| /* |
| * 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 |
| |
| //#define VERY_VERBOSE_LOGGING |
| #ifdef VERY_VERBOSE_LOGGING |
| #define ALOGVV ALOGV |
| #else |
| #define ALOGVV(a...) do { } while(0) |
| #endif |
| |
| // A device mask for all audio input devices that are considered "virtual" when evaluating |
| // active inputs in getActiveInput() |
| #define APM_AUDIO_IN_DEVICE_VIRTUAL_ALL AUDIO_DEVICE_IN_REMOTE_SUBMIX |
| // A device mask for all audio output devices that are considered "remote" when evaluating |
| // active output devices in isStreamActiveRemotely() |
| #define APM_AUDIO_OUT_DEVICE_REMOTE_ALL AUDIO_DEVICE_OUT_REMOTE_SUBMIX |
| |
| #include <inttypes.h> |
| #include <math.h> |
| |
| #include <cutils/properties.h> |
| #include <utils/Log.h> |
| #include <utils/Timers.h> |
| |
| #include <hardware/audio.h> |
| #include <hardware/audio_effect.h> |
| #include <hardware_legacy/audio_policy_conf.h> |
| #include <hardware_legacy/AudioPolicyManagerBase.h> |
| |
| namespace android_audio_legacy { |
| |
| // ---------------------------------------------------------------------------- |
| // AudioPolicyInterface implementation |
| // ---------------------------------------------------------------------------- |
| |
| |
| status_t AudioPolicyManagerBase::setDeviceConnectionState(audio_devices_t device, |
| AudioSystem::device_connection_state state, |
| const char *device_address) |
| { |
| // device_address can be NULL and should be handled as an empty string in this case, |
| // and it is not checked by AudioPolicyInterfaceImpl.cpp |
| if (device_address == NULL) { |
| device_address = ""; |
| } |
| ALOGV("setDeviceConnectionState() device: 0x%X, state %d, address %s", device, state, device_address); |
| |
| // connect/disconnect only 1 device at a time |
| if (!audio_is_output_device(device) && !audio_is_input_device(device)) return BAD_VALUE; |
| |
| if (strlen(device_address) >= MAX_DEVICE_ADDRESS_LEN) { |
| ALOGE("setDeviceConnectionState() invalid address: %s", device_address); |
| return BAD_VALUE; |
| } |
| |
| // handle output devices |
| if (audio_is_output_device(device)) { |
| SortedVector <audio_io_handle_t> outputs; |
| |
| if (!mHasA2dp && audio_is_a2dp_out_device(device)) { |
| ALOGE("setDeviceConnectionState() invalid A2DP device: %x", device); |
| return BAD_VALUE; |
| } |
| if (!mHasUsb && audio_is_usb_out_device(device)) { |
| ALOGE("setDeviceConnectionState() invalid USB audio device: %x", device); |
| return BAD_VALUE; |
| } |
| if (!mHasRemoteSubmix && audio_is_remote_submix_device((audio_devices_t)device)) { |
| ALOGE("setDeviceConnectionState() invalid remote submix audio device: %x", device); |
| return BAD_VALUE; |
| } |
| |
| // save a copy of the opened output descriptors before any output is opened or closed |
| // by checkOutputsForDevice(). This will be needed by checkOutputForAllStrategies() |
| mPreviousOutputs = mOutputs; |
| String8 paramStr; |
| switch (state) |
| { |
| // handle output device connection |
| case AudioSystem::DEVICE_STATE_AVAILABLE: |
| if (mAvailableOutputDevices & device) { |
| ALOGW("setDeviceConnectionState() device already connected: %x", device); |
| return INVALID_OPERATION; |
| } |
| ALOGV("setDeviceConnectionState() connecting device %x", device); |
| |
| if (mHasA2dp && audio_is_a2dp_out_device(device)) { |
| // handle A2DP device connection |
| AudioParameter param; |
| param.add(String8(AUDIO_PARAMETER_A2DP_SINK_ADDRESS), String8(device_address)); |
| paramStr = param.toString(); |
| } else if (mHasUsb && audio_is_usb_out_device(device)) { |
| // handle USB device connection |
| paramStr = String8(device_address, MAX_DEVICE_ADDRESS_LEN); |
| } |
| |
| if (checkOutputsForDevice(device, state, outputs, paramStr) != NO_ERROR) { |
| return INVALID_OPERATION; |
| } |
| ALOGV("setDeviceConnectionState() checkOutputsForDevice() returned %zu outputs", |
| outputs.size()); |
| // register new device as available |
| mAvailableOutputDevices = (audio_devices_t)(mAvailableOutputDevices | device); |
| |
| if (mHasA2dp && audio_is_a2dp_out_device(device)) { |
| // handle A2DP device connection |
| mA2dpDeviceAddress = String8(device_address, MAX_DEVICE_ADDRESS_LEN); |
| mA2dpSuspended = false; |
| } else if (audio_is_bluetooth_sco_device(device)) { |
| // handle SCO device connection |
| mScoDeviceAddress = String8(device_address, MAX_DEVICE_ADDRESS_LEN); |
| } else if (mHasUsb && audio_is_usb_out_device(device)) { |
| // handle USB device connection |
| mUsbOutCardAndDevice = String8(device_address, MAX_DEVICE_ADDRESS_LEN); |
| } |
| |
| break; |
| // handle output device disconnection |
| case AudioSystem::DEVICE_STATE_UNAVAILABLE: { |
| if (!(mAvailableOutputDevices & device)) { |
| ALOGW("setDeviceConnectionState() device not connected: %x", device); |
| return INVALID_OPERATION; |
| } |
| |
| ALOGV("setDeviceConnectionState() disconnecting device %x", device); |
| // remove device from available output devices |
| mAvailableOutputDevices = (audio_devices_t)(mAvailableOutputDevices & ~device); |
| checkOutputsForDevice(device, state, outputs, paramStr); |
| |
| if (mHasA2dp && audio_is_a2dp_out_device(device)) { |
| // handle A2DP device disconnection |
| mA2dpDeviceAddress = ""; |
| mA2dpSuspended = false; |
| } else if (audio_is_bluetooth_sco_device(device)) { |
| // handle SCO device disconnection |
| mScoDeviceAddress = ""; |
| } else if (mHasUsb && audio_is_usb_out_device(device)) { |
| // handle USB device disconnection |
| mUsbOutCardAndDevice = ""; |
| } |
| // not currently handling multiple simultaneous submixes: ignoring remote submix |
| // case and address |
| } break; |
| |
| default: |
| ALOGE("setDeviceConnectionState() invalid state: %x", state); |
| return BAD_VALUE; |
| } |
| |
| checkA2dpSuspend(); |
| checkOutputForAllStrategies(); |
| // outputs must be closed after checkOutputForAllStrategies() is executed |
| if (!outputs.isEmpty()) { |
| for (size_t i = 0; i < outputs.size(); i++) { |
| AudioOutputDescriptor *desc = mOutputs.valueFor(outputs[i]); |
| // close unused outputs after device disconnection or direct outputs that have been |
| // opened by checkOutputsForDevice() to query dynamic parameters |
| if ((state == AudioSystem::DEVICE_STATE_UNAVAILABLE) || |
| (((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) && |
| (desc->mDirectOpenCount == 0))) { |
| closeOutput(outputs[i]); |
| } |
| } |
| } |
| |
| updateDevicesAndOutputs(); |
| for (size_t i = 0; i < mOutputs.size(); i++) { |
| // do not force device change on duplicated output because if device is 0, it will |
| // also force a device 0 for the two outputs it is duplicated to which may override |
| // a valid device selection on those outputs. |
| setOutputDevice(mOutputs.keyAt(i), |
| getNewDevice(mOutputs.keyAt(i), true /*fromCache*/), |
| !mOutputs.valueAt(i)->isDuplicated(), |
| 0); |
| } |
| |
| return NO_ERROR; |
| } // end if is output device |
| |
| // handle input devices |
| if (audio_is_input_device(device)) { |
| SortedVector <audio_io_handle_t> inputs; |
| |
| String8 paramStr; |
| switch (state) |
| { |
| // handle input device connection |
| case AudioSystem::DEVICE_STATE_AVAILABLE: { |
| if (mAvailableInputDevices & device) { |
| ALOGW("setDeviceConnectionState() device already connected: %d", device); |
| return INVALID_OPERATION; |
| } |
| |
| if (mHasUsb && audio_is_usb_in_device(device)) { |
| // handle USB device connection |
| paramStr = String8(device_address, MAX_DEVICE_ADDRESS_LEN); |
| } else if (mHasA2dp && audio_is_a2dp_in_device(device)) { |
| // handle A2DP device connection |
| AudioParameter param; |
| param.add(String8(AUDIO_PARAMETER_A2DP_SOURCE_ADDRESS), String8(device_address)); |
| paramStr = param.toString(); |
| } |
| |
| if (checkInputsForDevice(device, state, inputs, paramStr) != NO_ERROR) { |
| return INVALID_OPERATION; |
| } |
| mAvailableInputDevices = mAvailableInputDevices | (device & ~AUDIO_DEVICE_BIT_IN); |
| } |
| break; |
| |
| // handle input device disconnection |
| case AudioSystem::DEVICE_STATE_UNAVAILABLE: { |
| if (!(mAvailableInputDevices & device)) { |
| ALOGW("setDeviceConnectionState() device not connected: %d", device); |
| return INVALID_OPERATION; |
| } |
| checkInputsForDevice(device, state, inputs, paramStr); |
| mAvailableInputDevices = (audio_devices_t) (mAvailableInputDevices & ~device); |
| } break; |
| |
| default: |
| ALOGE("setDeviceConnectionState() invalid state: %x", state); |
| return BAD_VALUE; |
| } |
| |
| closeAllInputs(); |
| |
| return NO_ERROR; |
| } // end if is input device |
| |
| ALOGW("setDeviceConnectionState() invalid device: %x", device); |
| return BAD_VALUE; |
| } |
| |
| AudioSystem::device_connection_state AudioPolicyManagerBase::getDeviceConnectionState(audio_devices_t device, |
| const char *device_address) |
| { |
| // similar to setDeviceConnectionState |
| if (device_address == NULL) { |
| device_address = ""; |
| } |
| AudioSystem::device_connection_state state = AudioSystem::DEVICE_STATE_UNAVAILABLE; |
| String8 address = String8(device_address); |
| if (audio_is_output_device(device)) { |
| if (device & mAvailableOutputDevices) { |
| if (audio_is_a2dp_out_device(device) && |
| (!mHasA2dp || (address != "" && mA2dpDeviceAddress != address))) { |
| return state; |
| } |
| if (audio_is_bluetooth_sco_device(device) && |
| address != "" && mScoDeviceAddress != address) { |
| return state; |
| } |
| if (audio_is_usb_out_device(device) && |
| (!mHasUsb || (address != "" && mUsbOutCardAndDevice != address))) { |
| ALOGE("getDeviceConnectionState() invalid device: %x", device); |
| return state; |
| } |
| if (audio_is_remote_submix_device((audio_devices_t)device) && !mHasRemoteSubmix) { |
| return state; |
| } |
| state = AudioSystem::DEVICE_STATE_AVAILABLE; |
| } |
| } else if (audio_is_input_device(device)) { |
| if (device & mAvailableInputDevices) { |
| state = AudioSystem::DEVICE_STATE_AVAILABLE; |
| } |
| } |
| |
| return state; |
| } |
| |
| void AudioPolicyManagerBase::setPhoneState(int state) |
| { |
| ALOGV("setPhoneState() state %d", state); |
| audio_devices_t newDevice = AUDIO_DEVICE_NONE; |
| if (state < 0 || state >= AudioSystem::NUM_MODES) { |
| ALOGW("setPhoneState() invalid state %d", state); |
| return; |
| } |
| |
| if (state == mPhoneState ) { |
| ALOGW("setPhoneState() setting same state %d", state); |
| return; |
| } |
| |
| // if leaving call state, handle special case of active streams |
| // pertaining to sonification strategy see handleIncallSonification() |
| if (isInCall()) { |
| ALOGV("setPhoneState() in call state management: new state is %d", state); |
| for (int stream = 0; stream < 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)) { |
| ALOGV(" Entering call in setPhoneState()"); |
| // force routing command to audio hardware when starting a call |
| // even if no device change is needed |
| force = true; |
| for (int j = 0; j < DEVICE_CATEGORY_CNT; j++) { |
| mStreams[AUDIO_STREAM_DTMF].mVolumeCurve[j] = |
| sVolumeProfiles[AUDIO_STREAM_VOICE_CALL][j]; |
| } |
| } else if (isStateInCall(oldState) && !isStateInCall(state)) { |
| ALOGV(" Exiting call in setPhoneState()"); |
| // force routing command to audio hardware when exiting a call |
| // even if no device change is needed |
| force = true; |
| for (int j = 0; j < DEVICE_CATEGORY_CNT; j++) { |
