diff options
author | mauimauer <sebastian@n-unity.de> | 2011-12-20 17:21:14 +0100 |
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committer | mauimauer <sebastian@n-unity.de> | 2011-12-20 17:21:14 +0100 |
commit | 4bb3b01048c4cff240f4f6ccdb680d855e43eab7 (patch) | |
tree | 038acdff2c1a674a76bc20d8e9ce296c5656a424 | |
parent | d28b97d9cdecfda13c3f912ac64c536c17e7dbd0 (diff) | |
download | device_samsung_n7000-4bb3b01048c4cff240f4f6ccdb680d855e43eab7.zip device_samsung_n7000-4bb3b01048c4cff240f4f6ccdb680d855e43eab7.tar.gz device_samsung_n7000-4bb3b01048c4cff240f4f6ccdb680d855e43eab7.tar.bz2 |
Conf
77 files changed, 4751 insertions, 62 deletions
diff --git a/BoardConfig.mk b/BoardConfig.mk index cd6657d..f18ef7f 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 := s5pc210 -TARGET_BOOTLOADER_BOARD_NAME := smdk4210 +TARGET_BOARD_PLATFORM := smdkv310 +TARGET_BOOTLOADER_BOARD_NAME := GT-N7000 TARGET_BOARD_INFO_FILE := device/samsung/galaxynote/board-info.txt TARGET_NO_BOOTLOADER := true @@ -60,10 +60,9 @@ 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_USE_SECTVOUT := true +BOARD_USES_LEGACY_EGL := true +COMMON_GLOBAL_CFLAGS += -DMISSING_EGL_EXTERNAL_IMAGE -DMISSING_EGL_PIXEL_FORMAT_YV12 -DMISSING_GRALLOC_BUFFERS # Audio BOARD_USE_YAMAHAPLAYER := true diff --git a/audio/Android.mk b/audio/Android.mk new file mode 100755 index 0000000..5026c54 --- /dev/null +++ b/audio/Android.mk @@ -0,0 +1,103 @@ +# +# 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 new file mode 100755 index 0000000..53ae657 --- /dev/null +++ b/audio/AudioPolicyCompatClient.cpp @@ -0,0 +1,142 @@ +/* + * 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 new file mode 100755 index 0000000..4feee37 --- /dev/null +++ b/audio/AudioPolicyCompatClient.h @@ -0,0 +1,79 @@ +/* + * 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 new file mode 100755 index 0000000..78f87da --- /dev/null +++ b/audio/AudioPolicyInterface.h @@ -0,0 +1,238 @@ +/* + * 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 new file mode 100755 index 0000000..0c3c119 --- /dev/null +++ b/audio/AudioPolicyManagerBase.cpp @@ -0,0 +1,2439 @@ +/* + * 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() bad bad bad. 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: +#ifdef HAVE_FM_RADIO + case AudioSystem::FM: +#endif + 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: + LOGV("getDeviceForInputSource() CALL/UPLINK/DOWNLINK returning DEVICE_IN_VOICE_CALL"); + device = AudioSystem::DEVICE_IN_BUILTIN_MIC; + break; + default: + LOGW("getDeviceForInputSource() invalid input source %d", inputSource); + //device = 0; + device = AudioSystem::DEVICE_IN_BUILTIN_MIC; + 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 new file mode 100755 index 0000000..3ab748d --- /dev/null +++ b/audio/AudioPolicyManagerBase.h @@ -0,0 +1,393 @@ +/* + * 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 new file mode 100755 index 0000000..0b726d2 --- /dev/null +++ b/audio/AudioPolicyManagerDefault.cpp @@ -0,0 +1,34 @@ +/* + * 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 new file mode 100755 index 0000000..b2b2576 --- /dev/null +++ b/audio/AudioPolicyManagerDefault.h @@ -0,0 +1,43 @@ +/* + * 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 new file mode 100755 index 0000000..bd75a73 --- /dev/null +++ b/audio/AudioSystem.h @@ -0,0 +1,562 @@ +/* + * 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 new file mode 100755 index 0000000..e69de29 --- /dev/null +++ b/audio/MODULE_LICENSE_APACHE2 diff --git a/audio/NOTICE b/audio/NOTICE new file mode 100755 index 0000000..3237da6 --- /dev/null +++ b/audio/NOTICE @@ -0,0 +1,190 @@ + + 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. Definitions. + + "License" shall mean the terms and conditions for use, reproduction, + and distribution as defined by Sections 1 through 9 of this document. + + "Licensor" shall mean the copyright owner or entity authorized by + the copyright owner that is granting the License. + + "Legal Entity" shall mean the union of the acting entity and all + other entities that control, are controlled by, or are under common + control with that entity. 