace android diff --git a/audio/AudioPolicyManagerBase.h b/audio/AudioPolicyManagerBase.h deleted file mode 100755 index 3ab748d..0000000 --- a/audio/AudioPolicyManagerBase.h +++ /dev/null @@ -1,393 +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 <stdint.h> -#include <sys/types.h> -#include <utils/Timers.h> -#include <utils/Errors.h> -#include <utils/KeyedVector.h> -#include <hardware_legacy/AudioPolicyInterface.h> - - -namespace android_audio_legacy { - using android::KeyedVector; - -// ---------------------------------------------------------------------------- - -#define MAX_DEVICE_ADDRESS_LEN 20 -// 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 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 - -// ---------------------------------------------------------------------------- -// AudioPolicyManagerBase implements audio policy manager behavior common to all platforms. -// Each platform must implement an AudioPolicyManager class derived from AudioPolicyManagerBase -// and override methods for which the platform specific behavior differs from the implementation -// in AudioPolicyManagerBase. Even if no specific behavior is required, the AudioPolicyManager -// class must be implemented as well as the class factory function createAudioPolicyManager() -// and provided in a shared library libaudiopolicy.so. -// ---------------------------------------------------------------------------- - -class AudioPolicyManagerBase: public AudioPolicyInterface -#ifdef AUDIO_POLICY_TEST - , public Thread -#endif //AUDIO_POLICY_TEST -{ - -public: - AudioPolicyManagerBase(AudioPolicyClientInterface *clientInterface); - virtual ~AudioPolicyManagerBase(); - - // AudioPolicyInterface - virtual status_t setDeviceConnectionState(AudioSystem::audio_devices device, - AudioSystem::device_connection_state state, - const char *device_address); - virtual AudioSystem::device_connection_state getDeviceConnectionState(AudioSystem::audio_devices device, - const char *device_address); - virtual void setPhoneState(int state); - virtual void setRingerMode(uint32_t mode, uint32_t mask); - virtual void setForceUse(AudioSystem::force_use usage, AudioSystem::forced_config config); - virtual AudioSystem::forced_config getForceUse(AudioSystem::force_use usage); - virtual void setSystemProperty(const char* property, const char* value); - virtual status_t initCheck(); - virtual audio_io_handle_t getOutput(AudioSystem::stream_type stream, - uint32_t samplingRate = 0, - uint32_t format = AudioSystem::FORMAT_DEFAULT, - uint32_t channels = 0, - AudioSystem::output_flags flags = - AudioSystem::OUTPUT_FLAG_INDIRECT); - virtual status_t startOutput(audio_io_handle_t output, - AudioSystem::stream_type stream, - int session = 0); - virtual status_t stopOutput(audio_io_handle_t output, - AudioSystem::stream_type stream, - int session = 0); - virtual void releaseOutput(audio_io_handle_t output); - virtual audio_io_handle_t getInput(int inputSource, - uint32_t samplingRate, - uint32_t format, - uint32_t channels, - AudioSystem::audio_in_acoustics acoustics); - // indicates to the audio policy manager that the input starts being used. - virtual status_t startInput(audio_io_handle_t input); - // indicates to the audio policy manager that the input stops being used. - virtual status_t stopInput(audio_io_handle_t input); - virtual void releaseInput(audio_io_handle_t input); - virtual void initStreamVolume(AudioSystem::stream_type stream, - int indexMin, - int indexMax); - virtual status_t setStreamVolumeIndex(AudioSystem::stream_type stream, int index); - virtual status_t getStreamVolumeIndex(AudioSystem::stream_type stream, int *index); - - // return the strategy corresponding to a given stream type - virtual uint32_t getStrategyForStream(AudioSystem::stream_type stream); - - // return the enabled output devices for the given stream type - virtual uint32_t getDevicesForStream(AudioSystem::stream_type stream); - - virtual audio_io_handle_t getOutputForEffect(effect_descriptor_t *desc); - virtual status_t registerEffect(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(int stream, uint32_t inPastMs = 0) const; - - virtual status_t dump(int fd); - -protected: - - enum routing_strategy { - STRATEGY_MEDIA, - STRATEGY_PHONE, - STRATEGY_SONIFICATION, - STRATEGY_DTMF, - STRATEGY_ENFORCED_AUDIBLE, - 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_CNT - }; - - // default volume curve - static const VolumeCurvePoint sDefaultVolumeCurve[AudioPolicyManagerBase::VOLCNT]; - // default volume curve for media strategy - static const VolumeCurvePoint sDefaultMediaVolumeCurve[AudioPolicyManagerBase::VOLCNT]; - // volume curve for media strategy on speakers - static const VolumeCurvePoint sSpeakerMediaVolumeCurve[AudioPolicyManagerBase::VOLCNT]; - // volume curve for sonification strategy on speakers - static const VolumeCurvePoint sSpeakerSonificationVolumeCurve[AudioPolicyManagerBase::VOLCNT]; - // default volume curves per strategy and device category. See initializeVolumeCurves() - static const VolumeCurvePoint *sVolumeProfiles[NUM_STRATEGIES][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: - AudioOutputDescriptor(); - - status_t dump(int fd); - - uint32_t device(); - void changeRefCount(AudioSystem::stream_type, int delta); - uint32_t refCount(); - uint32_t strategyRefCount(routing_strategy strategy); - bool isUsedByStrategy(routing_strategy strategy) { return (strategyRefCount(strategy) != 0);} - bool isDuplicated() { return (mOutput1 != NULL && mOutput2 != NULL); } - - audio_io_handle_t mId; // output handle - uint32_t mSamplingRate; // - uint32_t mFormat; // - uint32_t mChannels; // output configuration - uint32_t mLatency; // - AudioSystem::output_flags mFlags; // - uint32_t mDevice; // current device this output is routed to - uint32_t mRefCount[AudioSystem::NUM_STREAM_TYPES]; // number of streams of each type using this output - nsecs_t mStopTime[AudioSystem::NUM_STREAM_TYPES]; - AudioOutputDescriptor *mOutput1; // used by duplicated outputs: first output - AudioOutputDescriptor *mOutput2; // used by duplicated outputs: second output - float mCurVolume[AudioSystem::NUM_STREAM_TYPES]; // current stream volume - int mMuteCount[AudioSystem::NUM_STREAM_TYPES]; // mute request counter - }; - - // 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: - AudioInputDescriptor(); - - status_t dump(int fd); - - uint32_t mSamplingRate; // - uint32_t mFormat; // input configuration - uint32_t mChannels; // - AudioSystem::audio_in_acoustics mAcoustics; // - uint32_t mDevice; // current device this