| mStreams[AUDIO_STREAM_DTMF].mVolumeCurve[j] = |
| sVolumeProfiles[AUDIO_STREAM_DTMF][j]; |
| } |
| } else if (isStateInCall(state) && (state != oldState)) { |
| ALOGV(" Switching between telephony and VoIP in setPhoneState()"); |
| // force routing command to audio hardware when switching between telephony and VoIP |
| // even if no device change is needed |
| force = true; |
| } |
| |
| // check for device and output changes triggered by new phone state |
| newDevice = getNewDevice(mPrimaryOutput, false /*fromCache*/); |
| checkA2dpSuspend(); |
| checkOutputForAllStrategies(); |
| updateDevicesAndOutputs(); |
| |
| AudioOutputDescriptor *hwOutputDesc = mOutputs.valueFor(mPrimaryOutput); |
| |
| // force routing command to audio hardware when ending call |
| // even if no device change is needed |
| if (isStateInCall(oldState) && newDevice == AUDIO_DEVICE_NONE) { |
| newDevice = hwOutputDesc->device(); |
| } |
| |
| int delayMs = 0; |
| if (isStateInCall(state)) { |
| nsecs_t sysTime = systemTime(); |
| for (size_t i = 0; i < mOutputs.size(); i++) { |
| AudioOutputDescriptor *desc = mOutputs.valueAt(i); |
| // mute media and sonification strategies and delay device switch by the largest |
| // latency of any output where either strategy is active. |
| // This avoid sending the ring tone or music tail into the earpiece or headset. |
| if ((desc->isStrategyActive(STRATEGY_MEDIA, |
| SONIFICATION_HEADSET_MUSIC_DELAY, |
| sysTime) || |
| desc->isStrategyActive(STRATEGY_SONIFICATION, |
| SONIFICATION_HEADSET_MUSIC_DELAY, |
| sysTime)) && |
| (delayMs < (int)desc->mLatency*2)) { |
| delayMs = desc->mLatency*2; |
| } |
| setStrategyMute(STRATEGY_MEDIA, true, mOutputs.keyAt(i)); |
| setStrategyMute(STRATEGY_MEDIA, false, mOutputs.keyAt(i), MUTE_TIME_MS, |
| getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/)); |
| setStrategyMute(STRATEGY_SONIFICATION, true, mOutputs.keyAt(i)); |
| setStrategyMute(STRATEGY_SONIFICATION, false, mOutputs.keyAt(i), MUTE_TIME_MS, |
| getDeviceForStrategy(STRATEGY_SONIFICATION, true /*fromCache*/)); |
| } |
| } |
| |
| // change routing is necessary |
| setOutputDevice(mPrimaryOutput, newDevice, force, delayMs); |
| |
| // if entering in call state, handle special case of active streams |
| // pertaining to sonification strategy see handleIncallSonification() |
| if (isStateInCall(state)) { |
| ALOGV("setPhoneState() in call state management: new state is %d", state); |
| for (int stream = 0; stream < 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::setForceUse(AudioSystem::force_use usage, AudioSystem::forced_config config) |
| { |
| ALOGV("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) { |
| ALOGW("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 && |
| config != AudioSystem::FORCE_NO_BT_A2DP) { |
| ALOGW("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) { |
| ALOGW("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) { |
| ALOGW("setForceUse() invalid config %d for FOR_DOCK", config); |
| } |
| forceVolumeReeval = true; |
| mForceUse[usage] = config; |
| break; |
| case AudioSystem::FOR_SYSTEM: |
| if (config != AudioSystem::FORCE_NONE && |
| config != AudioSystem::FORCE_SYSTEM_ENFORCED) { |
| ALOGW("setForceUse() invalid config %d for FOR_SYSTEM", config); |
| } |
| forceVolumeReeval = true; |
| mForceUse[usage] = config; |
| break; |
| default: |
| ALOGW("setForceUse() invalid usage %d", usage); |
| break; |
| } |
| |
| // check for device and output changes triggered by new force usage |
| checkA2dpSuspend(); |
| checkOutputForAllStrategies(); |
| updateDevicesAndOutputs(); |
| for (size_t i = 0; i < mOutputs.size(); i++) { |
| audio_io_handle_t output = mOutputs.keyAt(i); |
| audio_devices_t newDevice = getNewDevice(output, true /*fromCache*/); |
| setOutputDevice(output, newDevice, (newDevice != AUDIO_DEVICE_NONE)); |
| if (forceVolumeReeval && (newDevice != AUDIO_DEVICE_NONE)) { |
| applyStreamVolumes(output, newDevice, 0, true); |
| } |
| } |
| |
| audio_io_handle_t activeInput = getActiveInput(); |
| if (activeInput != 0) { |
| AudioInputDescriptor *inputDesc = mInputs.valueFor(activeInput); |
| audio_devices_t newDevice = getDeviceForInputSource(inputDesc->mInputSource); |
| if ((newDevice != AUDIO_DEVICE_NONE) && (newDevice != inputDesc->mDevice)) { |
| ALOGV("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) |
| { |
| ALOGV("setSystemProperty() property %s, value %s", property, value); |
| } |
| |
| // Find a direct output profile compatible with the parameters passed, even if the input flags do |
| // not explicitly request a direct output |
| AudioPolicyManagerBase::IOProfile *AudioPolicyManagerBase::getProfileForDirectOutput( |
| audio_devices_t device, |
| uint32_t samplingRate, |
| audio_format_t format, |
| audio_channel_mask_t channelMask, |
| audio_output_flags_t flags) |
| { |
| for (size_t i = 0; i < mHwModules.size(); i++) { |
| if (mHwModules[i]->mHandle == 0) { |
| continue; |
| } |
| for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++) { |
| IOProfile *profile = mHwModules[i]->mOutputProfiles[j]; |
| if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { |
| if (profile->isCompatibleProfile(device, samplingRate, format, |
| channelMask, |
| AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)) { |
| if (mAvailableOutputDevices & profile->mSupportedDevices) { |
| return mHwModules[i]->mOutputProfiles[j]; |
| } |
| } |
| } else { |
| if (profile->isCompatibleProfile(device, samplingRate, format, |
| channelMask, |
| AUDIO_OUTPUT_FLAG_DIRECT)) { |
| if (mAvailableOutputDevices & profile->mSupportedDevices) { |
| return mHwModules[i]->mOutputProfiles[j]; |
| } |
| } |
| } |
| } |
| } |
| return 0; |
| } |
| |
| audio_io_handle_t AudioPolicyManagerBase::getOutput(AudioSystem::stream_type stream, |
| uint32_t samplingRate, |
| audio_format_t format, |
| audio_channel_mask_t channelMask, |
| AudioSystem::output_flags flags, |
| const audio_offload_info_t *offloadInfo) |
| { |
| audio_io_handle_t output = 0; |
| uint32_t latency = 0; |
| routing_strategy strategy = getStrategy((AudioSystem::stream_type)stream); |
| audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/); |
| ALOGV("getOutput() device %d, stream %d, samplingRate %d, format %x, channelMask %x, flags %x", |
| device, stream, samplingRate, format, channelMask, flags); |
| |
| #ifdef AUDIO_POLICY_TEST |
| if (mCurOutput != 0) { |
| ALOGV("getOutput() test output mCurOutput %d, samplingRate %d, format %d, channelMask %x, mDirectOutput %d", |
| mCurOutput, mTestSamplingRate, mTestFormat, mTestChannels, mDirectOutput); |
| |
| if (mTestOutputs[mCurOutput] == 0) { |
| ALOGV("getOutput() opening test output"); |
| AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(NULL); |
| outputDesc->mDevice = mTestDevice; |
| outputDesc->mSamplingRate = mTestSamplingRate; |
| outputDesc->mFormat = mTestFormat; |
| outputDesc->mChannelMask = mTestChannels; |
| outputDesc->mLatency = mTestLatencyMs; |
| outputDesc->mFlags = (audio_output_flags_t)(mDirectOutput ? AudioSystem::OUTPUT_FLAG_DIRECT : 0); |
| outputDesc->mRefCount[stream] = 0; |
| mTestOutputs[mCurOutput] = mpClientInterface->openOutput(0, &outputDesc->mDevice, |
| &outputDesc->mSamplingRate, |
| &outputDesc->mFormat, |
| &outputDesc->mChannelMask, |
| &outputDesc->mLatency, |
| outputDesc->mFlags, |
| offloadInfo); |
| 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 |
| //force direct flag if offload flag is set: offloading implies a direct output stream |
| // and all common behaviors are driven by checking only the direct flag |
| // this should normally be set appropriately in the policy configuration file |
| if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) { |
| flags = (AudioSystem::output_flags)(flags | AUDIO_OUTPUT_FLAG_DIRECT); |
| } |
| |
| // Do not allow offloading if one non offloadable effect is enabled. This prevents from |
| // creating an offloaded track and tearing it down immediately after start when audioflinger |
| // detects there is an active non offloadable effect. |
| // FIXME: We should check the audio session here but we do not have it in this context. |
| // This may prevent offloading in rare situations where effects are left active by apps |
| // in the background. |
| IOProfile *profile = NULL; |
| if (((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) == 0) || |
| !isNonOffloadableEffectEnabled()) { |
| profile = getProfileForDirectOutput(device, |
| samplingRate, |
| format, |
| channelMask, |
| (audio_output_flags_t)flags); |
| } |
| |
| if (profile != NULL) { |
| AudioOutputDescriptor *outputDesc = NULL; |
| |
| for (size_t i = 0; i < mOutputs.size(); i++) { |
| AudioOutputDescriptor *desc = mOutputs.valueAt(i); |
| if (!desc->isDuplicated() && (profile == desc->mProfile)) { |
| outputDesc = desc; |
| // reuse direct output if currently open and configured with same parameters |
| if ((samplingRate == outputDesc->mSamplingRate) && |
| (format == outputDesc->mFormat) && |
| (channelMask == outputDesc->mChannelMask)) { |
| outputDesc->mDirectOpenCount++; |
| ALOGV("getOutput() reusing direct output %d", mOutputs.keyAt(i)); |
| return mOutputs.keyAt(i); |
| } |
| } |
| } |
| // close direct output if currently open and configured with different parameters |
| if (outputDesc != NULL) { |
| closeOutput(outputDesc->mId); |
| } |
| outputDesc = new AudioOutputDescriptor(profile); |
| outputDesc->mDevice = device; |
| outputDesc->mSamplingRate = samplingRate; |
| outputDesc->mFormat = format; |
| outputDesc->mChannelMask = channelMask; |
| outputDesc->mLatency = 0; |
| outputDesc->mFlags =(audio_output_flags_t) (outputDesc->mFlags | flags); |
| outputDesc->mRefCount[stream] = 0; |
| outputDesc->mStopTime[stream] = 0; |
| outputDesc->mDirectOpenCount = 1; |
| output = mpClientInterface->openOutput(profile->mModule->mHandle, |
| &outputDesc->mDevice, |
| &outputDesc->mSamplingRate, |
| &outputDesc->mFormat, |
| &outputDesc->mChannelMask, |
| &outputDesc->mLatency, |
| outputDesc->mFlags, |
| offloadInfo); |
| |
| // only accept an output with the requested parameters |
| if (output == 0 || |
| (samplingRate != 0 && samplingRate != outputDesc->mSamplingRate) || |
| (format != AUDIO_FORMAT_DEFAULT && format != outputDesc->mFormat) || |
| (channelMask != 0 && channelMask != outputDesc->mChannelMask)) { |
| ALOGV("getOutput() failed opening direct output: output %d samplingRate %d %d," |
| "format %d %d, channelMask %04x %04x", output, samplingRate, |
| outputDesc->mSamplingRate, format, outputDesc->mFormat, channelMask, |
| outputDesc->mChannelMask); |
| if (output != 0) { |
| mpClientInterface->closeOutput(output); |
| } |
| delete outputDesc; |
| return 0; |
| } |
| audio_io_handle_t srcOutput = getOutputForEffect(); |
| addOutput(output, outputDesc); |
| audio_io_handle_t dstOutput = getOutputForEffect(); |
| if (dstOutput == output) { |
| mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, srcOutput, dstOutput); |
| } |
| mPreviousOutputs = mOutputs; |
| ALOGV("getOutput() returns new direct output %d", output); |
| return output; |
| } |
| |
| // ignoring channel mask due to downmix capability in mixer |
| |
| // open a non direct output |
| |
| // for non direct outputs, only PCM is supported |
| if (audio_is_linear_pcm(format)) { |
| // get which output is suitable for the specified stream. The actual |
