| /* |
| * 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 <media/mediarecorder.h> |
| |
| namespace android { |
| |
| |
| // ---------------------------------------------------------------------------- |
| // 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); |
| #ifdef WITH_A2DP |
| if (mA2dpOutput != 0 && |
| mPhoneState != AudioSystem::MODE_NORMAL) { |
| mpClientInterface->suspendOutput(mA2dpOutput); |
| } |
| #endif |
| } |
| } |
| 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 = ""; |
| #ifdef WITH_A2DP |
| if (mA2dpOutput != 0 && |
| mPhoneState != AudioSystem::MODE_NORMAL) { |
| mpClientInterface->restoreOutput(mA2dpOutput); |
| } |
| #endif |
| } |
| } |
| } 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 |
| checkOutputForAllStrategies(newDevice); |
| // 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 (mPhoneState == AudioSystem::MODE_IN_CALL) { |
| 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 ((oldState != AudioSystem::MODE_IN_CALL) && (state == AudioSystem::MODE_IN_CALL)) { |
| 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 ((oldState == AudioSystem::MODE_IN_CALL) && (state != AudioSystem::MODE_IN_CALL)) { |
| LOGV(" Exiting call in setPhoneState()"); |
| // force routing command to audio hardware when exiting a call |
| // 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 |
| checkOutputForAllStrategies(newDevice); |
| // suspend A2DP output if a SCO device is present. |
| if (mA2dpOutput != 0 && mScoDeviceAddress != "") { |
| if (oldState == AudioSystem::MODE_NORMAL) { |
| mpClientInterface->suspendOutput(mA2dpOutput); |
| } else if (state == AudioSystem::MODE_NORMAL) { |
| mpClientInterface->restoreOutput(mA2dpOutput); |
| } |
| } |
| #endif |
| updateDeviceForStrategy(); |
| |
| AudioOutputDescriptor *hwOutputDesc = mOutputs.valueFor(mHardwareOutput); |
| |
| // force routing command to audio hardware when ending call |
| // even if no device change is needed |
| if (oldState == AudioSystem::MODE_IN_CALL && 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 (state == AudioSystem::MODE_IN_CALL && 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 (state == AudioSystem::MODE_IN_CALL) { |
| 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 && |
| (hwOutputDesc->mRefCount[AudioSystem::MUSIC] || |
| (systemTime() - mMusicStopTime) < seconds(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; |
| } |
| 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_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) { |
| 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 |
| checkOutputForAllStrategies(newDevice); |
| #endif |
| updateDeviceForStrategy(); |
| setOutputDevice(mHardwareOutput, newDevice); |
| if (forceVolumeReeval) { |
| applyStreamVolumes(mHardwareOutput, newDevice); |
| } |
| } |
| |
| 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; |
| 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) |
| { |
| LOGV("startOutput() output %d, stream %d", output, stream); |
| 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) { |
| 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); |
| |
| setOutputDevice(output, getNewDevice(output)); |
| |
| // handle special case for sonification while in call |
| if (mPhoneState == AudioSystem::MODE_IN_CALL) { |
| handleIncallSonification(stream, true, false); |
| } |
| |
| // apply volume rules for current stream and device if necessary |
| checkAndSetVolume(stream, mStreams[stream].mIndexCur, output, outputDesc->device()); |
| |
| return NO_ERROR; |
| } |
| |
| status_t AudioPolicyManagerBase::stopOutput(audio_io_handle_t output, AudioSystem::stream_type stream) |
| { |
| LOGV("stopOutput() output %d, stream %d", output, stream); |
| 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 (mPhoneState == AudioSystem::MODE_IN_CALL) { |
| 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 last music track was stopped - see computeVolume() |