input is routed to - uint32_t mRefCount; // number of AudioRecord clients using this output - int mInputSource; // input source selected by application (mediarecorder.h) - }; - - // stream descriptor used for volume control - class StreamDescriptor - { - public: - StreamDescriptor() - : mIndexMin(0), mIndexMax(1), mIndexCur(1), mCanBeMuted(true) {} - - void dump(char* buffer, size_t size); - - int mIndexMin; // min volume index - int mIndexMax; // max volume index - int mIndexCur; // current volume index - bool mCanBeMuted; // true is the stream can be muted - - const VolumeCurvePoint *mVolumeCurve[DEVICE_CATEGORY_CNT]; - }; - - // stream descriptor used for volume control - class EffectDescriptor - { - 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 id, AudioOutputDescriptor *outputDesc); - - // return the strategy corresponding to a given stream type - static routing_strategy getStrategy(AudioSystem::stream_type 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 determined - // 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 updateDeviceForStrategy() is called. - virtual uint32_t getDeviceForStrategy(routing_strategy strategy, bool fromCache = true); - // change the route of the specified output - void setOutputDevice(audio_io_handle_t output, uint32_t device, bool force = false, int delayMs = 0); - // select input device corresponding to requested audio source - virtual uint32_t getDeviceForInputSource(int inputSource); - // return io handle of active input or 0 if no input is active - audio_io_handle_t getActiveInput(); - // 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(int stream, int index, audio_io_handle_t output, uint32_t device); - // check that volume change is permitted, compute and send new volume to audio hardware - status_t checkAndSetVolume(int stream, int index, audio_io_handle_t output, uint32_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, uint32_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); - // Mute or unmute the stream on the specified output - void setStreamMute(int stream, bool on, audio_io_handle_t output, int delayMs = 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(int stream, bool starting, bool stateChange); - // true is current platform implements a back microphone - virtual bool hasBackMicrophone() const { return false; } - // 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); - -#ifdef WITH_A2DP - // true is current platform supports suplication of notifications and ringtones over A2DP output - virtual bool a2dpUsedForSonification() const { return true; } - status_t handleA2dpConnection(AudioSystem::audio_devices device, - const char *device_address); - status_t handleA2dpDisconnection(AudioSystem::audio_devices device, - const char *device_address); - void closeA2dpOutputs(); - // checks and if necessary changes output (a2dp, duplicated or hardware) used for all strategies. - // must be called every time a condition that affects the output choice for a given strategy is - // changed: connected device, phone state, force use... - // Must be called before updateDeviceForStrategy() - 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(); -#endif - // 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 - uint32_t getNewDevice(audio_io_handle_t output, bool fromCache = true); - // 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 updateDeviceForStrategy(); - // true if current platform requires a specific output to be opened for this particular - // set of parameters. This function is called by getOutput() and is implemented by platform - // specific audio policy manager. - virtual bool needsDirectOuput(AudioSystem::stream_type stream, - uint32_t samplingRate, - uint32_t format, - uint32_t channels, - AudioSystem::output_flags flags, - uint32_t device); - 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(EffectDescriptor *pDesc, bool enabled); - - // returns the category the device belongs to with regard to volume curve management - static device_category getDeviceCategory(uint32_t device); - - AudioPolicyClientInterface *mpClientInterface; // audio policy client interface - audio_io_handle_t mHardwareOutput; // hardware output handler - audio_io_handle_t mA2dpOutput; // A2DP output handler - audio_io_handle_t mDuplicatedOutput; // duplicated output handler: outputs to hardware and A2DP. - - KeyedVector<audio_io_handle_t, AudioOutputDescriptor *> mOutputs; // list of output descriptors - KeyedVector<audio_io_handle_t, AudioInputDescriptor *> mInputs; // list of input descriptors - uint32_t mAvailableOutputDevices; // bit field of all available output devices - uint32_t mAvailableInputDevices; // bit field of all available input devices - int mPhoneState; // current phone state - uint32_t mRingerMode; // current ringer mode - AudioSystem::forced_config mForceUse[AudioSystem::NUM_FORCE_USE]; // current forced use configuration - - StreamDescriptor mStreams[AudioSystem::NUM_STREAM_TYPES]; // stream descriptors for volume control - String8 mA2dpDeviceAddress; // A2DP device MAC address - String8 mScoDeviceAddress; // SCO device MAC address - bool mLimitRingtoneVolume; // limit ringtone volume to music volume if headset connected - uint32_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<int, EffectDescriptor *> mEffects; // list of registered audio effects - bool mA2dpSuspended; // true if A2DP output is suspended - -#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 - -private: - static float volIndexToAmpl(uint32_t device, const StreamDescriptor& streamDesc, - int indexInUi); -}; - -}; diff --git a/audio/AudioPolicyManagerDefault.cpp b/audio/AudioPolicyManagerDefault.cpp deleted file mode 100755 index 9083638..0000000 --- a/audio/AudioPolicyManagerDefault.cpp +++ /dev/null @@ -1,34 +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 "AudioPolicyManagerDefault" -//#define LOG_NDEBUG 0 - -#include "AudioPolicyManagerDefault.h" - -namespace android_audio_legacy { - -extern "C" AudioPolicyInterface* createAudioPolicyManager(AudioPolicyClientInterface *clientInterface) -{ - return new AudioPolicyManagerDefault(clientInterface); -} - -extern "C" void destroyAudioPolicyManager(AudioPolicyInterface *interface) -{ - delete interface; -} - -}; // namespace android diff --git a/audio/AudioPolicyManagerDefault.h b/audio/AudioPolicyManagerDefault.h deleted file mode 100755 index b2b2576..0000000 --- a/audio/AudioPolicyManagerDefault.h +++ /dev/null @@ -1,43 +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 <stdint.h> -#include <stdbool.h> - -#include <hardware_legacy/AudioPolicyManagerBase.h> - -namespace android_audio_legacy { - -class AudioPolicyManagerDefault: public AudioPolicyManagerBase -{ - -public: - AudioPolicyManagerDefault(AudioPolicyClientInterface *clientInterface) - : AudioPolicyManagerBase(clientInterface) {} - - virtual ~AudioPolicyManagerDefault() {} - -protected: - // true is current platform implements a back microphone - virtual bool hasBackMicrophone() const { return false; } -#ifdef WITH_A2DP - // true is current platform supports suplication of notifications and ringtones over A2DP output - virtual bool a2dpUsedForSonification() const { return true; } -#endif - -}; -}; diff --git a/audio/AudioSystem.h b/audio/AudioSystem.h deleted file mode 100755 index bd75a73..0000000 --- a/audio/AudioSystem.h +++ /dev/null @@ -1,562 +0,0 @@ -/* - * Copyright (C) 2008 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_AUDIOSYSTEM_H_ -#define ANDROID_AUDIOSYSTEM_H_ -#define ANDROID_AUDIOPARAMETER_H_ - -#include <utils/RefBase.h> -#include <utils/threads.h> -#include <media/IAudioFlinger.h> - -namespace android { - -typedef void (*audio_error_callback)(status_t err); -typedef int audio_io_handle_t; - -class IAudioPolicyService; -class String8; - -class AudioSystem -{ -public: - - enum stream_type { - DEFAULT =-1, - VOICE_CALL = 0, - SYSTEM = 1, - RING = 2, - MUSIC = 3, - ALARM = 4, - NOTIFICATION = 5, - BLUETOOTH_SCO = 6, - ENFORCED_AUDIBLE = 7, // Sounds that cannot be muted by user and must be routed to speaker - DTMF = 8, - TTS = 9, -#ifdef HAVE_FM_RADIO - FM = 10, -#endif - NUM_STREAM_TYPES - }; - - // Audio sub formats (see AudioSystem::audio_format). - enum pcm_sub_format { - PCM_SUB_16_BIT = 0x1, // must be 1 for backward compatibility - PCM_SUB_8_BIT = 0x2, // must be 2 for backward compatibility - }; - - // MP3 sub format field definition : can use 11 LSBs in the same way as MP3 frame header to specify - // bit rate, stereo mode, version... - enum mp3_sub_format { - //TODO - }; - - // AMR NB/WB sub format field definition: specify frame block interleaving, bandwidth efficient or octet aligned, - // encoding mode for recording... - enum amr_sub_format { - //TODO - }; - - // AAC sub format field definition: specify profile or bitrate for recording... - enum aac_sub_format { - //TODO - }; - - // VORBIS sub format field definition: specify quality for recording... - enum vorbis_sub_format { - //TODO - }; - - // Audio format consists in a main format field (upper 8 bits) and a sub format field (lower 24 bits). - // The main format indicates the main codec type. The sub format field indicates options and parameters - // for each format. The sub format is mainly used for record to indicate for instance the requested bitrate - // or profile. It can also be used for certain formats to give informations not present in the encoded - // audio stream (e.g. octet alignement for AMR). - enum audio_format { - INVALID_FORMAT = -1, - FORMAT_DEFAULT = 0, - PCM = 0x00000000, // must be 0 for backward compatibility - MP3 = 0x01000000, - AMR_NB = 0x02000000, - AMR_WB = 0x03000000, - AAC = 0x04000000, - HE_AAC_V1 = 0x05000000, - HE_AAC_V2 = 0x06000000, - VORBIS = 0x07000000, - MAIN_FORMAT_MASK = 0xFF000000, - SUB_FORMAT_MASK = 0x00FFFFFF, - // Aliases - PCM_16_BIT = (PCM|PCM_SUB_16_BIT), - PCM_8_BIT = (PCM|PCM_SUB_8_BIT) - }; - - - // Channel mask definitions must be kept in sync with JAVA values in /media/java/android/media/AudioFormat.java - enum audio_channels { - // output channels - CHANNEL_OUT_FRONT_LEFT = 0x4, - CHANNEL_OUT_FRONT_RIGHT = 0x8, - CHANNEL_OUT_FRONT_CENTER = 0x10, - CHANNEL_OUT_LOW_FREQUENCY = 0x20, - CHANNEL_OUT_BACK_LEFT = 0x40, - CHANNEL_OUT_BACK_RIGHT = 0x80, - CHANNEL_OUT_FRONT_LEFT_OF_CENTER = 0x100, - CHANNEL_OUT_FRONT_RIGHT_OF_CENTER = 0x200, - CHANNEL_OUT_BACK_CENTER = 0x400, - CHANNEL_OUT_MONO = CHANNEL_OUT_FRONT_LEFT, - CHANNEL_OUT_STEREO = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT), - CHANNEL_OUT_QUAD = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT | - CHANNEL_OUT_BACK_LEFT | CHANNEL_OUT_BACK_RIGHT), - CHANNEL_OUT_SURROUND = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT | - CHANNEL_OUT_FRONT_CENTER | CHANNEL_OUT_BACK_CENTER), - CHANNEL_OUT_5POINT1 = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT | - CHANNEL_OUT_FRONT_CENTER | CHANNEL_OUT_LOW_FREQUENCY | CHANNEL_OUT_BACK_LEFT | CHANNEL_OUT_BACK_RIGHT), - CHANNEL_OUT_7POINT1 = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT | - CHANNEL_OUT_FRONT_CENTER | CHANNEL_OUT_LOW_FREQUENCY | CHANNEL_OUT_BACK_LEFT | CHANNEL_OUT_BACK_RIGHT | - CHANNEL_OUT_FRONT_LEFT_OF_CENTER | CHANNEL_OUT_FRONT_RIGHT_OF_CENTER), - CHANNEL_OUT_ALL = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT | - CHANNEL_OUT_FRONT_CENTER | CHANNEL_OUT_LOW_FREQUENCY | CHANNEL_OUT_BACK_LEFT | CHANNEL_OUT_BACK_RIGHT | - CHANNEL_OUT_FRONT_LEFT_OF_CENTER | CHANNEL_OUT_FRONT_RIGHT_OF_CENTER | CHANNEL_OUT_BACK_CENTER), - - // input channels - CHANNEL_IN_LEFT = 0x4, - CHANNEL_IN_RIGHT = 0x8, - CHANNEL_IN_FRONT = 0x10, - CHANNEL_IN_BACK = 0x20, - CHANNEL_IN_LEFT_PROCESSED = 0x40, - CHANNEL_IN_RIGHT_PROCESSED = 0x80, - CHANNEL_IN_FRONT_PROCESSED = 0x100, - CHANNEL_IN_BACK_PROCESSED = 0x200, - CHANNEL_IN_PRESSURE = 0x400, - CHANNEL_IN_X_AXIS = 0x800, - CHANNEL_IN_Y_AXIS = 0x1000, - CHANNEL_IN_Z_AXIS = 0x2000, - CHANNEL_IN_VOICE_UPLINK = 0x4000, - CHANNEL_IN_VOICE_DNLINK = 0x8000, -#ifdef OMAP_ENHANCEMENT - CHANNEL_IN_VOICE_UPLINK_DNLINK = 0x10000, -#endif - CHANNEL_IN_MONO = CHANNEL_IN_FRONT, - CHANNEL_IN_STEREO = (CHANNEL_IN_LEFT | CHANNEL_IN_RIGHT), - CHANNEL_IN_ALL = (CHANNEL_IN_LEFT | CHANNEL_IN_RIGHT | CHANNEL_IN_FRONT | CHANNEL_IN_BACK| - CHANNEL_IN_LEFT_PROCESSED | CHANNEL_IN_RIGHT_PROCESSED | CHANNEL_IN_FRONT_PROCESSED | CHANNEL_IN_BACK_PROCESSED| - CHANNEL_IN_PRESSURE | CHANNEL_IN_X_AXIS | CHANNEL_IN_Y_AXIS | CHANNEL_IN_Z_AXIS | -#ifdef OMAP_ENHANCEMENT - CHANNEL_IN_VOICE_UPLINK | CHANNEL_IN_VOICE_DNLINK | CHANNEL_IN_VOICE_UPLINK_DNLINK) -#else - CHANNEL_IN_VOICE_UPLINK | CHANNEL_IN_VOICE_DNLINK ) -#endif - }; - - enum audio_mode { - MODE_INVALID = -2, - MODE_CURRENT = -1, - MODE_NORMAL = 0, - MODE_RINGTONE, - MODE_IN_CALL, - MODE_IN_COMMUNICATION, - NUM_MODES // not a valid entry, denotes end-of-list - }; - - enum audio_in_acoustics { - AGC_ENABLE = 0x0001, - AGC_DISABLE = 0, - NS_ENABLE = 0x0002, - NS_DISABLE = 0, - TX_IIR_ENABLE = 0x0004, - TX_DISABLE = 0 - }; - - // special audio session values - enum audio_sessions { - SESSION_OUTPUT_STAGE = -1, // session for effects attached to a particular output stream - // (value must be less