| // routing change will happen when startOutput() will be called |
| SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(device, mOutputs); |
| |
| output = selectOutput(outputs, flags); |
| } |
| ALOGW_IF((output == 0), "getOutput() could not find output for stream %d, samplingRate %d," |
| "format %d, channels %x, flags %x", stream, samplingRate, format, channelMask, flags); |
| |
| ALOGV("getOutput() returns output %d", output); |
| |
| return output; |
| } |
| |
| audio_io_handle_t AudioPolicyManagerBase::selectOutput(const SortedVector<audio_io_handle_t>& outputs, |
| AudioSystem::output_flags flags) |
| { |
| // select one output among several that provide a path to a particular device or set of |
| // devices (the list was previously build by getOutputsForDevice()). |
| // The priority is as follows: |
| // 1: the output with the highest number of requested policy flags |
| // 2: the primary output |
| // 3: the first output in the list |
| |
| if (outputs.size() == 0) { |
| return 0; |
| } |
| if (outputs.size() == 1) { |
| return outputs[0]; |
| } |
| |
| int maxCommonFlags = 0; |
| audio_io_handle_t outputFlags = 0; |
| audio_io_handle_t outputPrimary = 0; |
| |
| for (size_t i = 0; i < outputs.size(); i++) { |
| AudioOutputDescriptor *outputDesc = mOutputs.valueFor(outputs[i]); |
| if (!outputDesc->isDuplicated()) { |
| int commonFlags = (int)AudioSystem::popCount(outputDesc->mProfile->mFlags & flags); |
| if (commonFlags > maxCommonFlags) { |
| outputFlags = outputs[i]; |
| maxCommonFlags = commonFlags; |
| ALOGV("selectOutput() commonFlags for output %d, %04x", outputs[i], commonFlags); |
| } |
| if (outputDesc->mProfile->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY) { |
| outputPrimary = outputs[i]; |
| } |
| } |
| } |
| |
| if (outputFlags != 0) { |
| return outputFlags; |
| } |
| if (outputPrimary != 0) { |
| return outputPrimary; |
| } |
| |
| return outputs[0]; |
| } |
| |
| status_t AudioPolicyManagerBase::startOutput(audio_io_handle_t output, |
| AudioSystem::stream_type stream, |
| int session) |
| { |
| ALOGV("startOutput() output %d, stream %d, session %d", output, stream, session); |
| ssize_t index = mOutputs.indexOfKey(output); |
| if (index < 0) { |
| ALOGW("startOutput() unknown output %d", output); |
| return BAD_VALUE; |
| } |
| |
| AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index); |
| |
| // increment usage count for this stream on the requested output: |
| // NOTE that the usage count is the same for duplicated output and hardware output which is |
| // necessary for a correct control of hardware output routing by startOutput() and stopOutput() |
| outputDesc->changeRefCount(stream, 1); |
| |
| if (outputDesc->mRefCount[stream] == 1) { |
| audio_devices_t newDevice = getNewDevice(output, false /*fromCache*/); |
| routing_strategy strategy = getStrategy(stream); |
| bool shouldWait = (strategy == STRATEGY_SONIFICATION) || |
| (strategy == STRATEGY_SONIFICATION_RESPECTFUL); |
| uint32_t waitMs = 0; |
| bool force = false; |
| for (size_t i = 0; i < mOutputs.size(); i++) { |
| AudioOutputDescriptor *desc = mOutputs.valueAt(i); |
| if (desc != outputDesc) { |
| // force a device change if any other output is managed by the same hw |
| // module and has a current device selection that differs from selected device. |
| // In this case, the audio HAL must receive the new device selection so that it can |
| // change the device currently selected by the other active output. |
| if (outputDesc->sharesHwModuleWith(desc) && |
| desc->device() != newDevice) { |
| force = true; |
| } |
| // wait for audio on other active outputs to be presented when starting |
| // a notification so that audio focus effect can propagate. |
| uint32_t latency = desc->latency(); |
| if (shouldWait && desc->isActive(latency * 2) && (waitMs < latency)) { |
| waitMs = latency; |
| } |
| } |
| } |
| uint32_t muteWaitMs = setOutputDevice(output, newDevice, force); |
| |
| // handle special case for sonification while in call |
| if (isInCall()) { |
| handleIncallSonification(stream, true, false); |
| } |
| |
| // apply volume rules for current stream and device if necessary |
| checkAndSetVolume(stream, |
| mStreams[stream].getVolumeIndex(newDevice), |
| output, |
| newDevice); |
| |
| // update the outputs if starting an output with a stream that can affect notification |
| // routing |
| handleNotificationRoutingForStream(stream); |
| if (waitMs > muteWaitMs) { |
| usleep((waitMs - muteWaitMs) * 2 * 1000); |
| } |
| } |
| return NO_ERROR; |
| } |
| |
| |
| status_t AudioPolicyManagerBase::stopOutput(audio_io_handle_t output, |
| AudioSystem::stream_type stream, |
| int session) |
| { |
| ALOGV("stopOutput() output %d, stream %d, session %d", output, stream, session); |
| ssize_t index = mOutputs.indexOfKey(output); |
| if (index < 0) { |
| ALOGW("stopOutput() unknown output %d", output); |
| return BAD_VALUE; |
| } |
| |
| AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index); |
| |
| // handle special case for sonification while in call |
| if (isInCall()) { |
| handleIncallSonification(stream, false, false); |
| } |
| |
| if (outputDesc->mRefCount[stream] > 0) { |
| // decrement usage count of this stream on the output |
| outputDesc->changeRefCount(stream, -1); |
| // store time at which the stream was stopped - see isStreamActive() |
| if (outputDesc->mRefCount[stream] == 0) { |
| outputDesc->mStopTime[stream] = systemTime(); |
| audio_devices_t newDevice = getNewDevice(output, false /*fromCache*/); |
| // delay the device switch by twice the latency because stopOutput() is executed when |
| // the track stop() command is received and at that time the audio track buffer can |
| // still contain data that needs to be drained. The latency only covers the audio HAL |
| // and kernel buffers. Also the latency does not always include additional delay in the |
| // audio path (audio DSP, CODEC ...) |
| setOutputDevice(output, newDevice, false, outputDesc->mLatency*2); |
| |
| // force restoring the device selection on other active outputs if it differs from the |
| // one being selected for this output |
| for (size_t i = 0; i < mOutputs.size(); i++) { |
| audio_io_handle_t curOutput = mOutputs.keyAt(i); |
| AudioOutputDescriptor *desc = mOutputs.valueAt(i); |
| if (curOutput != output && |
| desc->isActive() && |
| outputDesc->sharesHwModuleWith(desc) && |
| (newDevice != desc->device())) { |
| setOutputDevice(curOutput, |
| getNewDevice(curOutput, false /*fromCache*/), |
| true, |
| outputDesc->mLatency*2); |
| } |
| } |
| // update the outputs if stopping one with a stream that can affect notification routing |
| handleNotificationRoutingForStream(stream); |
| } |
| return NO_ERROR; |
| } else { |
| ALOGW("stopOutput() refcount is already 0 for output %d", output); |
| return INVALID_OPERATION; |
| } |
| } |
| |
| void AudioPolicyManagerBase::releaseOutput(audio_io_handle_t output) |
| { |
| ALOGV("releaseOutput() %d", output); |
| ssize_t index = mOutputs.indexOfKey(output); |
| if (index < 0) { |
| ALOGW("releaseOutput() releasing unknown output %d", output); |
| return; |
| } |
| |
| #ifdef AUDIO_POLICY_TEST |
| int testIndex = testOutputIndex(output); |
| if (testIndex != 0) { |
| AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index); |
| if (outputDesc->isActive()) { |
| mpClientInterface->closeOutput(output); |
| delete mOutputs.valueAt(index); |
| mOutputs.removeItem(output); |
| mTestOutputs[testIndex] = 0; |
| } |
| return; |
| } |
| #endif //AUDIO_POLICY_TEST |
| |
| AudioOutputDescriptor *desc = mOutputs.valueAt(index); |
| if (desc->mFlags & AudioSystem::OUTPUT_FLAG_DIRECT) { |
| if (desc->mDirectOpenCount <= 0) { |
| ALOGW("releaseOutput() invalid open count %d for output %d", |
| desc->mDirectOpenCount, output); |
| return; |
| } |
| if (--desc->mDirectOpenCount == 0) { |
| closeOutput(output); |
| // If effects where present on the output, audioflinger moved them to the primary |
| // output by default: move them back to the appropriate output. |
| audio_io_handle_t dstOutput = getOutputForEffect(); |
| if (dstOutput != mPrimaryOutput) { |
| mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, mPrimaryOutput, dstOutput); |
| } |
| } |
| } |
| } |
| |
| |
| audio_io_handle_t AudioPolicyManagerBase::getInput(int inputSource, |
| uint32_t samplingRate, |
| audio_format_t format, |
| audio_channel_mask_t channelMask, |
| AudioSystem::audio_in_acoustics acoustics) |
| { |
| audio_io_handle_t input = 0; |
| audio_devices_t device = getDeviceForInputSource(inputSource); |
| |
| ALOGV("getInput() inputSource %d, samplingRate %d, format %d, channelMask %x, acoustics %x", |
| inputSource, samplingRate, format, channelMask, acoustics); |
| |
| if (device == AUDIO_DEVICE_NONE) { |
| ALOGW("getInput() could not find device for inputSource %d", inputSource); |
| return 0; |
| } |
| |
| // adapt channel selection to input source |
| switch(inputSource) { |
| case AUDIO_SOURCE_VOICE_UPLINK: |
| channelMask = AUDIO_CHANNEL_IN_VOICE_UPLINK; |
| break; |
| case AUDIO_SOURCE_VOICE_DOWNLINK: |
| channelMask = AUDIO_CHANNEL_IN_VOICE_DNLINK; |
| break; |
| case AUDIO_SOURCE_VOICE_CALL: |
| channelMask = AUDIO_CHANNEL_IN_VOICE_UPLINK | AUDIO_CHANNEL_IN_VOICE_DNLINK; |
| break; |
| default: |
| break; |
| } |
| |
| IOProfile *profile = getInputProfile(device, |
| samplingRate, |
| format, |
| channelMask); |
| if (profile == NULL) { |
| ALOGW("getInput() could not find profile for device 0x%X, samplingRate %d, format %d, " |
| "channelMask 0x%X", |
| device, samplingRate, format, channelMask); |
| return 0; |
| } |
| |
| if (profile->mModule->mHandle == 0) { |
| ALOGE("getInput(): HW module %s not opened", profile->mModule->mName); |
| return 0; |
| } |
| |
| AudioInputDescriptor *inputDesc = new AudioInputDescriptor(profile); |
| |
| inputDesc->mInputSource = inputSource; |
| inputDesc->mDevice = device; |
| inputDesc->mSamplingRate = samplingRate; |
| inputDesc->mFormat = format; |
| inputDesc->mChannelMask = channelMask; |
| inputDesc->mRefCount = 0; |
| |
| input = mpClientInterface->openInput(profile->mModule->mHandle, |
| &inputDesc->mDevice, |
| &inputDesc->mSamplingRate, |
| &inputDesc->mFormat, |
| &inputDesc->mChannelMask); |
| |
| // only accept input with the exact requested set of parameters |
| if (input == 0 || |
| (samplingRate != inputDesc->mSamplingRate) || |
| (format != inputDesc->mFormat) || |
| (channelMask != inputDesc->mChannelMask)) { |
| ALOGI("getInput() failed opening input: samplingRate %d, format %d, channelMask 0x%X", |
| samplingRate, format, channelMask); |
| if (input != 0) { |
| mpClientInterface->closeInput(input); |
| } |
| delete inputDesc; |
| return 0; |
| } |
| addInput(input, inputDesc); |
| |
| return input; |
| } |
| |
| status_t AudioPolicyManagerBase::startInput(audio_io_handle_t input) |
| { |
| ALOGV("startInput() input %d", input); |
| ssize_t index = mInputs.indexOfKey(input); |
| if (index < 0) { |
| ALOGW("startInput() unknown 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 except if the active input |
| // uses AUDIO_SOURCE_HOTWORD in which case it is closed. |
| audio_io_handle_t activeInput = getActiveInput(); |
| if (!isVirtualInputDevice(inputDesc->mDevice) && activeInput != 0) { |