| if (stream == AudioSystem::MUSIC) { |
| mMusicStopTime = systemTime(); |
| } |
| |
| setOutputDevice(output, getNewDevice(output)); |
| |
| #ifdef WITH_A2DP |
| if (mA2dpOutput != 0 && !a2dpUsedForSonification() && strategy == STRATEGY_SONIFICATION) { |
| setStrategyMute(STRATEGY_MEDIA, false, mA2dpOutput, mOutputs.valueFor(mHardwareOutput)->mLatency*2); |
| } |
| #endif |
| if (output != mHardwareOutput) { |
| setOutputDevice(mHardwareOutput, getNewDevice(mHardwareOutput), true); |
| } |
| return NO_ERROR; |
| } else { |
| LOGW("stopOutput() refcount is already 0 for output %d", output); |
| return INVALID_OPERATION; |
| } |
| } |
| |
| void AudioPolicyManagerBase::releaseOutput(audio_io_handle_t output) |
| { |
| LOGV("releaseOutput() %d", output); |
| ssize_t index = mOutputs.indexOfKey(output); |
| if (index < 0) { |
| LOGW("releaseOutput() releasing unknown output %d", output); |
| return; |
| } |
| |
| #ifdef AUDIO_POLICY_TEST |
| int testIndex = testOutputIndex(output); |
| if (testIndex != 0) { |
| AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index); |
| if (outputDesc->refCount() == 0) { |
| mpClientInterface->closeOutput(output); |
| delete mOutputs.valueAt(index); |
| mOutputs.removeItem(output); |
| mTestOutputs[testIndex] = 0; |
| } |
| return; |
| } |
| #endif //AUDIO_POLICY_TEST |
| |
| if (mOutputs.valueAt(index)->mFlags & AudioSystem::OUTPUT_FLAG_DIRECT) { |
| mpClientInterface->closeOutput(output); |
| delete mOutputs.valueAt(index); |
| mOutputs.removeItem(output); |
| } |
| } |
| |
| audio_io_handle_t AudioPolicyManagerBase::getInput(int inputSource, |
| uint32_t samplingRate, |
| uint32_t format, |
| uint32_t channels, |
| AudioSystem::audio_in_acoustics acoustics) |
| { |
| audio_io_handle_t input = 0; |
| uint32_t device = getDeviceForInputSource(inputSource); |
| |
| LOGV("getInput() inputSource %d, samplingRate %d, format %d, channels %x, acoustics %x", inputSource, samplingRate, format, channels, acoustics); |
| |
| if (device == 0) { |
| return 0; |
| } |
| |
| // adapt channel selection to input source |
| switch(inputSource) { |
| case AUDIO_SOURCE_VOICE_UPLINK: |
| channels = AudioSystem::CHANNEL_IN_VOICE_UPLINK; |
| break; |
| case AUDIO_SOURCE_VOICE_DOWNLINK: |
| channels = AudioSystem::CHANNEL_IN_VOICE_DNLINK; |
| break; |
| case AUDIO_SOURCE_VOICE_CALL: |
| channels = (AudioSystem::CHANNEL_IN_VOICE_UPLINK | AudioSystem::CHANNEL_IN_VOICE_DNLINK); |
| break; |
| default: |
| break; |
| } |
| |
| AudioInputDescriptor *inputDesc = new AudioInputDescriptor(); |
| |
| inputDesc->mInputSource = inputSource; |
| inputDesc->mDevice = device; |
| inputDesc->mSamplingRate = samplingRate; |
| inputDesc->mFormat = format; |
| inputDesc->mChannels = channels; |
| inputDesc->mAcoustics = acoustics; |
| inputDesc->mRefCount = 0; |
| input = mpClientInterface->openInput(&inputDesc->mDevice, |
| &inputDesc->mSamplingRate, |
| &inputDesc->mFormat, |
| &inputDesc->mChannels, |
| inputDesc->mAcoustics); |
| |
| // only accept input with the exact requested set of parameters |
| if (input == 0 || |
| (samplingRate != inputDesc->mSamplingRate) || |
| (format != inputDesc->mFormat) || |
| (channels != inputDesc->mChannels)) { |
| LOGV("getInput() failed opening input: samplingRate %d, format %d, channels %d", |
| samplingRate, format, channels); |
| if (input != 0) { |
| mpClientInterface->closeInput(input); |
| } |
| delete inputDesc; |
| return 0; |
| } |
| mInputs.add(input, inputDesc); |
| return input; |
| } |
| |
| status_t AudioPolicyManagerBase::startInput(audio_io_handle_t input) |
| { |
| LOGV("startInput() input %d", input); |
| ssize_t index = mInputs.indexOfKey(input); |
| if (index < 0) { |
| LOGW("startInput() unknow input %d", input); |
| return BAD_VALUE; |
| } |
| AudioInputDescriptor *inputDesc = mInputs.valueAt(index); |
| |
| #ifdef AUDIO_POLICY_TEST |