than 0) - SESSION_OUTPUT_MIX = 0, // session for effects applied to output mix. These effects can - // be moved by audio policy manager to another output stream - // (value must be 0) - }; - - /* These are static methods to control the system-wide AudioFlinger - * only privileged processes can have access to them - */ - - // mute/unmute microphone - static status_t muteMicrophone(bool state); - static status_t isMicrophoneMuted(bool *state); - - // set/get master volume - static status_t setMasterVolume(float value); - static status_t getMasterVolume(float* volume); - // mute/unmute audio outputs - static status_t setMasterMute(bool mute); - static status_t getMasterMute(bool* mute); - - // set/get stream volume on specified output - static status_t setStreamVolume(int stream, float value, int output); - static status_t getStreamVolume(int stream, float* volume, int output); - - // mute/unmute stream - static status_t setStreamMute(int stream, bool mute); - static status_t getStreamMute(int stream, bool* mute); - - // set audio mode in audio hardware (see AudioSystem::audio_mode) - static status_t setMode(int mode); - - // returns true in *state if tracks are active on the specified stream - static status_t isStreamActive(int stream, bool *state); - - // set/get audio hardware parameters. The function accepts a list of parameters - // key value pairs in the form: key1=value1;key2=value2;... - // Some keys are reserved for standard parameters (See AudioParameter class). - static status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs); - static String8 getParameters(audio_io_handle_t ioHandle, const String8& keys); - - static void setErrorCallback(audio_error_callback cb); - - // helper function to obtain AudioFlinger service handle - static const sp<IAudioFlinger>& get_audio_flinger(); - - static float linearToLog(int volume); - static int logToLinear(float volume); - - static status_t getOutputSamplingRate(int* samplingRate, int stream = DEFAULT); - static status_t getOutputFrameCount(int* frameCount, int stream = DEFAULT); - static status_t getOutputLatency(uint32_t* latency, int stream = DEFAULT); - - static bool routedToA2dpOutput(int streamType); - - static status_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount, - size_t* buffSize); - - static status_t setVoiceVolume(float volume); -#ifdef HAVE_FM_RADIO - static status_t setFmVolume(float volume); -#endif - - // return the number of audio frames written by AudioFlinger to audio HAL and - // audio dsp to DAC since the output on which the specificed stream is playing - // has exited standby. - // returned status (from utils/Errors.h) can be: - // - NO_ERROR: successful operation, halFrames and dspFrames point to valid data - // - INVALID_OPERATION: Not supported on current hardware platform - // - BAD_VALUE: invalid parameter - // NOTE: this feature is not supported on all hardware platforms and it is - // necessary to check returned status before using the returned values. - static status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int stream = DEFAULT); - - static unsigned int getInputFramesLost(audio_io_handle_t ioHandle); - - static int newAudioSessionId(); - // - // AudioPolicyService interface - // - - enum audio_devices { - // output devices - DEVICE_OUT_EARPIECE = 0x1, - DEVICE_OUT_SPEAKER = 0x2, - DEVICE_OUT_WIRED_HEADSET = 0x4, - DEVICE_OUT_WIRED_HEADPHONE = 0x8, - DEVICE_OUT_BLUETOOTH_SCO = 0x10, - DEVICE_OUT_BLUETOOTH_SCO_HEADSET = 0x20, - DEVICE_OUT_BLUETOOTH_SCO_CARKIT = 0x40, - DEVICE_OUT_BLUETOOTH_A2DP = 0x80, - DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES = 0x100, - DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER = 0x200, - DEVICE_OUT_AUX_DIGITAL = 0x400, -#ifdef HAVE_FM_RADIO - DEVICE_OUT_FM = 0x800, - DEVICE_OUT_FM_SPEAKER = 0x1000, - DEVICE_OUT_FM_ALL = (DEVICE_OUT_FM | DEVICE_OUT_FM_SPEAKER), -#elif defined(OMAP_ENHANCEMENT) - DEVICE_OUT_FM_TRANSMIT = 0x800, - DEVICE_OUT_LOW_POWER = 0x1000, -#endif - DEVICE_OUT_HDMI = 0x2000, - DEVICE_OUT_DEFAULT = 0x8000, - DEVICE_OUT_ALL = (DEVICE_OUT_EARPIECE | DEVICE_OUT_SPEAKER | DEVICE_OUT_WIRED_HEADSET | -#ifdef HAVE_FM_RADIO - DEVICE_OUT_WIRED_HEADPHONE | DEVICE_OUT_FM | DEVICE_OUT_FM_SPEAKER | DEVICE_OUT_BLUETOOTH_SCO | DEVICE_OUT_BLUETOOTH_SCO_HEADSET | -#else - DEVICE_OUT_WIRED_HEADPHONE | DEVICE_OUT_BLUETOOTH_SCO | DEVICE_OUT_BLUETOOTH_SCO_HEADSET | -#endif - DEVICE_OUT_BLUETOOTH_SCO_CARKIT | DEVICE_OUT_BLUETOOTH_A2DP | DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES | -#if defined(OMAP_ENHANCEMENT) && !defined(HAVE_FM_RADIO) - DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER | DEVICE_OUT_AUX_DIGITAL | DEVICE_OUT_LOW_POWER | - DEVICE_OUT_FM_TRANSMIT | DEVICE_OUT_DEFAULT), -#else - DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER | DEVICE_OUT_AUX_DIGITAL | DEVICE_OUT_HDMI | DEVICE_OUT_DEFAULT), -#endif - DEVICE_OUT_ALL_A2DP = (DEVICE_OUT_BLUETOOTH_A2DP | DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES | - DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER), - - // input devices - DEVICE_IN_COMMUNICATION = 0x10000, - DEVICE_IN_AMBIENT = 0x20000, - DEVICE_IN_BUILTIN_MIC = 0x40000, - DEVICE_IN_BLUETOOTH_SCO_HEADSET = 0x80000, - DEVICE_IN_WIRED_HEADSET = 0x100000, - DEVICE_IN_AUX_DIGITAL = 0x200000, - DEVICE_IN_VOICE_CALL = 0x400000, - DEVICE_IN_BACK_MIC = 0x800000, -#ifdef HAVE_FM_RADIO - DEVICE_IN_FM_RX = 0x1000000, - DEVICE_IN_FM_RX_A2DP = 0x2000000, -#endif -#ifdef OMAP_ENHANCEMENT - DEVICE_IN_FM_ANALOG = 0x1000000, -#endif - DEVICE_IN_DEFAULT = 0x80000000, - - DEVICE_IN_ALL = (DEVICE_IN_COMMUNICATION | DEVICE_IN_AMBIENT | DEVICE_IN_BUILTIN_MIC | - DEVICE_IN_BLUETOOTH_SCO_HEADSET | DEVICE_IN_WIRED_HEADSET | DEVICE_IN_AUX_DIGITAL | -#ifdef HAVE_FM_RADIO - DEVICE_IN_VOICE_CALL | DEVICE_IN_BACK_MIC | DEVICE_IN_FM_RX | DEVICE_IN_FM_RX_A2DP | DEVICE_IN_DEFAULT) -#elif OMAP_ENHANCEMENT - DEVICE_IN_VOICE_CALL | DEVICE_IN_BACK_MIC | DEVICE_IN_FM_ANALOG | DEVICE_IN_DEFAULT) -#else - DEVICE_IN_VOICE_CALL | DEVICE_IN_BACK_MIC | DEVICE_IN_DEFAULT) -#endif - - }; - - // device connection states used for setDeviceConnectionState() - enum device_connection_state { - DEVICE_STATE_UNAVAILABLE, - DEVICE_STATE_AVAILABLE, - NUM_DEVICE_STATES - }; - - // request to open a direct output with getOutput() (by opposition to sharing an output with other AudioTracks) - enum output_flags { - OUTPUT_FLAG_INDIRECT = 0x0, - OUTPUT_FLAG_DIRECT = 0x1 - }; - - // device categories used for setForceUse() - enum forced_config { - FORCE_NONE, - FORCE_SPEAKER, - FORCE_HEADPHONES, - FORCE_BT_SCO, - FORCE_BT_A2DP, - FORCE_WIRED_ACCESSORY, - FORCE_BT_CAR_DOCK, - FORCE_BT_DESK_DOCK, - NUM_FORCE_CONFIG, - FORCE_DEFAULT = FORCE_NONE - }; - - // usages used for setForceUse() - enum force_use { - FOR_COMMUNICATION, - FOR_MEDIA, - FOR_RECORD, - FOR_DOCK, - NUM_FORCE_USE - }; - - // types of io configuration change events received with ioConfigChanged() - enum io_config_event { - OUTPUT_OPENED, - OUTPUT_CLOSED, - OUTPUT_CONFIG_CHANGED, - INPUT_OPENED, - INPUT_CLOSED, - INPUT_CONFIG_CHANGED, - STREAM_CONFIG_CHANGED, - NUM_CONFIG_EVENTS - }; - - // audio output descritor used to cache output configurations in client process to avoid frequent calls - // through IAudioFlinger - class OutputDescriptor { - public: - OutputDescriptor() - : samplingRate(0), format(0), channels(0), frameCount(0), latency(0) {} - - uint32_t samplingRate; - int32_t format; - int32_t channels; - size_t frameCount; - uint32_t latency; - }; - - // - // IAudioPolicyService interface (see AudioPolicyInterface for method descriptions) - // - static status_t setDeviceConnectionState(audio_devices device, device_connection_state state, const char *device_address); - static device_connection_state getDeviceConnectionState(audio_devices device, const char *device_address); - static status_t setPhoneState(int state); - static status_t setRingerMode(uint32_t mode, uint32_t mask); - static status_t setForceUse(force_use usage, forced_config config); - static forced_config getForceUse(force_use usage); - static audio_io_handle_t getOutput(stream_type stream, - uint32_t samplingRate = 0, - uint32_t format = FORMAT_DEFAULT, - uint32_t channels = CHANNEL_OUT_STEREO, - output_flags flags = OUTPUT_FLAG_INDIRECT); - static status_t startOutput(audio_io_handle_t output, - AudioSystem::stream_type stream, - int session = 0); - static status_t stopOutput(audio_io_handle_t output, - AudioSystem::stream_type stream, - int session = 0); - static void releaseOutput(audio_io_handle_t output); - static audio_io_handle_t getInput(int inputSource, - uint32_t samplingRate = 0, - uint32_t format = FORMAT_DEFAULT, - uint32_t channels = CHANNEL_IN_MONO, - audio_in_acoustics acoustics = (audio_in_acoustics)0); - static status_t startInput(audio_io_handle_t input); - static status_t stopInput(audio_io_handle_t input); - static void releaseInput(audio_io_handle_t input); - static status_t initStreamVolume(stream_type stream, - int indexMin, - int indexMax); - static status_t setStreamVolumeIndex(stream_type stream, int index); - static status_t getStreamVolumeIndex(stream_type stream, int *index); - - static uint32_t getStrategyForStream(stream_type stream); - - static audio_io_handle_t getOutputForEffect(effect_descriptor_t *desc); - static status_t registerEffect(effect_descriptor_t *desc, - audio_io_handle_t output, - uint32_t strategy, - int session, - int id); - static status_t unregisterEffect(int id); - - static const sp<IAudioPolicyService>& get_audio_policy_service(); - - // ---------------------------------------------------------------------------- - - static uint32_t popCount(uint32_t u); - static bool isOutputDevice(audio_devices device); - static bool isInputDevice(audio_devices device); - static bool isA2dpDevice(audio_devices device); -#ifdef HAVE_FM_RADIO - static bool isFmDevice(audio_devices device); -#endif - static bool isBluetoothScoDevice(audio_devices device); - static bool isLowVisibility(stream_type stream); - static bool isOutputChannel(uint32_t channel); - static bool isInputChannel(uint32_t channel); - static bool isValidFormat(uint32_t format); - static bool isLinearPCM(uint32_t format); - -private: - - class AudioFlingerClient: public IBinder::DeathRecipient, public BnAudioFlingerClient - { - public: - AudioFlingerClient() { - } - - // DeathRecipient - virtual void binderDied(const wp<IBinder>& who); - - // IAudioFlingerClient - - // indicate a change in the configuration of an output or input: keeps the cached - // values for output/input parameters upto date in client process - virtual void ioConfigChanged(int event, int ioHandle, void *param2); - }; - - class AudioPolicyServiceClient: public IBinder::DeathRecipient - { - public: - AudioPolicyServiceClient() { - } - - // DeathRecipient - virtual void binderDied(const wp<IBinder>& who); - }; - - static sp<AudioFlingerClient> gAudioFlingerClient; - static sp<AudioPolicyServiceClient> gAudioPolicyServiceClient; - friend class AudioFlingerClient; - friend class AudioPolicyServiceClient; - - static Mutex gLock; - static sp<IAudioFlinger> gAudioFlinger; - static audio_error_callback gAudioErrorCallback; - - static size_t gInBuffSize; - // previous parameters for recording buffer size queries - static uint32_t gPrevInSamplingRate; - static int gPrevInFormat; - static int gPrevInChannelCount; - - static sp<IAudioPolicyService> gAudioPolicyService; - - // mapping between stream types and outputs - static DefaultKeyedVector<int, audio_io_handle_t> gStreamOutputMap; - // list of output descritor containing cached parameters (sampling rate, framecount, channel count...) - static DefaultKeyedVector<audio_io_handle_t, OutputDescriptor *> gOutputs; -}; - -class AudioParameter { - -public: - AudioParameter() {} - AudioParameter(const String8& keyValuePairs); - virtual ~AudioParameter(); - - // reserved parameter keys for changing standard parameters with setParameters() function. - // Using these keys is mandatory for AudioFlinger to properly monitor audio output/input - // configuration changes and act accordingly. - // keyRouting: to change audio routing, value is an int in AudioSystem::audio_devices - // keySamplingRate: to change sampling rate routing, value is an int - // keyFormat: to change audio format, value is an int in AudioSystem::audio_format - // keyChannels: to change audio channel configuration, value is an int in AudioSystem::audio_channels - // keyFrameCount: to change audio output frame count, value is an int - // keyInputSource: to change audio input source, value is an int in audio_source - // (defined in media/mediarecorder.h) - static const char *keyRouting; - static const char *keySamplingRate; - static const char *keyFormat; - static const char *keyChannels; - static const char *keyFrameCount; -#ifdef HAVE_FM_RADIO - static const char *keyFmOn; - static const char *keyFmOff; -#endif - static const char *keyInputSource; - - String8 toString(); - - status_t add(const String8& key, const String8& value); - status_t addInt(const String8& key, const int value); - status_t addFloat(const String8& key, const float value); - - status_t