| AudioInputDescriptor *activeDesc = mInputs.valueFor(activeInput); |
| if (activeDesc->mInputSource == AUDIO_SOURCE_HOTWORD) { |
| ALOGW("startInput() preempting already started low-priority input %d", activeInput); |
| stopInput(activeInput); |
| releaseInput(activeInput); |
| } else { |
| ALOGW("startInput() input %d failed: other input already started", input); |
| return INVALID_OPERATION; |
| } |
| } |
| } |
| |
| audio_devices_t newDevice = getDeviceForInputSource(inputDesc->mInputSource); |
| if ((newDevice != AUDIO_DEVICE_NONE) && (newDevice != inputDesc->mDevice)) { |
| inputDesc->mDevice = newDevice; |
| } |
| |
| // automatically enable the remote submix output when input is started |
| if (audio_is_remote_submix_device(inputDesc->mDevice)) { |
| setDeviceConnectionState(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, |
| AudioSystem::DEVICE_STATE_AVAILABLE, AUDIO_REMOTE_SUBMIX_DEVICE_ADDRESS); |
| } |
| |
| AudioParameter param = AudioParameter(); |
| param.addInt(String8(AudioParameter::keyRouting), (int)inputDesc->mDevice); |
| |
| int aliasSource = (inputDesc->mInputSource == AUDIO_SOURCE_HOTWORD) ? |
| AUDIO_SOURCE_VOICE_RECOGNITION : inputDesc->mInputSource; |
| |
| param.addInt(String8(AudioParameter::keyInputSource), aliasSource); |
| ALOGV("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) |
| { |
| ALOGV("stopInput() input %d", input); |
| ssize_t index = mInputs.indexOfKey(input); |
| if (index < 0) { |
| ALOGW("stopInput() unknown input %d", input); |
| return BAD_VALUE; |
| } |
| AudioInputDescriptor *inputDesc = mInputs.valueAt(index); |
| |
| if (inputDesc->mRefCount == 0) { |
| ALOGW("stopInput() input %d already stopped", input); |
| return INVALID_OPERATION; |
| } else { |
| // automatically disable the remote submix output when input is stopped |
| if (audio_is_remote_submix_device(inputDesc->mDevice)) { |
| setDeviceConnectionState(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, |
| AudioSystem::DEVICE_STATE_UNAVAILABLE, AUDIO_REMOTE_SUBMIX_DEVICE_ADDRESS); |
| } |
| |
| 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) |
| { |
| ALOGV("releaseInput() %d", input); |
| ssize_t index = mInputs.indexOfKey(input); |
| if (index < 0) { |
| ALOGW("releaseInput() releasing unknown input %d", input); |
| return; |
| } |
| mpClientInterface->closeInput(input); |
| delete mInputs.valueAt(index); |
| mInputs.removeItem(input); |
| |
| ALOGV("releaseInput() exit"); |
| } |
| |
| void AudioPolicyManagerBase::closeAllInputs() { |
| for(size_t input_index = 0; input_index < mInputs.size(); input_index++) { |
| mpClientInterface->closeInput(mInputs.keyAt(input_index)); |
| } |
| mInputs.clear(); |
| } |
| |
| void AudioPolicyManagerBase::initStreamVolume(AudioSystem::stream_type stream, |
| int indexMin, |
| int indexMax) |
| { |
| ALOGV("initStreamVolume() stream %d, min %d, max %d", stream , indexMin, indexMax); |
| if (indexMin < 0 || indexMin >= indexMax) { |
| ALOGW("initStreamVolume() invalid index limits for stream %d, min %d, max %d", stream , indexMin, indexMax); |
| return; |
| } |
| mStreams[stream].mIndexMin = indexMin; |
| mStreams[stream].mIndexMax = indexMax; |
| } |
| |
| status_t AudioPolicyManagerBase::setStreamVolumeIndex(AudioSystem::stream_type stream, |
| int index, |
| audio_devices_t device) |
| { |
| |
| if ((index < mStreams[stream].mIndexMin) || (index > mStreams[stream].mIndexMax)) { |
| return BAD_VALUE; |
| } |
| if (!audio_is_output_device(device)) { |
| return BAD_VALUE; |
| } |
| |
| // Force max volume if stream cannot be muted |
| if (!mStreams[stream].mCanBeMuted) index = mStreams[stream].mIndexMax; |
| |
| ALOGV("setStreamVolumeIndex() stream %d, device %04x, index %d", |
| stream, device, index); |
| |
| // if device is AUDIO_DEVICE_OUT_DEFAULT set default value and |
| // clear all device specific values |
| if (device == AUDIO_DEVICE_OUT_DEFAULT) { |
| mStreams[stream].mIndexCur.clear(); |
| } |
| mStreams[stream].mIndexCur.add(device, index); |
| |
| // 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++) { |
| audio_devices_t curDevice = |
| getDeviceForVolume(mOutputs.valueAt(i)->device()); |
| if ((device == AUDIO_DEVICE_OUT_DEFAULT) || (device == curDevice)) { |
| status_t volStatus = checkAndSetVolume(stream, index, mOutputs.keyAt(i), curDevice); |
| if (volStatus != NO_ERROR) { |
| status = volStatus; |
| } |
| } |
| } |
| return status; |
| } |
| |
| status_t AudioPolicyManagerBase::getStreamVolumeIndex(AudioSystem::stream_type stream, |
| int *index, |
| audio_devices_t device) |
| { |
| if (index == NULL) { |
| return BAD_VALUE; |
| } |
| if (!audio_is_output_device(device)) { |
| return BAD_VALUE; |
| } |
| // if device is AUDIO_DEVICE_OUT_DEFAULT, return volume for device corresponding to |
| // the strategy the stream belongs to. |
| if (device == AUDIO_DEVICE_OUT_DEFAULT) { |
| device = getDeviceForStrategy(getStrategy(stream), true /*fromCache*/); |
| } |
| device = getDeviceForVolume(device); |
| |
| *index = mStreams[stream].getVolumeIndex(device); |
| ALOGV("getStreamVolumeIndex() stream %d device %08x index %d", stream, device, *index); |
| return NO_ERROR; |
| } |
| |
| audio_io_handle_t AudioPolicyManagerBase::selectOutputForEffects( |
| const SortedVector<audio_io_handle_t>& outputs) |
| { |
| // select one output among several suitable for global effects. |
| // The priority is as follows: |
| // 1: An offloaded output. If the effect ends up not being offloadable, |
| // AudioFlinger will invalidate the track and the offloaded output |
| // will be closed causing the effect to be moved to a PCM output. |
| // 2: A deep buffer output |
| // 3: the first output in the list |
| |
| if (outputs.size() == 0) { |
| return 0; |
| } |
| |
| audio_io_handle_t outputOffloaded = 0; |
| audio_io_handle_t outputDeepBuffer = 0; |
| |
| for (size_t i = 0; i < outputs.size(); i++) { |
| AudioOutputDescriptor *desc = mOutputs.valueFor(outputs[i]); |
| ALOGV("selectOutputForEffects outputs[%zu] flags %x", i, desc->mFlags); |
| if ((desc->mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) { |
| outputOffloaded = outputs[i]; |
| } |
| if ((desc->mFlags & AUDIO_OUTPUT_FLAG_DEEP_BUFFER) != 0) { |
| outputDeepBuffer = outputs[i]; |
| } |
| } |
| |
| ALOGV("selectOutputForEffects outputOffloaded %d outputDeepBuffer %d", |
| outputOffloaded, outputDeepBuffer); |
| if (outputOffloaded != 0) { |
| return outputOffloaded; |
| } |
| if (outputDeepBuffer != 0) { |
| return outputDeepBuffer; |
| } |
| |
| return outputs[0]; |
| } |
| |
| audio_io_handle_t AudioPolicyManagerBase::getOutputForEffect(const effect_descriptor_t *desc) |
| { |
| // apply simple rule where global effects are attached to the same output as MUSIC streams |
| |
| routing_strategy strategy = getStrategy(AudioSystem::MUSIC); |
| audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/); |
| SortedVector<audio_io_handle_t> dstOutputs = getOutputsForDevice(device, mOutputs); |
| |
| audio_io_handle_t output = selectOutputForEffects(dstOutputs); |
| ALOGV("getOutputForEffect() got output %d for fx %s flags %x", |
| output, (desc == NULL) ? "unspecified" : desc->name, (desc == NULL) ? 0 : desc->flags); |
| |
| return output; |
| } |
| |
| status_t AudioPolicyManagerBase::registerEffect(const effect_descriptor_t *desc, |
| audio_io_handle_t io, |
| uint32_t strategy, |
| int session, |
| int id) |
| { |
| ssize_t index = mOutputs.indexOfKey(io); |
| if (index < 0) { |
| index = mInputs.indexOfKey(io); |
| if (index < 0) { |
| ALOGW("registerEffect() unknown io %d", io); |
| return INVALID_OPERATION; |
| } |
| } |
| |
| if (mTotalEffectsMemory + desc->memoryUsage > getMaxEffectsMemory()) { |
| ALOGW("registerEffect() memory limit exceeded for Fx %s, Memory %d KB", |
| desc->name, desc->memoryUsage); |
| return INVALID_OPERATION; |
| } |
| mTotalEffectsMemory += desc->memoryUsage; |
| ALOGV("registerEffect() effect %s, io %d, strategy %d session %d id %d", |
| desc->name, io, strategy, session, id); |
| ALOGV("registerEffect() memory %d, total memory %d", desc->memoryUsage, mTotalEffectsMemory); |
| |
| 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) { |
| ALOGW("unregisterEffect() unknown effect ID %d", id); |
| return INVALID_OPERATION; |
| } |
| |
| EffectDescriptor *pDesc = mEffects.valueAt(index); |
| |
| setEffectEnabled(pDesc, false); |
| |
| if (mTotalEffectsMemory < pDesc->mDesc.memoryUsage) { |
| ALOGW("unregisterEffect() memory %d too big for total %d", |
| pDesc->mDesc.memoryUsage, mTotalEffectsMemory); |
| pDesc->mDesc.memoryUsage = mTotalEffectsMemory; |
| } |
| mTotalEffectsMemory -= pDesc->mDesc.memoryUsage; |
| ALOGV("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) { |
| ALOGW("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) { |
| ALOGV("setEffectEnabled(%s) effect already %s", |
| enabled?"true":"false", enabled?"enabled":"disabled"); |
| return INVALID_OPERATION; |
| } |
| |
| if (enabled) { |
| if (mTotalEffectsCpuLoad + pDesc->mDesc.cpuLoad > getMaxEffectsCpuLoad()) { |
| ALOGW("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; |
| ALOGV("setEffectEnabled(true) total CPU %d", mTotalEffectsCpuLoad); |
| } else { |
| if (mTotalEffectsCpuLoad < pDesc->mDesc.cpuLoad) { |
| ALOGW("setEffectEnabled(false) CPU load %d too high for total %d", |
| pDesc->mDesc.cpuLoad, mTotalEffectsCpuLoad); |
| pDesc->mDesc.cpuLoad = mTotalEffectsCpuLoad; |
| } |
| mTotalEffectsCpuLoad -= pDesc->mDesc.cpuLoad; |
| ALOGV("setEffectEnabled(false) total CPU %d", mTotalEffectsCpuLoad); |
| } |
| pDesc->mEnabled = enabled; |
| return NO_ERROR; |
| } |
| |
| bool AudioPolicyManagerBase::isNonOffloadableEffectEnabled() |
| { |
| for (size_t i = 0; i < mEffects.size(); i++) { |
| const EffectDescriptor * const pDesc = mEffects.valueAt(i); |
| if (pDesc->mEnabled && (pDesc->mStrategy == STRATEGY_MEDIA) && |
| ((pDesc->mDesc.flags & EFFECT_FLAG_OFFLOAD_SUPPORTED) == 0)) { |
| ALOGV("isNonOffloadableEffectEnabled() non offloadable effect %s enabled on session %d", |
| pDesc->mDesc.name, pDesc->mSession); |
| return true; |
| } |
| } |
| return false; |
| } |
| |
| bool AudioPolicyManagerBase::isStreamActive(int stream, uint32_t inPastMs) const |
| { |
| nsecs_t sysTime = systemTime(); |
| for (size_t i = 0; i < mOutputs.size(); i++) { |
| const AudioOutputDescriptor *outputDesc = mOutputs.valueAt(i); |
| if (outputDesc->isStreamActive((AudioSystem::stream_type)stream, inPastMs, sysTime)) { |
| return true; |
| } |
| } |
| return false; |
| } |
| |
| bool AudioPolicyManagerBase::isStreamActiveRemotely(int stream, uint32_t inPastMs) const |
| { |
| nsecs_t sysTime = systemTime(); |
| for (size_t i = 0; i < mOutputs.size(); i++) { |
| const AudioOutputDescriptor *outputDesc = mOutputs.valueAt(i); |
| if (((outputDesc->device() & APM_AUDIO_OUT_DEVICE_REMOTE_ALL) != 0) && |
| outputDesc->isStreamActive((AudioSystem::stream_type)stream, inPastMs, sysTime)) { |
| return true; |
| } |
| } |
| return false; |
| } |
| |
| bool AudioPolicyManagerBase::isSourceActive(audio_source_t source) const |
| { |
| for (size_t i = 0; i < mInputs.size(); i++) { |
| const AudioInputDescriptor * inputDescriptor = mInputs.valueAt(i); |
| if ((inputDescriptor->mInputSource == (int)source || |
| (source == (audio_source_t)AUDIO_SOURCE_VOICE_RECOGNITION && |
| inputDescriptor->mInputSource == AUDIO_SOURCE_HOTWORD)) |