| if (mTestInput == 0) |
| #endif //AUDIO_POLICY_TEST |
| { |
| // refuse 2 active AudioRecord clients at the same time |
| if (getActiveInput() != 0) { |
| LOGW("startInput() input %d failed: other input already started", input); |
| return INVALID_OPERATION; |
| } |
| } |
| |
| AudioParameter param = AudioParameter(); |
| param.addInt(String8(AudioParameter::keyRouting), (int)inputDesc->mDevice); |
| |
| // use Voice Recognition mode or not for this input based on input source |
| int vr_enabled = inputDesc->mInputSource == AUDIO_SOURCE_VOICE_RECOGNITION ? 1 : 0; |
| param.addInt(String8("vr_mode"), vr_enabled); |
| LOGV("AudioPolicyManager::startInput(%d), setting vr_mode to %d", inputDesc->mInputSource, vr_enabled); |
| |
| 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; |
| } |
| |
| 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)); |
| } |
| |
| return NO_ERROR; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // AudioPolicyManagerBase |
| // ---------------------------------------------------------------------------- |
| |
| AudioPolicyManagerBase::AudioPolicyManagerBase(AudioPolicyClientInterface *clientInterface) |
| : |
| #ifdef AUDIO_POLICY_TEST |
| Thread(false), |
| #endif //AUDIO_POLICY_TEST |
| mPhoneState(AudioSystem::MODE_NORMAL), mRingerMode(0), mMusicStopTime(0), mLimitRingtoneVolume(false) |
| { |
| mpClientInterface = clientInterface; |
| |
| for (int i = 0; i < AudioSystem::NUM_FORCE_USE; i++) { |
| mForceUse[i] = AudioSystem::FORCE_NONE; |
| } |
| |
| // 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); |
| } |
| |
| updateDeviceForStrategy(); |
| #ifdef AUDIO_POLICY_TEST |
| 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(); |
| } |
| |
| #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); |
| // 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 (mScoDeviceAddress != "") { |
| // It is normal to suspend twice if we are both in call, |
| // and have the hardware audio output routed to BT SCO |
| if (mPhoneState != AudioSystem::MODE_NORMAL) { |
| mpClientInterface->suspendOutput(mA2dpOutput); |
| } |
| if (AudioSystem::isBluetoothScoDevice((AudioSystem::audio_devices)hwOutputDesc->device())) { |
| mpClientInterface->suspendOutput(mA2dpOutput); |
| } |
| } |
| |
| if (!a2dpUsedForSonification()) { |
| // mute music on A2DP output if a notification or ringtone is playing |
| uint32_t refCount = hwOutputDesc->strategyRefCount(STRATEGY_SONIFICATION); |
| for (uint32_t i = 0; i < refCount; i++) { |
| setStrategyMute(STRATEGY_MEDIA, true, mA2dpOutput); |
| } |
| } |
| 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); |
| for (uint32_t i = 0; i < refCount; i++) { |
| setStrategyMute(STRATEGY_MEDIA, false, mA2dpOutput); |
| } |
| } |
| mA2dpDeviceAddress = ""; |
| 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 &newDevice) |
| { |
| 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)); |
| AudioOutputDescriptor *hwOutputDesc = mOutputs.valueFor(mHardwareOutput); |
| AudioOutputDescriptor *a2dpOutputDesc; |
| |
| if (a2dpWasUsed && !a2dpIsUsed) { |
| bool dupUsed = a2dpUsedForSonification() && a2dpWasUsed && (AudioSystem::popCount(prevDevice) == 2); |
| |
| if (dupUsed) { |
| LOGV("checkOutputForStrategy() moving strategy %d to duplicated", strategy); |
| a2dpOutputDesc = mOutputs.valueFor(mDuplicatedOutput); |
| } else { |
| LOGV("checkOutputForStrategy() moving strategy %d to a2dp", strategy); |
| a2dpOutputDesc = mOutputs.valueFor(mA2dpOutput); |
| } |
| |
| for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) { |
| if (getStrategy((AudioSystem::stream_type)i) == strategy) { |
| mpClientInterface->setStreamOutput((AudioSystem::stream_type)i, mHardwareOutput); |
| } |
| } |
| // do not change newDevice if it was already set before this call by a previous call to |
| // getNewDevice() or checkOutputForStrategy() for a strategy with higher priority |