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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 deleted file mode 100755 index 703e73b..0000000 --- a/configs/melfas_ts.idc +++ /dev/null @@ -1,24 +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 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 4bd511a..d258f04 100755 --- a/configs/sec_touchscreen.idc +++ b/configs/sec_touchscreen.idc @@ -13,11 +13,11 @@ # limitations under the License. # -# Input Device Configuration File for the Atmel Maxtouch touch screen. +# 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 configuration documentation for more details. +# Refer to the input device calibration 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) * 6 + 0. +# using size = sqrt(22 * rawToolArea + 0) * 9.2 + 0. touch.toolSize.calibration = area touch.toolSize.areaScale = 22 touch.toolSize.areaBias = 0 -touch.toolSize.linearScale = 6 +touch.toolSize.linearScale = 9.2 touch.toolSize.linearBias = 0 touch.toolSize.isSummed = 0 # Pressure # Driver reports signal strength as pressure. # -# A normal index finger touch typically registers about 80 signal strength +# 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 = 0.0125 +touch.pressure.scale = 0.01 # Size touch.size.calibration = normalized # Orientation -touch.orientation.calibration = vector +touch.orientation.calibration = none diff --git a/configs/sec_ts_ics_bio.idc b/configs/sec_ts_ics_bio.idc new file mode 100755 index 0000000..eb46d0a --- /dev/null +++ b/configs/sec_ts_ics_bio.idc @@ -0,0 +1,56 @@ +# 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 a867401..6845c66 100755 --- a/galaxynote.mk +++ b/galaxynote.mk @@ -21,10 +21,10 @@ PRODUCT_LOCALES += xhdpi # Init files PRODUCT_COPY_FILES := \ device/samsung/galaxynote/lpm.rc:root/lpm.rc \ - device/samsung/galaxynote/init.smdk4210.usb.rc:root/init.smdk4210.usb.rc \ + device/samsung/galaxynote/init.smdkv310.usb.rc:root/init.smdkv310.usb.rc \ device/samsung/galaxynote/init.smdkc210.rc:root/init.smdkc210.rc \ - device/samsung/galaxynote/init.smdk4210.rc:root/init.smdk4210.rc \ - device/samsung/galaxynote/ueventd.smdk4210.rc:root/ueventd.smdk4210.rc + device/samsung/galaxynote/init.smdkv310.rc:root/init.smdkv310.rc \ + device/samsung/galaxynote/ueventd.smdkv310.rc:root/ueventd.smdkv310.rc # APNs - REMOVE IF VENDOR CYANOGEN IS BACK PRODUCT_COPY_FILES += \ @@ -35,19 +35,12 @@ 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/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_ts_ics_bio.idc \ + device/samsung/galaxynote/configs/sec_ts_ics_bio.idc:system/usr/idc/sec_touchscreen.idc \ device/samsung/galaxynote/configs/sec_ts_ics_bio.idc:system/usr/idc/sec_e-pen.idc \ # Keylayout @@ -85,6 +78,10 @@ PRODUCT_COPY_FILES += \ # Packages PRODUCT_PACKAGES := \ +# audio.primary.smdkv310 \ +# audio_policy.smdkv310 \ + gps.smdkv310 \ + smdkv310_hdcp_keys \ com.android.future.usb.accessory # Charger @@ -97,12 +94,15 @@ PRODUCT_PACKAGES += \ Camera # Sensors -# PRODUCT_PACKAGES += \ -# sensors.smdkv310 - +PRODUCT_PACKAGES += \ + lights.smdkv310 \ + 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 @@ -129,6 +129,7 @@ PRODUCT_PACKAGES += \ # These are the hardware-specific features PRODUCT_COPY_FILES += \ + frameworks/base/data/etc/android.hardware.audio.low_latency.xml:system/etc/permissions/android.hardware.audio.low_latency.xml \ frameworks/base/data/etc/handheld_core_hardware.xml:system/etc/permissions/handheld_core_hardware.xml \ frameworks/base/data/etc/android.hardware.camera.flash-autofocus.xml:system/etc/permissions/android.hardware.camera.flash-autofocus.xml \ frameworks/base/data/etc/android.hardware.camera.front.xml:system/etc/permissions/android.hardware.camera.front.xml \ @@ -167,7 +168,7 @@ PRODUCT_DEFAULT_PROPERTY_OVERRIDES += \ # kernel modules for ramdisk RAMDISK_MODULES := $(addprefix device/samsung/galaxynote/modules/, gspca_main.ko j4fs.ko \ - scsi_wait_scan.ko Si4709_driver.ko vibrator.ko) + scsi_wait_scan.ko vibrator.ko) PRODUCT_COPY_FILES += $(foreach module,\ $(RAMDISK_MODULES),\ $(module):root/lib/modules/$(notdir $(module))) @@ -193,4 +194,4 @@ PRODUCT_COPY_FILES += \ $(call inherit-product, frameworks/base/build/phone-xhdpi-1024-dalvik-heap.mk) $(call inherit-product-if-exists, vendor/samsung/galaxynote/galaxynote-vendor.mk) -$(call inherit-product-if-exists, hardware/broadcom/wlan/bcmdhd/firmware/bcm4330/device-bcm.mk) +$(call inherit-product-if-exists, hardware/broadcom/wlan/bcmdhd/firmware/bcm4330/device-bcm.mk)