| && (inputDescriptor->mRefCount > 0)) { |
| 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, " Primary Output: %d\n", mPrimaryOutput); |
| result.append(buffer); |
| snprintf(buffer, SIZE, " A2DP device address: %s\n", mA2dpDeviceAddress.string()); |
| result.append(buffer); |
| snprintf(buffer, SIZE, " SCO device address: %s\n", mScoDeviceAddress.string()); |
| result.append(buffer); |
| snprintf(buffer, SIZE, " USB audio ALSA %s\n", mUsbOutCardAndDevice.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, " 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); |
| snprintf(buffer, SIZE, " Force use for system %d\n", mForceUse[AudioSystem::FOR_SYSTEM]); |
| result.append(buffer); |
| write(fd, result.string(), result.size()); |
| |
| |
| snprintf(buffer, SIZE, "\nHW Modules dump:\n"); |
| write(fd, buffer, strlen(buffer)); |
| for (size_t i = 0; i < mHwModules.size(); i++) { |
| snprintf(buffer, SIZE, "- HW Module %zu:\n", i + 1); |
| write(fd, buffer, strlen(buffer)); |
| mHwModules[i]->dump(fd); |
| } |
| |
| snprintf(buffer, SIZE, "\nOutputs dump:\n"); |
| write(fd, buffer, strlen(buffer)); |
| for (size_t i = 0; i < mOutputs.size(); i++) { |
| snprintf(buffer, SIZE, "- Output %d dump:\n", mOutputs.keyAt(i)); |
| write(fd, buffer, strlen(buffer)); |
| mOutputs.valueAt(i)->dump(fd); |
| } |
| |
| snprintf(buffer, SIZE, "\nInputs dump:\n"); |
| write(fd, buffer, strlen(buffer)); |
| for (size_t i = 0; i < mInputs.size(); i++) { |
| snprintf(buffer, SIZE, "- Input %d dump:\n", mInputs.keyAt(i)); |
| write(fd, buffer, strlen(buffer)); |
| mInputs.valueAt(i)->dump(fd); |
| } |
| |
| snprintf(buffer, SIZE, "\nStreams dump:\n"); |
| write(fd, buffer, strlen(buffer)); |
| snprintf(buffer, SIZE, |
| " Stream Can be muted Index Min Index Max Index Cur [device : index]...\n"); |
| write(fd, buffer, strlen(buffer)); |
| for (size_t i = 0; i < AudioSystem::NUM_STREAM_TYPES; i++) { |
| snprintf(buffer, SIZE, " %02zu ", i); |
| write(fd, buffer, strlen(buffer)); |
| mStreams[i].dump(fd); |
| } |
| |
| snprintf(buffer, SIZE, "\nTotal Effects CPU: %f MIPS, Total Effects memory: %d KB\n", |
| (float)mTotalEffectsCpuLoad/10, mTotalEffectsMemory); |
| write(fd, buffer, strlen(buffer)); |
| |
| snprintf(buffer, SIZE, "Registered effects:\n"); |
| write(fd, buffer, strlen(buffer)); |
| for (size_t i = 0; i < mEffects.size(); i++) { |
| snprintf(buffer, SIZE, "- Effect %d dump:\n", mEffects.keyAt(i)); |
| write(fd, buffer, strlen(buffer)); |
| mEffects.valueAt(i)->dump(fd); |
| } |
| |
| |
| return NO_ERROR; |
| } |
| |
| // This function checks for the parameters which can be offloaded. |
| // This can be enhanced depending on the capability of the DSP and policy |
| // of the system. |
| bool AudioPolicyManagerBase::isOffloadSupported(const audio_offload_info_t& offloadInfo) |
| { |
| ALOGV("isOffloadSupported: SR=%u, CM=0x%x, Format=0x%x, StreamType=%d," |
| " BitRate=%u, duration=%" PRId64 " us, has_video=%d", |
| offloadInfo.sample_rate, offloadInfo.channel_mask, |
| offloadInfo.format, |
| offloadInfo.stream_type, offloadInfo.bit_rate, offloadInfo.duration_us, |
| offloadInfo.has_video); |
| |
| // Check if offload has been disabled |
| char propValue[PROPERTY_VALUE_MAX]; |
| if (property_get("audio.offload.disable", propValue, "0")) { |
| if (atoi(propValue) != 0) { |
| ALOGV("offload disabled by audio.offload.disable=%s", propValue ); |
| return false; |
| } |
| } |
| |
| // Check if stream type is music, then only allow offload as of now. |
| if (offloadInfo.stream_type != AUDIO_STREAM_MUSIC) |
| { |
| ALOGV("isOffloadSupported: stream_type != MUSIC, returning false"); |
| return false; |
| } |
| |
| //TODO: enable audio offloading with video when ready |
| if (offloadInfo.has_video) |
| { |
| ALOGV("isOffloadSupported: has_video == true, returning false"); |
| return false; |
| } |
| |
| //If duration is less than minimum value defined in property, return false |
| if (property_get("audio.offload.min.duration.secs", propValue, NULL)) { |
| if (offloadInfo.duration_us < (atoi(propValue) * 1000000 )) { |
| ALOGV("Offload denied by duration < audio.offload.min.duration.secs(=%s)", propValue); |
| return false; |
| } |
| } else if (offloadInfo.duration_us < OFFLOAD_DEFAULT_MIN_DURATION_SECS * 1000000) { |
| ALOGV("Offload denied by duration < default min(=%u)", OFFLOAD_DEFAULT_MIN_DURATION_SECS); |
| return false; |
| } |
| |
| // Do not allow offloading if one non offloadable effect is enabled. This prevents from |
| // creating an offloaded track and tearing it down immediately after start when audioflinger |
| // detects there is an active non offloadable effect. |
| // FIXME: We should check the audio session here but we do not have it in this context. |
| // This may prevent offloading in rare situations where effects are left active by apps |
| // in the background. |
| if (isNonOffloadableEffectEnabled()) { |
| return false; |
| } |
| |
| // See if there is a profile to support this. |
| // AUDIO_DEVICE_NONE |
| IOProfile *profile = getProfileForDirectOutput(AUDIO_DEVICE_NONE /*ignore device */, |
| offloadInfo.sample_rate, |
| offloadInfo.format, |
| offloadInfo.channel_mask, |
| AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD); |
| ALOGV("isOffloadSupported() profile %sfound", profile != NULL ? "" : "NOT "); |
| return (profile != NULL); |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // AudioPolicyManagerBase |
| // ---------------------------------------------------------------------------- |
| |
| AudioPolicyManagerBase::AudioPolicyManagerBase(AudioPolicyClientInterface *clientInterface) |
| : |
| #ifdef AUDIO_POLICY_TEST |
| Thread(false), |
| #endif //AUDIO_POLICY_TEST |
| mPrimaryOutput((audio_io_handle_t)0), |
| mAvailableOutputDevices(AUDIO_DEVICE_NONE), |
| mPhoneState(AudioSystem::MODE_NORMAL), |
| mLimitRingtoneVolume(false), mLastVoiceVolume(-1.0f), |
| mTotalEffectsCpuLoad(0), mTotalEffectsMemory(0), |
| mA2dpSuspended(false), mHasA2dp(false), mHasUsb(false), mHasRemoteSubmix(false), |
| mSpeakerDrcEnabled(false) |
| { |
| mpClientInterface = clientInterface; |
| |
| for (int i = 0; i < AudioSystem::NUM_FORCE_USE; i++) { |
| mForceUse[i] = AudioSystem::FORCE_NONE; |
| } |
| |
| mA2dpDeviceAddress = String8(""); |
| mScoDeviceAddress = String8(""); |
| mUsbOutCardAndDevice = String8(""); |
| |
| if (loadAudioPolicyConfig(AUDIO_POLICY_VENDOR_CONFIG_FILE) != NO_ERROR) { |
| if (loadAudioPolicyConfig(AUDIO_POLICY_CONFIG_FILE) != NO_ERROR) { |
| ALOGE("could not load audio policy configuration file, setting defaults"); |
| defaultAudioPolicyConfig(); |
| } |
| } |
| |
| // must be done after reading the policy |
| initializeVolumeCurves(); |
| |
| // open all output streams needed to access attached devices |
| for (size_t i = 0; i < mHwModules.size(); i++) { |
| mHwModules[i]->mHandle = mpClientInterface->loadHwModule(mHwModules[i]->mName); |
| if (mHwModules[i]->mHandle == 0) { |
| ALOGW("could not open HW module %s", mHwModules[i]->mName); |
| continue; |
| } |
| // open all output streams needed to access attached devices |
| // except for direct output streams that are only opened when they are actually |
| // required by an app. |
| for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++) |
| { |
| const IOProfile *outProfile = mHwModules[i]->mOutputProfiles[j]; |
| |
| if ((outProfile->mSupportedDevices & mAttachedOutputDevices) && |
| ((outProfile->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) == 0)) { |
| AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(outProfile); |
| outputDesc->mDevice = (audio_devices_t)(mDefaultOutputDevice & |
| outProfile->mSupportedDevices); |
| audio_io_handle_t output = mpClientInterface->openOutput( |
| outProfile->mModule->mHandle, |
| &outputDesc->mDevice, |
| &outputDesc->mSamplingRate, |
| &outputDesc->mFormat, |
| &outputDesc->mChannelMask, |
| &outputDesc->mLatency, |
| outputDesc->mFlags); |
| if (output == 0) { |
| delete outputDesc; |
| } else { |
| mAvailableOutputDevices = (audio_devices_t)(mAvailableOutputDevices | |
| (outProfile->mSupportedDevices & mAttachedOutputDevices)); |
| if (mPrimaryOutput == 0 && |
| outProfile->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY) { |
| mPrimaryOutput = output; |
| } |
| addOutput(output, outputDesc); |
| setOutputDevice(output, |
| (audio_devices_t)(mDefaultOutputDevice & |
| outProfile->mSupportedDevices), |
| true); |
| } |
| } |
| } |
| } |
| |
| ALOGE_IF((mAttachedOutputDevices & ~mAvailableOutputDevices), |
| "Not output found for attached devices %08x", |
| (mAttachedOutputDevices & ~mAvailableOutputDevices)); |
| |
| ALOGE_IF((mPrimaryOutput == 0), "Failed to open primary output"); |
| |
| updateDevicesAndOutputs(); |
| |
| #ifdef AUDIO_POLICY_TEST |
| if (mPrimaryOutput != 0) { |
| AudioParameter outputCmd = AudioParameter(); |
| outputCmd.addInt(String8("set_id"), 0); |
| mpClientInterface->setParameters(mPrimaryOutput, outputCmd.toString()); |
| |
| mTestDevice = AUDIO_DEVICE_OUT_SPEAKER; |
| mTestSamplingRate = 44100; |
| mTestFormat = 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); |
| } |
| for (size_t i = 0; i < mInputs.size(); i++) { |
| mpClientInterface->closeInput(mInputs.keyAt(i)); |
| delete mInputs.valueAt(i); |
| } |
| for (size_t i = 0; i < mHwModules.size(); i++) { |
| delete mHwModules[i]; |
| } |
| } |
| |
| status_t AudioPolicyManagerBase::initCheck() |
| { |
| return (mPrimaryOutput == 0) ? NO_INIT : NO_ERROR; |
| } |
| |
| #ifdef AUDIO_POLICY_TEST |
| bool AudioPolicyManagerBase::threadLoop() |
| { |
| ALOGV("entering threadLoop()"); |
| while (!exitPending()) |
| { |
| String8 command; |
| int valueInt; |
| String8 value; |
| |
| Mutex::Autolock _l(mLock); |
| mWaitWorkCV.waitRelative(mLock, milliseconds(50)); |
| |
| command = mpClientInterface->getParameters(0, String8("test_cmd_policy")); |
| AudioParameter param = AudioParameter(command); |
| |
| if (param.getInt(String8("test_cmd_policy"), valueInt) == NO_ERROR && |
| valueInt != 0) { |
| ALOGV("Test command %s received", command.string()); |
| String8 target; |
| if (param.get(String8("target"), target) != NO_ERROR) { |
| target = "Manager"; |
| } |
| if (param.getInt(String8("test_cmd_policy_output"), valueInt) == NO_ERROR) { |
| param.remove(String8("test_cmd_policy_output")); |
| mCurOutput = valueInt; |
| } |
| if (param.get(String8("test_cmd_policy_direct"), value) == NO_ERROR) { |
| param.remove(String8("test_cmd_policy_direct")); |
| if (value == "false") { |
| mDirectOutput = false; |
| } else if (value == "true") { |
| mDirectOutput = true; |
| } |
| } |
| if (param.getInt(String8("test_cmd_policy_input"), valueInt) == NO_ERROR) { |
| param.remove(String8("test_cmd_policy_input")); |
| mTestInput = valueInt; |
| } |
| |
| if (param.get(String8("test_cmd_policy_format"), value) == NO_ERROR) { |
| param.remove(String8("test_cmd_policy_format")); |
| int format = 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")); |
| |
| AudioOutputDescriptor *outputDesc = mOutputs.valueFor(mPrimaryOutput); |
| mpClientInterface->closeOutput(mPrimaryOutput); |
| |
| audio_module_handle_t moduleHandle = outputDesc->mModule->mHandle; |
| |
| delete mOutputs.valueFor(mPrimaryOutput); |
| mOutputs.removeItem(mPrimaryOutput); |
| |
| AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(NULL); |
| outputDesc->mDevice = AUDIO_DEVICE_OUT_SPEAKER; |