| if (newDevice == 0 && hwOutputDesc->isUsedByStrategy(strategy)) { |
| newDevice = getDeviceForStrategy(strategy, false); |
| } |
| } |
| if (a2dpIsUsed && !a2dpWasUsed) { |
| bool dupUsed = a2dpUsedForSonification() && a2dpIsUsed && (AudioSystem::popCount(curDevice) == 2); |
| audio_io_handle_t a2dpOutput; |
| |
| if (dupUsed) { |
| LOGV("checkOutputForStrategy() moving strategy %d from duplicated", strategy); |
| a2dpOutputDesc = mOutputs.valueFor(mDuplicatedOutput); |
| a2dpOutput = mDuplicatedOutput; |
| } else { |
| LOGV("checkOutputForStrategy() moving strategy %d from a2dp", strategy); |
| a2dpOutputDesc = mOutputs.valueFor(mA2dpOutput); |
| a2dpOutput = mA2dpOutput; |
| } |
| |
| for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) { |
| if (getStrategy((AudioSystem::stream_type)i) == strategy) { |
| mpClientInterface->setStreamOutput((AudioSystem::stream_type)i, a2dpOutput); |
| } |
| } |
| } |
| } |
| |
| void AudioPolicyManagerBase::checkOutputForAllStrategies(uint32_t &newDevice) |
| { |
| // Check strategies in order of priority so that once newDevice is set |
| // for a given strategy it is not modified by subsequent calls to |
| // checkOutputForStrategy() |
| checkOutputForStrategy(STRATEGY_PHONE, newDevice); |
| checkOutputForStrategy(STRATEGY_SONIFICATION, newDevice); |
| checkOutputForStrategy(STRATEGY_MEDIA, newDevice); |
| checkOutputForStrategy(STRATEGY_DTMF, newDevice); |
| } |
| |
| #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: we are in call or the strategy phone is active on the hardware output: |
| // use device for strategy phone |
| // 2: the strategy sonification is active on the hardware output: |
| // use device for strategy sonification |
| // 3: the strategy media is active on the hardware output: |
| // use device for strategy media |
| // 4: the strategy DTMF is active on the hardware output: |
| // use device for strategy DTMF |
| if (mPhoneState == AudioSystem::MODE_IN_CALL || |
| 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; |
| } |
| |
| AudioPolicyManagerBase::routing_strategy AudioPolicyManagerBase::getStrategy(AudioSystem::stream_type stream) |
| { |
| // stream to strategy mapping |
| switch (stream) { |
| case AudioSystem::VOICE_CALL: |
| case AudioSystem::BLUETOOTH_SCO: |
| return STRATEGY_PHONE; |
| case AudioSystem::RING: |
| case AudioSystem::NOTIFICATION: |
| case AudioSystem::ALARM: |
| case AudioSystem::ENFORCED_AUDIBLE: |
| 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; |
| } |
| } |
| |
| 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 (mPhoneState != AudioSystem::MODE_IN_CALL) { |
| // 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 (mPhoneState != AudioSystem::MODE_IN_CALL || 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 (mPhoneState != AudioSystem::MODE_IN_CALL) { |
| 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_EARPIECE; |
| if (device == 0) { |
| LOGE("getDeviceForStrategy() earpiece device not found"); |
| } |
| break; |
| |
| case AudioSystem::FORCE_SPEAKER: |
| if (mPhoneState != AudioSystem::MODE_IN_CALL || strategy != STRATEGY_DTMF) { |
| device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT; |
| if (device) break; |
| } |
| #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 (mPhoneState != AudioSystem::MODE_IN_CALL) { |
| device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER; |
| if (device) break; |
| } |
| #endif |
| 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 (mPhoneState == AudioSystem::MODE_IN_CALL) { |
| device = getDeviceForStrategy(STRATEGY_PHONE, false); |
| break; |
| } |
| 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_AUX_DIGITAL; |
| if (device2 == 0) { |
| device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_WIRED_HEADPHONE; |
| } |
| if (device2 == 0) { |
| device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_WIRED_HEADSET; |
| } |
| #ifdef WITH_A2DP |