remove(const String8& key); - - status_t get(const String8& key, String8& value); - status_t getInt(const String8& key, int& value); - status_t getFloat(const String8& key, float& value); - status_t getAt(size_t index, String8& key, String8& value); - - size_t size() { return mParameters.size(); } - -private: - String8 mKeyValuePairs; - KeyedVector <String8, String8> mParameters; -}; - -}; // namespace android - -#endif /*ANDROID_AUDIOSYSTEM_H_*/ diff --git a/audio/MODULE_LICENSE_APACHE2 b/audio/MODULE_LICENSE_APACHE2 deleted file mode 100755 index e69de29..0000000 --- a/audio/MODULE_LICENSE_APACHE2 +++ /dev/null diff --git a/audio/NOTICE b/audio/NOTICE deleted file mode 100755 index 3237da6..0000000 --- a/audio/NOTICE +++ /dev/null @@ -1,190 +0,0 @@ - - Copyright (c) 2008-2009, 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. - - 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. - - - Apache License - Version 2.0, January 2004 - http://www.apache.org/licenses/ - - TERMS AND CONDITIONS FOR USE, REPRODUCTION, AND DISTRIBUTION - - 1. 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"0" : "1"); -} - -static int ap_init_check(const struct audio_policy *pol) -{ - const struct legacy_audio_policy *lap = to_clap(pol); - return lap->apm->initCheck(); -} - -static audio_io_handle_t ap_get_output(struct audio_policy *pol, - audio_stream_type_t stream, - uint32_t sampling_rate, - uint32_t format, - uint32_t channels, - audio_policy_output_flags_t flags) -{ - struct legacy_audio_policy *lap = to_lap(pol); - - LOGV("%s: tid %d", __func__, gettid()); - return lap->apm->getOutput((AudioSystem::stream_type)stream, - sampling_rate, format, channels, - (AudioSystem::output_flags)flags); -} - -static int ap_start_output(struct audio_policy *pol, audio_io_handle_t output, - audio_stream_type_t stream, int session) -{ - struct legacy_audio_policy *lap = to_lap(pol); - return lap->apm->startOutput(output, (AudioSystem::stream_type)stream, - session); -} - -static int ap_stop_output(struct audio_policy *pol, audio_io_handle_t output, - audio_stream_type_t stream, int session) -{ - struct legacy_audio_policy *lap = to_lap(pol); - return lap->apm->stopOutput(output, (AudioSystem::stream_type)stream, - session); -} - -static void ap_release_output(struct audio_policy *pol, - audio_io_handle_t output) -{ - struct legacy_audio_policy *lap = to_lap(pol); - lap->apm->releaseOutput(output); -} - -static audio_io_handle_t ap_get_input(struct audio_policy *pol, int inputSource, - uint32_t sampling_rate, - uint32_t format, - uint32_t channels, - audio_in_acoustics_t acoustics) -{ - struct legacy_audio_policy *lap = to_lap(pol); - return lap->apm->getInput(inputSource, sampling_rate, format, channels, - (AudioSystem::audio_in_acoustics)acoustics); -} - -static int ap_start_input(struct audio_policy *pol, audio_io_handle_t input) -{ - struct legacy_audio_policy *lap = to_lap(pol); - return lap->apm->startInput(input); -} - -static int ap_stop_input(struct audio_policy *pol, audio_io_handle_t input) -{ - struct legacy_audio_policy *lap = to_lap(pol); - return lap->apm->stopInput(input); -} - -static void ap_release_input(struct audio_policy *pol, audio_io_handle_t input) -{ - struct legacy_audio_policy *lap = to_lap(pol); - lap->apm->releaseInput(input); -} - -static void ap_init_stream_volume(struct audio_policy *pol, - audio_stream_type_t stream, int index_min, - int index_max) -{ - struct legacy_audio_policy *lap = to_lap(pol); - lap->apm->initStreamVolume((AudioSystem::stream_type)stream, index_min, - index_max); -} - -static int ap_set_stream_volume_index(struct audio_policy *pol, - audio_stream_type_t stream, - int index) -{ - struct legacy_audio_policy *lap = to_lap(pol); - return lap->apm->setStreamVolumeIndex((AudioSystem::stream_type)stream, - index); -} - -static int ap_get_stream_volume_index(const struct audio_policy *pol, - audio_stream_type_t stream, - int *index) -{ - const struct legacy_audio_policy *lap = to_clap(pol); - return lap->apm->getStreamVolumeIndex((AudioSystem::stream_type)stream, - index); -} - -static uint32_t ap_get_strategy_for_stream(const struct audio_policy *pol, - audio_stream_type_t stream) -{ - const struct legacy_audio_policy *lap = to_clap(pol); - return lap->apm->getStrategyForStream((AudioSystem::stream_type)stream); -} - -static uint32_t ap_get_devices_for_stream(const struct audio_policy *pol, - audio_stream_type_t stream) -{ - const struct legacy_audio_policy *lap = to_clap(pol); - return lap->apm->getDevicesForStream((AudioSystem::stream_type)stream); -} - -static audio_io_handle_t ap_get_output_for_effect(struct audio_policy *pol, - struct effect_descriptor_s *desc) -{ - struct legacy_audio_policy *lap = to_lap(pol); - return lap->apm->getOutputForEffect(desc); -} - -static int ap_register_effect(struct audio_policy *pol, - struct effect_descriptor_s *desc, - audio_io_handle_t io, - uint32_t strategy, - int session, - int id) -{ - struct legacy_audio_policy *lap = to_lap(pol); - return lap->apm->registerEffect(desc, io, strategy, session, id); -} - -static int ap_unregister_effect(struct audio_policy *pol, int id) -{ - struct legacy_audio_policy *lap = to_lap(pol); - return lap->apm->unregisterEffect(id); -} - -static int ap_set_effect_enabled(struct audio_policy *pol, int id, bool enabled) -{ - return NO_ERROR; -} - -static bool ap_is_stream_active(const struct audio_policy *pol, int stream, - uint32_t in_past_ms) -{ - const struct legacy_audio_policy *lap = to_clap(pol); - return lap->apm->isStreamActive(stream, in_past_ms); -} - -static int ap_dump(const struct audio_policy *pol, int fd) -{ - const struct legacy_audio_policy *lap = to_clap(pol); - return lap->apm->dump(fd); -} - -static int create_legacy_ap(const struct audio_policy_device *device, - struct audio_policy_service_ops *aps_ops, - void *service, - struct audio_policy **ap) -{ - struct legacy_audio_policy *lap; - int ret; - - if (!service || !aps_ops) - return -EINVAL; - - lap = (struct legacy_audio_policy *)calloc(1, sizeof(*lap)); - if (!lap) - return -ENOMEM; - - lap->policy.set_device_connection_state = ap_set_device_connection_state; - lap->policy.get_device_connection_state = ap_get_device_connection_state; - lap->policy.set_phone_state = ap_set_phone_state; - lap->policy.set_ringer_mode = ap_set_ringer_mode; - lap->policy.set_force_use = ap_set_force_use; - lap->policy.get_force_use = ap_get_force_use; - lap->policy.set_can_mute_enforced_audible = - ap_set_can_mute_enforced_audible; - lap->policy.init_check = ap_init_check; - lap->policy.get_output = ap_get_output; - lap->policy.start_output = ap_start_output; - lap->policy.stop_output = ap_stop_output; - lap->policy.release_output = ap_release_output; - lap->policy.get_input = ap_get_input; - lap->policy.start_input = ap_start_input; - lap->policy.stop_input = ap_stop_input; - lap->policy.release_input = ap_release_input; - lap->policy.init_stream_volume = ap_init_stream_volume; - lap->policy.set_stream_volume_index = ap_set_stream_volume_index; - lap->policy.get_stream_volume_index = ap_get_stream_volume_index; - lap->policy.get_strategy_for_stream = ap_get_strategy_for_stream; - lap->policy.get_devices_for_stream = ap_get_devices_for_stream; - lap->policy.get_output_for_effect = ap_get_output_for_effect; - lap->policy.register_effect = ap_register_effect; - lap->policy.unregister_effect = ap_unregister_effect; - lap->policy.set_effect_enabled = ap_set_effect_enabled; - lap->policy.is_stream_active = ap_is_stream_active; - lap->policy.dump = ap_dump; - - lap->service = service; - lap->aps_ops = aps_ops; - lap->service_client = - new AudioPolicyCompatClient(aps_ops, service); - if (!lap->service_client) { - ret = -ENOMEM; - goto err_new_compat_client; - } - - lap->apm = createAudioPolicyManager(lap->service_client); - if (!lap->apm) { - ret = -ENOMEM; - goto err_create_apm; - } - - *ap = &lap->policy; - return 0; - -err_create_apm: - delete lap->service_client; -err_new_compat_client: - free(lap); - *ap = NULL; - return ret; -} - -static int destroy_legacy_ap(const struct audio_policy_device *ap_dev, - struct audio_policy *ap) -{ - struct legacy_audio_policy *lap = to_lap(ap); - - if (!lap) - return 0; - - if (lap->apm) - destroyAudioPolicyManager(lap->apm); - if (lap->service_client) - delete lap->service_client; - free(lap); - return 0; -} - -static int legacy_ap_dev_close(hw_device_t* device) -{ - if (device) - free(device); - return 0; -} - -static int legacy_ap_dev_open(const hw_module_t* module, const char* name, - hw_device_t** device) -{ - struct legacy_ap_device *dev; - - if (strcmp(name, AUDIO_POLICY_INTERFACE) != 0) - return -EINVAL; - - dev = (struct legacy_ap_device *)calloc(1, sizeof(*dev)); - if (!dev) - return -ENOMEM; - - dev->device.common.tag = HARDWARE_DEVICE_TAG; - dev->device.common.version = 0; - dev->device.common.module = const_cast<hw_module_t*>(module); - dev->device.common.close = legacy_ap_dev_close; - dev->device.create_audio_policy = create_legacy_ap; - dev->device.destroy_audio_policy = destroy_legacy_ap; - - *device = &dev->device.common; - - return 0; -} - -static struct hw_module_methods_t legacy_ap_module_methods = { - open: legacy_ap_dev_open -}; - -struct legacy_ap_module HAL_MODULE_INFO_SYM = { - module: { - common: { - tag: HARDWARE_MODULE_TAG, - version_major: 1, - version_minor: 0, - id: AUDIO_POLICY_HARDWARE_MODULE_ID, - name: "LEGACY Audio Policy HAL", - author: "The Android Open Source Project", - methods: &legacy_ap_module_methods, - dso : NULL, - reserved : {0}, - }, - }, -}; - -}; // extern "C" - -}; // namespace android_audio_legacy diff --git a/configs/melfas_ts.idc b/configs/melfas_ts.idc new file mode 100755 index 0000000..703e73b --- /dev/null +++ b/configs/melfas_ts.idc @@ -0,0 +1,24 @@ +# Copyright (C) 2010 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. + +# +# Input Device Configuration File for the Atmel Maxtouch touch screen. +# +# These calibration values are derived from empirical measurements +# and may not be appropriate for use with other touch screens. +# Refer to the input device configuration documentation for more details. +# + +# Basic Parameters +touch.deviceType = touchScreen diff --git a/configs/sec_touchscreen.idc b/configs/sec_touchscreen.idc index d258f04..4bd511a 100755 --- a/configs/sec_touchscreen.idc +++ b/configs/sec_touchscreen.idc @@ -13,11 +13,11 @@ # limitations under the License. # -# Input Device Calibration File for the Crespo touch screen. +# Input Device Configuration File for the Atmel Maxtouch touch screen. # # These calibration values are derived from empirical measurements # and may not be appropriate for use with other touch screens. -# Refer to the input device calibration documentation for more details. +# Refer to the input device configuration documentation for more details. # # Basic Parameters @@ -31,26 +31,26 @@ touch.touchSize.calibration = pressure # Driver reports tool size as an area measurement. # # Based on empirical measurements, we estimate the size of the tool -# using size = sqrt(22 * rawToolArea + 0) * 9.2 + 0. +# using size = sqrt(22 * rawToolArea + 0) * 6 + 0. touch.toolSize.calibration = area touch.toolSize.areaScale = 22 touch.toolSize.areaBias = 0 -touch.toolSize.linearScale = 9.2 +touch.toolSize.linearScale = 6 touch.toolSize.linearBias = 0 touch.toolSize.isSummed = 0 # Pressure # Driver reports signal strength as pressure. # -# A normal thumb touch typically registers about 100 signal strength +# A normal index finger touch typically registers about 80 signal strength # units although we don't expect these values to be accurate. touch.pressure.calibration = amplitude touch.pressure.source = default -touch.pressure.scale = 0.01 +touch.pressure.scale = 0.0125 # Size touch.size.calibration = normalized # Orientation -touch.orientation.calibration = none +touch.orientation.calibration = vector diff --git a/configs/sec_ts_ics_bio.idc b/configs/sec_ts_ics_bio.idc deleted file mode 100755 index eb46d0a..0000000 --- a/configs/sec_ts_ics_bio.idc +++ /dev/null @@ -1,56 +0,0 @@ -# Copyright (C) 2010 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.
-
-#
-# Input Device Calibration File for the Crespo touch screen.
-#
-# These calibration values are derived from empirical measurements
-# and may not be appropriate for use with other touch screens.
-# Refer to the input device calibration documentation for more details.
-#
-
-# Basic Parameters
-touch.deviceType = touchScreen
-touch.orientationAware = 1
-
-# Touch Size
-touch.touchSize.calibration = pressure
-
-# Tool Size
-# Driver reports tool size as an area measurement.
-#
-# Based on empirical measurements, we estimate the size of the tool
-# using size = sqrt(22 * rawToolArea + 0) * 9.2 + 0.
-touch.toolSize.calibration = area
-touch.toolSize.areaScale = 22
-touch.toolSize.areaBias = 0
-touch.toolSize.linearScale = 9.2
-touch.toolSize.linearBias = 0
-touch.toolSize.isSummed = 0
-
-# Pressure
-# Driver reports signal strength as pressure.
-#
-# A normal thumb touch typically registers about 100 signal strength
-# units although we don't expect these values to be accurate.