| mPrimaryOutput = mpClientInterface->openOutput(moduleHandle, |
| &outputDesc->mDevice, |
| &outputDesc->mSamplingRate, |
| &outputDesc->mFormat, |
| &outputDesc->mChannelMask, |
| &outputDesc->mLatency, |
| outputDesc->mFlags); |
| if (mPrimaryOutput == 0) { |
| ALOGE("Failed to reopen hardware output stream, samplingRate: %d, format %d, channels %d", |
| outputDesc->mSamplingRate, outputDesc->mFormat, outputDesc->mChannelMask); |
| } else { |
| AudioParameter outputCmd = AudioParameter(); |
| outputCmd.addInt(String8("set_id"), 0); |
| mpClientInterface->setParameters(mPrimaryOutput, outputCmd.toString()); |
| addOutput(mPrimaryOutput, outputDesc); |
| } |
| } |
| |
| |
| mpClientInterface->setParameters(0, String8("test_cmd_policy=")); |
| } |
| } |
| return false; |
| } |
| |
| void 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); |
| } |
| |
| void AudioPolicyManagerBase::addInput(audio_io_handle_t id, AudioInputDescriptor *inputDesc) |
| { |
| inputDesc->mId = id; |
| mInputs.add(id, inputDesc); |
| } |
| |
| status_t AudioPolicyManagerBase::checkOutputsForDevice(audio_devices_t device, |
| AudioSystem::device_connection_state state, |
| SortedVector<audio_io_handle_t>& outputs, |
| const String8 paramStr) |
| { |
| AudioOutputDescriptor *desc; |
| |
| if (state == AudioSystem::DEVICE_STATE_AVAILABLE) { |
| // first list already open outputs that can be routed to this device |
| for (size_t i = 0; i < mOutputs.size(); i++) { |
| desc = mOutputs.valueAt(i); |
| if (!desc->isDuplicated() && (desc->mProfile->mSupportedDevices & device)) { |
| ALOGV("checkOutputsForDevice(): adding opened output %d", mOutputs.keyAt(i)); |
| outputs.add(mOutputs.keyAt(i)); |
| } |
| } |
| // then look for output profiles that can be routed to this device |
| SortedVector<IOProfile *> profiles; |
| for (size_t i = 0; i < mHwModules.size(); i++) |
| { |
| if (mHwModules[i]->mHandle == 0) { |
| continue; |
| } |
| for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++) |
| { |
| if (mHwModules[i]->mOutputProfiles[j]->mSupportedDevices & device) { |
| ALOGV("checkOutputsForDevice(): adding profile %zu from module %zu", j, i); |
| profiles.add(mHwModules[i]->mOutputProfiles[j]); |
| } |
| } |
| } |
| |
| if (profiles.isEmpty() && outputs.isEmpty()) { |
| ALOGW("checkOutputsForDevice(): No output available for device %04x", device); |
| return BAD_VALUE; |
| } |
| |
| // open outputs for matching profiles if needed. Direct outputs are also opened to |
| // query for dynamic parameters and will be closed later by setDeviceConnectionState() |
| for (ssize_t profile_index = 0; profile_index < (ssize_t)profiles.size(); profile_index++) { |
| IOProfile *profile = profiles[profile_index]; |
| |
| // nothing to do if one output is already opened for this profile |
| size_t j; |
| for (j = 0; j < mOutputs.size(); j++) { |
| desc = mOutputs.valueAt(j); |
| if (!desc->isDuplicated() && desc->mProfile == profile) { |
| break; |
| } |
| } |
| if (j != mOutputs.size()) { |
| continue; |
| } |
| |
| ALOGV("opening output for device %08x with params %s", device, paramStr.string()); |
| desc = new AudioOutputDescriptor(profile); |
| desc->mDevice = device; |
| audio_offload_info_t offloadInfo = AUDIO_INFO_INITIALIZER; |
| offloadInfo.sample_rate = desc->mSamplingRate; |
| offloadInfo.format = desc->mFormat; |
| offloadInfo.channel_mask = desc->mChannelMask; |
| |
| audio_io_handle_t output = mpClientInterface->openOutput(profile->mModule->mHandle, |
| &desc->mDevice, |
| &desc->mSamplingRate, |
| &desc->mFormat, |
| &desc->mChannelMask, |
| &desc->mLatency, |
| desc->mFlags, |
| &offloadInfo); |
| if (output != 0) { |
| if (!paramStr.isEmpty()) { |
| // Here is where the out_set_parameters() for card & device gets called |
| mpClientInterface->setParameters(output, paramStr); |
| } |
| |
| // Here is where we step through and resolve any "dynamic" fields |
| String8 reply; |
| char *value; |
| if (profile->mSamplingRates[0] == 0) { |
| reply = mpClientInterface->getParameters(output, |
| String8(AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES)); |
| ALOGV("checkOutputsForDevice() direct output sup sampling rates %s", |
| reply.string()); |
| value = strpbrk((char *)reply.string(), "="); |
| if (value != NULL) { |
| loadSamplingRates(value + 1, profile); |
| } |
| } |
| if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) { |
| reply = mpClientInterface->getParameters(output, |
| String8(AUDIO_PARAMETER_STREAM_SUP_FORMATS)); |
| ALOGV("checkOutputsForDevice() direct output sup formats %s", |
| reply.string()); |
| value = strpbrk((char *)reply.string(), "="); |
| if (value != NULL) { |
| loadFormats(value + 1, profile); |
| } |
| } |
| if (profile->mChannelMasks[0] == 0) { |
| reply = mpClientInterface->getParameters(output, |
| String8(AUDIO_PARAMETER_STREAM_SUP_CHANNELS)); |
| ALOGV("checkOutputsForDevice() direct output sup channel masks %s", |
| reply.string()); |
| value = strpbrk((char *)reply.string(), "="); |
| if (value != NULL) { |
| loadOutChannels(value + 1, profile); |
| } |
| } |
| if (((profile->mSamplingRates[0] == 0) && |
| (profile->mSamplingRates.size() < 2)) || |
| ((profile->mFormats[0] == 0) && |
| (profile->mFormats.size() < 2)) || |
| ((profile->mChannelMasks[0] == 0) && |
| (profile->mChannelMasks.size() < 2))) { |
| ALOGW("checkOutputsForDevice() direct output missing param"); |
| mpClientInterface->closeOutput(output); |
| output = 0; |
| } else if (profile->mSamplingRates[0] == 0) { |
| mpClientInterface->closeOutput(output); |
| desc->mSamplingRate = profile->mSamplingRates[1]; |
| offloadInfo.sample_rate = desc->mSamplingRate; |
| output = mpClientInterface->openOutput( |
| profile->mModule->mHandle, |
| &desc->mDevice, |
| &desc->mSamplingRate, |
| &desc->mFormat, |
| &desc->mChannelMask, |
| &desc->mLatency, |
| desc->mFlags, |
| &offloadInfo); |
| } |
| |
| if (output != 0) { |
| addOutput(output, desc); |
| if ((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) == 0) { |
| audio_io_handle_t duplicatedOutput = 0; |
| |
| // set initial stream volume for device |
| applyStreamVolumes(output, device, 0, true); |
| |
| //TODO: configure audio effect output stage here |
| |
| // open a duplicating output thread for the new output and the primary output |
| duplicatedOutput = mpClientInterface->openDuplicateOutput(output, |
| mPrimaryOutput); |
| if (duplicatedOutput != 0) { |
| // add duplicated output descriptor |
| AudioOutputDescriptor *dupOutputDesc = new AudioOutputDescriptor(NULL); |
| dupOutputDesc->mOutput1 = mOutputs.valueFor(mPrimaryOutput); |
| dupOutputDesc->mOutput2 = mOutputs.valueFor(output); |
| dupOutputDesc->mSamplingRate = desc->mSamplingRate; |
| dupOutputDesc->mFormat = desc->mFormat; |
| dupOutputDesc->mChannelMask = desc->mChannelMask; |
| dupOutputDesc->mLatency = desc->mLatency; |
| addOutput(duplicatedOutput, dupOutputDesc); |
| applyStreamVolumes(duplicatedOutput, device, 0, true); |
| } else { |
| ALOGW("checkOutputsForDevice() could not open dup output for %d and %d", |
| mPrimaryOutput, output); |
| mpClientInterface->closeOutput(output); |
| mOutputs.removeItem(output); |
| output = 0; |
| } |
| } |
| } |
| } |
| if (output == 0) { |
| ALOGW("checkOutputsForDevice() could not open output for device %x", device); |
| delete desc; |
| profiles.removeAt(profile_index); |
| profile_index--; |
| } else { |
| outputs.add(output); |
| ALOGV("checkOutputsForDevice(): adding output %d", output); |
| } |
| } |
| |
| if (profiles.isEmpty()) { |
| ALOGW("checkOutputsForDevice(): No output available for device %04x", device); |
| return BAD_VALUE; |
| } |
| } else { // Disconnect |
| // check if one opened output is not needed any more after disconnecting one device |
| for (size_t i = 0; i < mOutputs.size(); i++) { |
| desc = mOutputs.valueAt(i); |
| if (!desc->isDuplicated() && |
| !(desc->mProfile->mSupportedDevices & mAvailableOutputDevices)) { |
| ALOGV("checkOutputsForDevice(): disconnecting adding output %d", mOutputs.keyAt(i)); |
| outputs.add(mOutputs.keyAt(i)); |
| } |
| } |
| // Clear any profiles associated with the disconnected device. |
| for (size_t i = 0; i < mHwModules.size(); i++) |
| { |
| if (mHwModules[i]->mHandle == 0) { |
| continue; |
| } |
| for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++) |
| { |
| IOProfile *profile = mHwModules[i]->mOutputProfiles[j]; |
| if (profile->mSupportedDevices & device) { |
| ALOGV("checkOutputsForDevice(): clearing direct output profile %zu on module %zu", |
| j, i); |
| if (profile->mSamplingRates[0] == 0) { |
| profile->mSamplingRates.clear(); |
| profile->mSamplingRates.add(0); |
| } |
| if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) { |
| profile->mFormats.clear(); |
| profile->mFormats.add(AUDIO_FORMAT_DEFAULT); |
| } |
| if (profile->mChannelMasks[0] == 0) { |
| profile->mChannelMasks.clear(); |
| profile->mChannelMasks.add(0); |
| } |
| } |
| } |
| } |
| } |
| return NO_ERROR; |
| } |
| |
| status_t AudioPolicyManagerBase::checkInputsForDevice(audio_devices_t device, |
| AudioSystem::device_connection_state state, |
| SortedVector<audio_io_handle_t>& inputs, |
| const String8 paramStr) |
| { |
| AudioInputDescriptor *desc; |
| if (state == AudioSystem::DEVICE_STATE_AVAILABLE) { |
| // first list already open inputs that can be routed to this device |
| for (size_t input_index = 0; input_index < mInputs.size(); input_index++) { |
| desc = mInputs.valueAt(input_index); |
| if (desc->mProfile->mSupportedDevices & (device & ~AUDIO_DEVICE_BIT_IN)) { |
| ALOGV("checkInputsForDevice(): adding opened input %d", mInputs.keyAt(input_index)); |
| inputs.add(mInputs.keyAt(input_index)); |
| } |
| } |
| |
| // then look for input profiles that can be routed to this device |
| SortedVector<IOProfile *> profiles; |
| for (size_t module_index = 0; module_index < mHwModules.size(); module_index++) |
| { |
| if (mHwModules[module_index]->mHandle == 0) { |
| continue; |
| } |
| for (size_t profile_index = 0; |
| profile_index < mHwModules[module_index]->mInputProfiles.size(); |
| profile_index++) |
| { |
| if (mHwModules[module_index]->mInputProfiles[profile_index]->mSupportedDevices |
| & (device & ~AUDIO_DEVICE_BIT_IN)) { |
| ALOGV("checkInputsForDevice(): adding profile %zu from module %zu", |
| profile_index, module_index); |
| profiles.add(mHwModules[module_index]->mInputProfiles[profile_index]); |
| } |
| } |
| } |
| |
| if (profiles.isEmpty() && inputs.isEmpty()) { |
| ALOGW("checkInputsForDevice(): No input available for device 0x%X", device); |
| return BAD_VALUE; |
| } |
| |
| // open inputs for matching profiles if needed. Direct inputs are also opened to |
| // query for dynamic parameters and will be closed later by setDeviceConnectionState() |
| for (ssize_t profile_index = 0; profile_index < (ssize_t)profiles.size(); profile_index++) { |
| |
| IOProfile *profile = profiles[profile_index]; |
| // nothing to do if one input is already opened for this profile |
| size_t input_index; |
| for (input_index = 0; input_index < mInputs.size(); input_index++) { |
| desc = mInputs.valueAt(input_index); |
| if (desc->mProfile == profile) { |
| break; |
| } |
| } |
| if (input_index != mInputs.size()) { |
| continue; |
| } |
| |
| ALOGV("opening input for device 0x%X with params %s", device, paramStr.string()); |