\ No newline at end of file diff --git a/gpswrapper/Android.mk b/gpswrapper/Android.mk index e0b4e2b..c13be1b 100755 --- a/gpswrapper/Android.mk +++ b/gpswrapper/Android.mk @@ -3,7 +3,7 @@ include $(CLEAR_VARS) LOCAL_MODULE_TAGS := optional
-LOCAL_MODULE := gps.$(TARGET_BOARD_PLATFORM)
+LOCAL_MODULE := gps.smdkv310
LOCAL_SHARED_LIBRARIES:= \
liblog \
diff --git a/gpswrapper/gps.c b/gpswrapper/gps.c index a4e8f7f..d921e27 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.s5pc210.so"
+#define ORIGINAL_HAL_PATH "/system/lib/hw/vendor-gps.smdkv310.so"
static const AGpsRilInterface* oldAGPSRIL = NULL;
static AGpsRilInterface newAGPSRIL;
diff --git a/init.smdk4210.rc b/init.smdkv310.rc index 1bac747..069621e 100755 --- a/init.smdk4210.rc +++ b/init.smdkv310.rc @@ -126,11 +126,21 @@ on fs chown system system /sys/devices/platform/android_usb/tethering
# setup for alsa snd device
- 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
+ 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
# Permissions for gpio_keys
chown radio system /sys/devices/platform/sec_key.0/wakeup_keys
@@ -253,7 +263,7 @@ on fs chown radio system /sys/devices/virtual/net/svnet0/waketime
#bluetooth
- setprop ro.bt.bdaddr_path "/efs/bluetooth/bt_addr"
+ #setprop ro.bt.bdaddr_path "/efs/bluetooth/bt_addr"
chown bluetooth bluetooth ro.bt.bdaddr_path
chown bluetooth bluetooth /dev/s3c2410_serial0
chmod 0660 /dev/s3c2410_serial0
@@ -327,7 +337,6 @@ service hciattach /system/bin/logwrapper /system/bin/brcm_patchram_plus --enable service bdaddr /system/bin/bdaddr_read
class main
user root
- disabled
oneshot
# tvout
diff --git a/init.smdk4210.usb.rc b/init.smdkv310.usb.rc Binary files differindex 9307c56..9307c56 100755 --- a/init.smdk4210.usb.rc +++ b/init.smdkv310.usb.rc diff --git a/modules/Si4709_driver.ko b/modules/Si4709_driver.ko Binary files differindex 3de78b2..d99fb76 100755 --- a/modules/Si4709_driver.ko +++ b/modules/Si4709_driver.ko diff --git a/modules/bthid.ko b/modules/bthid.ko Binary files differindex 605094c..614169d 100644..100755 --- a/modules/bthid.ko +++ b/modules/bthid.ko diff --git a/modules/cifs.ko b/modules/cifs.ko Binary files differindex 424aa09..68c3307 100755 --- a/modules/cifs.ko +++ b/modules/cifs.ko diff --git a/modules/gspca_main.ko b/modules/gspca_main.ko Binary files differindex 24381a0..feb8b3e 100755 --- a/modules/gspca_main.ko +++ b/modules/gspca_main.ko diff --git a/modules/j4fs.ko b/modules/j4fs.ko Binary files differindex ca2fefa..18e27ad 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 46724ab..661365d 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 9ebf564..9b2e043 100755 --- a/modules/vibrator.ko +++ b/modules/vibrator.ko diff --git a/overlay/packages/apps/Settings/res/values-af/arrays.xml b/overlay/packages/apps/Settings/res/values-af/arrays.xml index 4f2780f..4f2780f 100644..100755 --- a/overlay/packages/apps/Settings/res/values-af/arrays.xml +++ b/overlay/packages/apps/Settings/res/values-af/arrays.xml diff --git a/overlay/packages/apps/Settings/res/values-am/arrays.xml b/overlay/packages/apps/Settings/res/values-am/arrays.xml index 73deb95..73deb95 100644..100755 --- a/overlay/packages/apps/Settings/res/values-am/arrays.xml +++ b/overlay/packages/apps/Settings/res/values-am/arrays.xml diff --git a/overlay/packages/apps/Settings/res/values-ar/arrays.xml b/overlay/packages/apps/Settings/res/values-ar/arrays.xml index 396b12e..396b12e 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a/overlay/packages/apps/Settings/res/values/arrays.xml +++ b/overlay/packages/apps/Settings/res/values/arrays.xml diff --git a/ueventd.smdk4210.rc b/ueventd.smdkv310.rc index 14c1481..14c1481 100755 --- a/ueventd.smdk4210.rc +++ b/ueventd.smdkv310.rc |