-touch.pressure.calibration = amplitude
-touch.pressure.source = default
-touch.pressure.scale = 0.01
-
-# Size
-touch.size.calibration = normalized
-
-# Orientation
-touch.orientation.calibration = none
-
diff --git a/galaxynote.mk b/galaxynote.mk index 1feabf8..679a028 100755 --- a/galaxynote.mk +++ b/galaxynote.mk @@ -35,12 +35,19 @@ PRODUCT_COPY_FILES += \ # soundbooster.txt - needs to be at /data/soundbooster.txt PRODUCT_COPY_FILES += \ device/samsung/galaxynote/configs/asound.conf:system/etc/asound.conf \ + device/samsung/galaxynote/configs/audio_effects.conf:system/etc/audio_effects.conf \ device/samsung/galaxynote/configs/soundbooster.txt:system/etc/audio/soundbooster.txt +# omx +PRODUCT_COPY_FILES += \ + device/samsung/galaxys2/configs/media_profiles.xml:system/etc/media_profiles.xml \ + device/samsung/galaxys2/configs/secomxregistry:system/etc/secomxregistry \ + device/samsung/galaxys2/configs/somxreg.conf:system/etc/somxreg.conf + # Touchscreen PRODUCT_COPY_FILES += \ - device/samsung/galaxynote/configs/sec_ts_ics_bio.idc:system/usr/idc/sec_ts_ics_bio.idc \ - device/samsung/galaxynote/configs/sec_ts_ics_bio.idc:system/usr/idc/sec_touchscreen.idc \ + device/samsung/galaxynote/configs/melfas_ts.idc:system/usr/idc/melfas_ts.idc \ + device/samsung/galaxynote/configs/sec_touchscreen.idc:system/usr/idc/sec_touchscreen.idc \ device/samsung/galaxynote/configs/sec_ts_ics_bio.idc:system/usr/idc/sec_e-pen.idc \ # Keylayout @@ -78,10 +85,6 @@ PRODUCT_COPY_FILES += \ # Packages PRODUCT_PACKAGES := \ - audio.primary.smdkv310 \ - audio_policy.smdkv310 \ - gps.smdkv310 \ - smdkv310_hdcp_keys \ com.android.future.usb.accessory # Charger @@ -94,15 +97,12 @@ PRODUCT_PACKAGES += \ Camera # Sensors -PRODUCT_PACKAGES += \ - lights.smdkv310 \ - sensors.smdkv310 +# PRODUCT_PACKAGES += \ +# sensors.smdkv310 # Ril PRODUCT_PROPERTY_OVERRIDES += \ ro.telephony.ril_class=samsung \ - ro.telephony.ril.v3=icccardstatus,datacall,signalstrength,facilitylock \ - ro.telephony.sends_barcount=1 \ mobiledata.interfaces=pdp0,wlan0,gprs,ppp0 # Filesystem management tools diff --git a/gpswrapper/Android.mk b/gpswrapper/Android.mk index c13be1b..e0b4e2b 100755 --- a/gpswrapper/Android.mk +++ b/gpswrapper/Android.mk @@ -3,7 +3,7 @@ include $(CLEAR_VARS) LOCAL_MODULE_TAGS := optional
-LOCAL_MODULE := gps.smdkv310
+LOCAL_MODULE := gps.$(TARGET_BOARD_PLATFORM)
LOCAL_SHARED_LIBRARIES:= \
liblog \
diff --git a/gpswrapper/gps.c b/gpswrapper/gps.c index d921e27..a4e8f7f 100755 --- a/gpswrapper/gps.c +++ b/gpswrapper/gps.c @@ -31,7 +31,7 @@ #define LOG_TAG "gps-wrapper"
#include <utils/Log.h>
-#define ORIGINAL_HAL_PATH "/system/lib/hw/vendor-gps.smdkv310.so"
+#define ORIGINAL_HAL_PATH "/system/lib/hw/vendor-gps.s5pc210.so"
static const AGpsRilInterface* oldAGPSRIL = NULL;
static AGpsRilInterface newAGPSRIL;
diff --git a/init.smdkv310.rc b/init.smdk4210.rc index a2cacaa..1bac747 100755 --- a/init.smdkv310.rc +++ b/init.smdk4210.rc @@ -126,21 +126,11 @@ on fs chown system system /sys/devices/platform/android_usb/tethering
# setup for alsa snd device
- symlink /dev/snd/pcmC0D0c /dev/pcmC0D0c
- symlink /dev/snd/pcmC0D0p /dev/pcmC0D0p
- symlink /dev/snd/controlC0 /dev/controlC0
- symlink /dev/snd/timer /dev/timer
- symlink /dev/snd/hwC0D0 /dev/hwC0D0
- chmod 0777 /dev/pcmC0D0c
- chmod 0777 /dev/pcmC0D0p
- chmod 0777 /dev/controlC0
- chmod 0777 /dev/timer
- chmod 0777 /dev/hwC0D0
- chmod 0777 /dev/snd/pcmC0D0c
- chmod 0777 /dev/snd/pcmC0D0p
- chmod 0777 /dev/snd/controlC0
- chmod 0777 /dev/snd/timer
- chmod 0777 /dev/snd/hwC0D0
+ chmod 0770 /dev/snd/pcmC0D0c
+ chmod 0770 /dev/snd/pcmC0D0p
+ chmod 0770 /dev/snd/controlC0
+ chmod 0770 /dev/snd/timer
+ chmod 0770 /dev/snd/hwC0D0
# Permissions for gpio_keys
chown radio system /sys/devices/platform/sec_key.0/wakeup_keys
diff --git a/init.smdkv310.usb.rc b/init.smdk4210.usb.rc Binary files differindex 9307c56..9307c56 100755 --- a/init.smdkv310.usb.rc +++ b/init.smdk4210.usb.rc diff --git a/modules/Si4709_driver.ko b/modules/Si4709_driver.ko Binary files differindex a94df4d..3de78b2 100755 --- a/modules/Si4709_driver.ko +++ b/modules/Si4709_driver.ko diff --git a/modules/cifs.ko b/modules/cifs.ko Binary files differindex d597f57..424aa09 100755 --- a/modules/cifs.ko +++ b/modules/cifs.ko diff --git a/modules/gspca_main.ko b/modules/gspca_main.ko Binary files differindex 58fef5b..24381a0 100755 --- a/modules/gspca_main.ko +++ b/modules/gspca_main.ko diff --git a/modules/j4fs.ko b/modules/j4fs.ko Binary files differindex 65cf1cc..ca2fefa 100755 --- a/modules/j4fs.ko +++ b/modules/j4fs.ko diff --git a/modules/scsi_wait_scan.ko b/modules/scsi_wait_scan.ko Binary files differindex b7479d8..46724ab 100755 --- a/modules/scsi_wait_scan.ko +++ b/modules/scsi_wait_scan.ko diff --git a/modules/vibrator.ko b/modules/vibrator.ko Binary files differindex b41e9f7..9ebf564 100755 --- a/modules/vibrator.ko +++ b/modules/vibrator.ko diff --git a/ueventd.smdkv310.rc b/ueventd.smdk4210.rc index 14c1481..14c1481 100755 --- a/ueventd.smdkv310.rc +++ b/ueventd.smdk4210.rc |