| desc = new AudioInputDescriptor(profile); |
| desc->mDevice = device; |
| |
| audio_io_handle_t input = mpClientInterface->openInput(profile->mModule->mHandle, |
| &desc->mDevice, |
| &desc->mSamplingRate, |
| &desc->mFormat, |
| &desc->mChannelMask); |
| |
| if (input != 0) { |
| if (!paramStr.isEmpty()) { |
| mpClientInterface->setParameters(input, paramStr); |
| } |
| |
| // Here is where we step through and resolve any "dynamic" fields |
| String8 reply; |
| char *value; |
| if (profile->mSamplingRates[0] == 0) { |
| reply = mpClientInterface->getParameters(input, |
| String8(AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES)); |
| ALOGV("checkInputsForDevice() direct input sup sampling rates %s", |
| reply.string()); |
| value = strpbrk((char *)reply.string(), "="); |
| if (value != NULL) { |
| loadSamplingRates(value + 1, profile); |
| } |
| } |
| if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) { |
| reply = mpClientInterface->getParameters(input, |
| String8(AUDIO_PARAMETER_STREAM_SUP_FORMATS)); |
| ALOGV("checkInputsForDevice() direct input sup formats %s", reply.string()); |
| value = strpbrk((char *)reply.string(), "="); |
| if (value != NULL) { |
| loadFormats(value + 1, profile); |
| } |
| } |
| if (profile->mChannelMasks[0] == 0) { |
| reply = mpClientInterface->getParameters(input, |
| String8(AUDIO_PARAMETER_STREAM_SUP_CHANNELS)); |
| ALOGV("checkInputsForDevice() direct input sup channel masks %s", |
| reply.string()); |
| value = strpbrk((char *)reply.string(), "="); |
| if (value != NULL) { |
| loadInChannels(value + 1, profile); |
| } |
| } |
| if (((profile->mSamplingRates[0] == 0) && (profile->mSamplingRates.size() < 2)) || |
| ((profile->mFormats[0] == 0) && (profile->mFormats.size() < 2)) || |
| ((profile->mChannelMasks[0] == 0) && (profile->mChannelMasks.size() < 2))) { |
| ALOGW("checkInputsForDevice() direct input missing param"); |
| mpClientInterface->closeInput(input); |
| input = 0; |
| } |
| |
| if (input != 0) { |
| addInput(input, desc); |
| } |
| } // endif input != 0 |
| |
| if (input == 0) { |
| ALOGW("checkInputsForDevice() could not open input for device 0x%X", device); |
| delete desc; |
| profiles.removeAt(profile_index); |
| profile_index--; |
| } else { |
| inputs.add(input); |
| ALOGV("checkInputsForDevice(): adding input %d", input); |
| } |
| } // end scan profiles |
| |
| if (profiles.isEmpty()) { |
| ALOGW("checkInputsForDevice(): No input available for device 0x%X", device); |
| return BAD_VALUE; |
| } |
| } else { |
| // Disconnect |
| // check if one opened input is not needed any more after disconnecting one device |
| for (size_t input_index = 0; input_index < mInputs.size(); input_index++) { |
| desc = mInputs.valueAt(input_index); |
| if (!(desc->mProfile->mSupportedDevices & mAvailableInputDevices)) { |
| ALOGV("checkInputsForDevice(): disconnecting adding input %d", |
| mInputs.keyAt(input_index)); |
| inputs.add(mInputs.keyAt(input_index)); |
| } |
| } |
| // Clear any profiles associated with the disconnected device. |
| for (size_t module_index = 0; module_index < mHwModules.size(); module_index++) |
| { |
| if (mHwModules[module_index]->mHandle == 0) { |
| continue; |
| } |
| for (size_t profile_index = 0; |
| profile_index < mHwModules[module_index]->mInputProfiles.size(); |
| profile_index++) |
| { |
| IOProfile *profile = mHwModules[module_index]->mInputProfiles[profile_index]; |
| if (profile->mSupportedDevices & device) { |
| ALOGV("checkInputsForDevice(): clearing direct input profile %zu on module %zu", |
| profile_index, module_index); |
| if (profile->mSamplingRates[0] == 0) { |
| profile->mSamplingRates.clear(); |
| profile->mSamplingRates.add(0); |
| } |
| if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) { |
| profile->mFormats.clear(); |
| profile->mFormats.add(AUDIO_FORMAT_DEFAULT); |
| } |
| if (profile->mChannelMasks[0] == 0) { |
| profile->mChannelMasks.clear(); |
| profile->mChannelMasks.add(0); |
| } |
| } |
| } |
| } |
| } // end disconnect |
| |
| return NO_ERROR; |
| } |
| |
| void AudioPolicyManagerBase::closeOutput(audio_io_handle_t output) |
| { |
| ALOGV("closeOutput(%d)", output); |
| |
| AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output); |
| if (outputDesc == NULL) { |
| ALOGW("closeOutput() unknown output %d", output); |
| return; |
| } |
| |
| // look for duplicated outputs connected to the output being removed. |
| for (size_t i = 0; i < mOutputs.size(); i++) { |
| AudioOutputDescriptor *dupOutputDesc = mOutputs.valueAt(i); |
| if (dupOutputDesc->isDuplicated() && |
| (dupOutputDesc->mOutput1 == outputDesc || |
| dupOutputDesc->mOutput2 == outputDesc)) { |
| AudioOutputDescriptor *outputDesc2; |
| if (dupOutputDesc->mOutput1 == outputDesc) { |
| outputDesc2 = dupOutputDesc->mOutput2; |
| } else { |
| outputDesc2 = dupOutputDesc->mOutput1; |
| } |
| // As all active tracks on duplicated output will be deleted, |
| // and as they were also referenced on the other output, the reference |
| // count for their stream type must be adjusted accordingly on |
| // the other output. |
| for (int j = 0; j < (int)AudioSystem::NUM_STREAM_TYPES; j++) { |
| int refCount = dupOutputDesc->mRefCount[j]; |
| outputDesc2->changeRefCount((AudioSystem::stream_type)j,-refCount); |
| } |
| audio_io_handle_t duplicatedOutput = mOutputs.keyAt(i); |
| ALOGV("closeOutput() closing also duplicated output %d", duplicatedOutput); |
| |
| mpClientInterface->closeOutput(duplicatedOutput); |
| delete mOutputs.valueFor(duplicatedOutput); |
| mOutputs.removeItem(duplicatedOutput); |
| } |
| } |
| |
| AudioParameter param; |
| param.add(String8("closing"), String8("true")); |
| mpClientInterface->setParameters(output, param.toString()); |
| |
| mpClientInterface->closeOutput(output); |
| delete outputDesc; |
| mOutputs.removeItem(output); |
| mPreviousOutputs = mOutputs; |
| } |
| |
| SortedVector<audio_io_handle_t> AudioPolicyManagerBase::getOutputsForDevice(audio_devices_t device, |
| DefaultKeyedVector<audio_io_handle_t, AudioOutputDescriptor *> openOutputs) |
| { |
| SortedVector<audio_io_handle_t> outputs; |
| |
| ALOGVV("getOutputsForDevice() device %04x", device); |
| for (size_t i = 0; i < openOutputs.size(); i++) { |
| ALOGVV("output %d isDuplicated=%d device=%04x", |
| i, openOutputs.valueAt(i)->isDuplicated(), openOutputs.valueAt(i)->supportedDevices()); |
| if ((device & openOutputs.valueAt(i)->supportedDevices()) == device) { |
| ALOGVV("getOutputsForDevice() found output %d", openOutputs.keyAt(i)); |
| outputs.add(openOutputs.keyAt(i)); |
| } |
| } |
| return outputs; |
| } |
| |
| bool AudioPolicyManagerBase::vectorsEqual(SortedVector<audio_io_handle_t>& outputs1, |
| SortedVector<audio_io_handle_t>& outputs2) |
| { |
| if (outputs1.size() != outputs2.size()) { |
| return false; |
| } |
| for (size_t i = 0; i < outputs1.size(); i++) { |
| if (outputs1[i] != outputs2[i]) { |
| return false; |
| } |
| } |
| return true; |
| } |
| |
| void AudioPolicyManagerBase::checkOutputForStrategy(routing_strategy strategy) |
| { |
| audio_devices_t oldDevice = getDeviceForStrategy(strategy, true /*fromCache*/); |
| audio_devices_t newDevice = getDeviceForStrategy(strategy, false /*fromCache*/); |
| SortedVector<audio_io_handle_t> srcOutputs = getOutputsForDevice(oldDevice, mPreviousOutputs); |
| SortedVector<audio_io_handle_t> dstOutputs = getOutputsForDevice(newDevice, mOutputs); |
| |
| if (!vectorsEqual(srcOutputs,dstOutputs)) { |
| ALOGV("checkOutputForStrategy() strategy %d, moving from output %d to output %d", |
| strategy, srcOutputs[0], dstOutputs[0]); |
| // mute strategy while moving tracks from one output to another |
| for (size_t i = 0; i < srcOutputs.size(); i++) { |
| AudioOutputDescriptor *desc = mOutputs.valueFor(srcOutputs[i]); |
| if (desc->isStrategyActive(strategy)) { |
| setStrategyMute(strategy, true, srcOutputs[i]); |
| setStrategyMute(strategy, false, srcOutputs[i], MUTE_TIME_MS, newDevice); |
| } |
| } |
| |
| // Move effects associated to this strategy from previous output to new output |
| if (strategy == STRATEGY_MEDIA) { |
| audio_io_handle_t fxOutput = selectOutputForEffects(dstOutputs); |
| SortedVector<audio_io_handle_t> moved; |
| for (size_t i = 0; i < mEffects.size(); i++) { |
| EffectDescriptor *desc = mEffects.valueAt(i); |
| if (desc->mSession == AUDIO_SESSION_OUTPUT_MIX && |
| desc->mIo != fxOutput) { |
| if (moved.indexOf(desc->mIo) < 0) { |
| ALOGV("checkOutputForStrategy() moving effect %d to output %d", |
| mEffects.keyAt(i), fxOutput); |
| mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, desc->mIo, |
| fxOutput); |
| moved.add(desc->mIo); |
| } |
| desc->mIo = fxOutput; |
| } |
| } |
| } |
| // 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->invalidateStream((AudioSystem::stream_type)i); |
| } |
| } |
| } |
| } |
| |
| void AudioPolicyManagerBase::checkOutputForAllStrategies() |
| { |
| checkOutputForStrategy(STRATEGY_ENFORCED_AUDIBLE); |
| checkOutputForStrategy(STRATEGY_PHONE); |
| checkOutputForStrategy(STRATEGY_SONIFICATION); |
| checkOutputForStrategy(STRATEGY_SONIFICATION_RESPECTFUL); |
| checkOutputForStrategy(STRATEGY_MEDIA); |
| checkOutputForStrategy(STRATEGY_DTMF); |
| } |
| |
| audio_io_handle_t AudioPolicyManagerBase::getA2dpOutput() |
| { |
| if (!mHasA2dp) { |
| return 0; |
| } |
| |
| for (size_t i = 0; i < mOutputs.size(); i++) { |
| AudioOutputDescriptor *outputDesc = mOutputs.valueAt(i); |
| if (!outputDesc->isDuplicated() && outputDesc->device() & AUDIO_DEVICE_OUT_ALL_A2DP) { |
| return mOutputs.keyAt(i); |
| } |
| } |
| |
| return 0; |
| } |
| |
| void AudioPolicyManagerBase::checkA2dpSuspend() |
| { |
| if (!mHasA2dp) { |
| return; |
| } |
| audio_io_handle_t a2dpOutput = getA2dpOutput(); |
| if (a2dpOutput == 0) { |
| return; |
| } |
| |
| // suspend A2DP output if: |
| // (NOT already suspended) && |
| // ((SCO device is connected && |
| // (forced usage for communication || for record is SCO))) || |
| // (phone state is ringing || in call) |
| // |
| // restore A2DP output if: |
| // (Already suspended) && |
| // ((SCO device is NOT connected || |
| // (forced usage NOT for communication && NOT for record is SCO))) && |
| // (phone state is NOT ringing && NOT in call) |
| // |
| if (mA2dpSuspended) { |
| if (((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(a2dpOutput); |
| 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(a2dpOutput); |
| mA2dpSuspended = true; |
| } |
| } |
| } |
| |
| audio_devices_t AudioPolicyManagerBase::getNewDevice(audio_io_handle_t output, bool fromCache) |
| { |
| audio_devices_t device = AUDIO_DEVICE_NONE; |
| |
| 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 "respectful" sonification is active on the output: |
| // use device for strategy "respectful" sonification |
| // 5: the strategy media is active on the output: |
| // use device for strategy media |
| // 6: the strategy DTMF is active on the output: |
| // use device for strategy DTMF |
| if (outputDesc->isStrategyActive(STRATEGY_ENFORCED_AUDIBLE)) { |
| device = getDeviceForStrategy(STRATEGY_ENFORCED_AUDIBLE, fromCache); |
| } else if (isInCall() || |
| outputDesc->isStrategyActive(STRATEGY_PHONE)) { |
| device = getDeviceForStrategy(STRATEGY_PHONE, fromCache); |
| } else if (outputDesc->isStrategyActive(STRATEGY_SONIFICATION)) { |