| if (mA2dpOutput != 0) { |
| if (strategy == STRATEGY_SONIFICATION && !a2dpUsedForSonification()) { |
| break; |
| } |
| 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_SPEAKER; |
| } |
| |
| // device is DEVICE_OUT_SPEAKER if we come from case STRATEGY_SONIFICATION, 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 |
| usleep(outputDesc->mLatency*2*1000); |
| } |
| #ifdef WITH_A2DP |
| // suspend A2DP output if SCO device is selected |
| if (AudioSystem::isBluetoothScoDevice((AudioSystem::audio_devices)device)) { |
| if (mA2dpOutput != 0) { |
| mpClientInterface->suspendOutput(mA2dpOutput); |
| } |
| } |
| #endif |
| // 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); |
| |
| #ifdef WITH_A2DP |
| // if disconnecting SCO device, restore A2DP output |
| if (AudioSystem::isBluetoothScoDevice((AudioSystem::audio_devices)prevDevice)) { |
| if (mA2dpOutput != 0) { |
| LOGV("restore A2DP output"); |
| mpClientInterface->restoreOutput(mA2dpOutput); |
| } |
| } |
| #endif |
| // 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: |
| if (mForceUse[AudioSystem::FOR_RECORD] == AudioSystem::FORCE_BT_SCO && |
| mAvailableInputDevices & AudioSystem::DEVICE_IN_BLUETOOTH_SCO_HEADSET) { |
| device = AudioSystem::DEVICE_IN_BLUETOOTH_SCO_HEADSET; |
| } else if (mAvailableInputDevices & AudioSystem::DEVICE_IN_WIRED_HEADSET) { |
| device = AudioSystem::DEVICE_IN_WIRED_HEADSET; |
| } else { |
| device = AudioSystem::DEVICE_IN_BUILTIN_MIC; |
| } |
| break; |
| case AUDIO_SOURCE_CAMCORDER: |
| if (hasBackMicrophone()) { |
| device = AudioSystem::DEVICE_IN_BACK_MIC; |
| } else { |
| device = AudioSystem::DEVICE_IN_BUILTIN_MIC; |
| } |
| break; |
| case AUDIO_SOURCE_VOICE_UPLINK: |
| case AUDIO_SOURCE_VOICE_DOWNLINK: |
| case AUDIO_SOURCE_VOICE_CALL: |
| device = AudioSystem::DEVICE_IN_VOICE_CALL; |
| break; |
| default: |
| LOGW("getInput() invalid input source %d", inputSource); |
| device = 0; |
| break; |
| } |
| LOGV("getDeviceForInputSource()input source %d, device %08x", inputSource, device); |
| return device; |
| } |
| |
| audio_io_handle_t AudioPolicyManagerBase::getActiveInput() |
| { |
| for (size_t i = 0; i < mInputs.size(); i++) { |
| if (mInputs.valueAt(i)->mRefCount > 0) { |
| return mInputs.keyAt(i); |
| } |
| } |
| return 0; |
| } |
| |
| 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(); |
| } |
| |
| int volInt = (100 * (index - streamDesc.mIndexMin)) / (streamDesc.mIndexMax - streamDesc.mIndexMin); |
| volume = AudioSystem::linearToLog(volInt); |
| |
| // 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) && |
| 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); |
| // do not set volume if the float value did not change |
| 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) { |
| float voiceVolume = -1.0; |
| // 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; |
| if (stream == AudioSystem::VOICE_CALL) { |
| voiceVolume = (float)index/(float)mStreams[stream].mIndexMax; |
| } else if (stream == AudioSystem::BLUETOOTH_SCO) { |
| voiceVolume = 1.0; |
| } |
| if (voiceVolume >= 0 && output == mHardwareOutput) { |
| mpClientInterface->setVoiceVolume(voiceVolume, delayMs); |
| } |
| } |
| mpClientInterface->setStreamVolume((AudioSystem::stream_type)stream, volume, output, delayMs); |
| } |
| |
| return NO_ERROR; |
| } |
| |
| void AudioPolicyManagerBase::applyStreamVolumes(audio_io_handle_t output, uint32_t device, int delayMs) |
| { |
| 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); |
| } |
| } |
| |
| 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::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))); |
| } |
| |
| // --- 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; |
| } |
| } |
| |
| 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) |
| { |
| } |
| |
| 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); |
| } |
| |
| |
| }; // namespace android |