| device = getDeviceForStrategy(STRATEGY_SONIFICATION, fromCache); |
| } else if (outputDesc->isStrategyActive(STRATEGY_SONIFICATION_RESPECTFUL)) { |
| device = getDeviceForStrategy(STRATEGY_SONIFICATION_RESPECTFUL, fromCache); |
| } else if (outputDesc->isStrategyActive(STRATEGY_MEDIA)) { |
| device = getDeviceForStrategy(STRATEGY_MEDIA, fromCache); |
| } else if (outputDesc->isStrategyActive(STRATEGY_DTMF)) { |
| device = getDeviceForStrategy(STRATEGY_DTMF, fromCache); |
| } |
| |
| ALOGV("getNewDevice() selected device %x", device); |
| return device; |
| } |
| |
| uint32_t AudioPolicyManagerBase::getStrategyForStream(AudioSystem::stream_type stream) { |
| return (uint32_t)getStrategy(stream); |
| } |
| |
| audio_devices_t AudioPolicyManagerBase::getDevicesForStream(AudioSystem::stream_type stream) { |
| audio_devices_t devices; |
| // By checking the range of stream before calling getStrategy, we avoid |
| // getStrategy's behavior for invalid streams. getStrategy would do a ALOGE |
| // and then return STRATEGY_MEDIA, but we want to return the empty set. |
| if (stream < (AudioSystem::stream_type) 0 || stream >= AudioSystem::NUM_STREAM_TYPES) { |
| devices = AUDIO_DEVICE_NONE; |
| } else { |
| AudioPolicyManagerBase::routing_strategy strategy = getStrategy(stream); |
| devices = getDeviceForStrategy(strategy, true /*fromCache*/); |
| } |
| return devices; |
| } |
| |
| AudioPolicyManagerBase::routing_strategy AudioPolicyManagerBase::getStrategy( |
| AudioSystem::stream_type stream) { |
| // stream to strategy mapping |
| switch (stream) { |
| case AudioSystem::VOICE_CALL: |
| case AudioSystem::BLUETOOTH_SCO: |
| return STRATEGY_PHONE; |
| case AudioSystem::RING: |
| case AudioSystem::ALARM: |
| return STRATEGY_SONIFICATION; |
| case AudioSystem::NOTIFICATION: |
| return STRATEGY_SONIFICATION_RESPECTFUL; |
| case AudioSystem::DTMF: |
| return STRATEGY_DTMF; |
| default: |
| ALOGE("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; |
| } |
| } |
| |
| void AudioPolicyManagerBase::handleNotificationRoutingForStream(AudioSystem::stream_type stream) { |
| switch(stream) { |
| case AudioSystem::MUSIC: |
| checkOutputForStrategy(STRATEGY_SONIFICATION_RESPECTFUL); |
| updateDevicesAndOutputs(); |
| break; |
| default: |
| break; |
| } |
| } |
| |
| audio_devices_t AudioPolicyManagerBase::getDeviceForStrategy(routing_strategy strategy, |
| bool fromCache) |
| { |
| uint32_t device = AUDIO_DEVICE_NONE; |
| |
| if (fromCache) { |
| ALOGVV("getDeviceForStrategy() from cache strategy %d, device %x", |
| strategy, mDeviceForStrategy[strategy]); |
| return mDeviceForStrategy[strategy]; |
| } |
| |
| switch (strategy) { |
| |
| case STRATEGY_SONIFICATION_RESPECTFUL: |
| if (isInCall()) { |
| device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/); |
| } else if (isStreamActiveRemotely(AudioSystem::MUSIC, |
| SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY)) { |
| // while media is playing on a remote device, use the the sonification behavior. |
| // Note that we test this usecase before testing if media is playing because |
| // the isStreamActive() method only informs about the activity of a stream, not |
| // if it's for local playback. Note also that we use the same delay between both tests |
| device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/); |
| } else if (isStreamActive(AudioSystem::MUSIC, SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY)) { |
| // while media is playing (or has recently played), use the same device |
| device = getDeviceForStrategy(STRATEGY_MEDIA, false /*fromCache*/); |
| } else { |
| // when media is not playing anymore, fall back on the sonification behavior |
| device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/); |
| } |
| |
| break; |
| |
| case STRATEGY_DTMF: |
| if (!isInCall()) { |
| // when off call, DTMF strategy follows the same rules as MEDIA strategy |
| device = getDeviceForStrategy(STRATEGY_MEDIA, false /*fromCache*/); |
| break; |
| } |
| // when in call, DTMF and PHONE strategies follow the same rules |
| // FALL THROUGH |
| |
| case STRATEGY_PHONE: |
| // 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 & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT; |
| if (device) break; |
| } |
| device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET; |
| if (device) break; |
| device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_SCO; |
| if (device) break; |
| // if SCO device is requested but no SCO device is available, fall back to default case |
| // FALL THROUGH |
| |
| default: // FORCE_NONE |
| // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to A2DP |
| if (mHasA2dp && !isInCall() && |
| (mForceUse[AudioSystem::FOR_MEDIA] != AudioSystem::FORCE_NO_BT_A2DP) && |
| (getA2dpOutput() != 0) && !mA2dpSuspended) { |
| device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP; |
| if (device) break; |
| device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES; |
| if (device) break; |
| } |
| device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_WIRED_HEADPHONE; |
| if (device) break; |
| device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_WIRED_HEADSET; |
| if (device) break; |
| if (mPhoneState != AudioSystem::MODE_IN_CALL) { |
| device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_ACCESSORY; |
| if (device) break; |
| device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_DEVICE; |
| if (device) break; |
| device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET; |
| if (device) break; |
| device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_AUX_DIGITAL; |
| if (device) break; |
| device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET; |
| if (device) break; |
| } |
| device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_EARPIECE; |
| if (device) break; |
| device = mDefaultOutputDevice; |
| if (device == AUDIO_DEVICE_NONE) { |
| ALOGE("getDeviceForStrategy() no device found for STRATEGY_PHONE"); |
| } |
| break; |
| |
| case AudioSystem::FORCE_SPEAKER: |
| // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to |
| // A2DP speaker when forcing to speaker output |
| if (mHasA2dp && !isInCall() && |
| (mForceUse[AudioSystem::FOR_MEDIA] != AudioSystem::FORCE_NO_BT_A2DP) && |
| (getA2dpOutput() != 0) && !mA2dpSuspended) { |
| device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER; |
| if (device) break; |
| } |
| if (mPhoneState != AudioSystem::MODE_IN_CALL) { |
| device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_ACCESSORY; |
| if (device) break; |
| device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_DEVICE; |
| if (device) break; |
| device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET; |
| if (device) break; |
| device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_AUX_DIGITAL; |
| if (device) break; |
| device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET; |
| if (device) break; |
| } |
| device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_SPEAKER; |
| if (device) break; |
| device = mDefaultOutputDevice; |
| if (device == AUDIO_DEVICE_NONE) { |
| ALOGE("getDeviceForStrategy() no device found for STRATEGY_PHONE, FORCE_SPEAKER"); |
| } |
| break; |
| } |
| break; |
| |
| case STRATEGY_SONIFICATION: |
| |
| // If incall, just select the STRATEGY_PHONE device: The rest of the behavior is handled by |
| // handleIncallSonification(). |
| if (isInCall()) { |
| device = getDeviceForStrategy(STRATEGY_PHONE, false /*fromCache*/); |
| break; |
| } |
| // FALL THROUGH |
| |
| case STRATEGY_ENFORCED_AUDIBLE: |
| // strategy STRATEGY_ENFORCED_AUDIBLE uses same routing policy as STRATEGY_SONIFICATION |
| // except: |
| // - when in call where it doesn't default to STRATEGY_PHONE behavior |
| // - in countries where not enforced in which case it follows STRATEGY_MEDIA |
| |
| if ((strategy == STRATEGY_SONIFICATION) || |
| (mForceUse[AudioSystem::FOR_SYSTEM] == AudioSystem::FORCE_SYSTEM_ENFORCED)) { |
| device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_SPEAKER; |
| if (device == AUDIO_DEVICE_NONE) { |
| ALOGE("getDeviceForStrategy() speaker device not found for STRATEGY_SONIFICATION"); |
| } |
| } |
| // The second device used for sonification is the same as the device used by media strategy |
| // FALL THROUGH |
| |
| case STRATEGY_MEDIA: { |
| uint32_t device2 = AUDIO_DEVICE_NONE; |
| if (strategy != STRATEGY_SONIFICATION) { |
| // no sonification on remote submix (e.g. WFD) |
| device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_REMOTE_SUBMIX; |
| } |
| if ((device2 == AUDIO_DEVICE_NONE) && |
| mHasA2dp && (mForceUse[AudioSystem::FOR_MEDIA] != AudioSystem::FORCE_NO_BT_A2DP) && |
| (getA2dpOutput() != 0) && !mA2dpSuspended) { |
| device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP; |
| if (device2 == AUDIO_DEVICE_NONE) { |
| device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES; |
| } |
| if (device2 == AUDIO_DEVICE_NONE) { |
| device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER; |
| } |
| } |
| if (device2 == AUDIO_DEVICE_NONE) { |
| device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_WIRED_HEADPHONE; |
| } |
| if (device2 == AUDIO_DEVICE_NONE) { |
| device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_WIRED_HEADSET; |
| } |
| if (device2 == AUDIO_DEVICE_NONE) { |
| device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_ACCESSORY; |
| } |
| if (device2 == AUDIO_DEVICE_NONE) { |
| device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_DEVICE; |
| } |
| if (device2 == AUDIO_DEVICE_NONE) { |
| device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET; |
| } |
| if ((device2 == AUDIO_DEVICE_NONE) && (strategy != STRATEGY_SONIFICATION)) { |
| // no sonification on aux digital (e.g. HDMI) |
| device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_AUX_DIGITAL; |
| } |
| if ((device2 == AUDIO_DEVICE_NONE) && |
| (mForceUse[AudioSystem::FOR_DOCK] == AudioSystem::FORCE_ANALOG_DOCK)) { |
| device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET; |
| } |
| if (device2 == AUDIO_DEVICE_NONE) { |
| device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_SPEAKER; |
| } |
| |
| // device is DEVICE_OUT_SPEAKER if we come from case STRATEGY_SONIFICATION or |
| // STRATEGY_ENFORCED_AUDIBLE, AUDIO_DEVICE_NONE otherwise |
| device |= device2; |
| if (device) break; |
| device = mDefaultOutputDevice; |
| if (device == AUDIO_DEVICE_NONE) { |
| ALOGE("getDeviceForStrategy() no device found for STRATEGY_MEDIA"); |
| } |
| } break; |
| |
| default: |
| ALOGW("getDeviceForStrategy() unknown strategy: %d", strategy); |
| break; |
| } |
| |
| ALOGVV("getDeviceForStrategy() strategy %d, device %x", strategy, device); |
| return device; |
| } |
| |
| void AudioPolicyManagerBase::updateDevicesAndOutputs() |
| { |
| for (int i = 0; i < NUM_STRATEGIES; i++) { |
| mDeviceForStrategy[i] = getDeviceForStrategy((routing_strategy)i, false /*fromCache*/); |
| } |
| mPreviousOutputs = mOutputs; |
| } |
| |
| uint32_t AudioPolicyManagerBase::checkDeviceMuteStrategies(AudioOutputDescriptor *outputDesc, |
| audio_devices_t prevDevice, |
| uint32_t delayMs) |
| { |
| // mute/unmute strategies using an incompatible device combination |
| // if muting, wait for the audio in pcm buffer to be drained before proceeding |
| // if unmuting, unmute only after the specified delay |
| if (outputDesc->isDuplicated()) { |
| return 0; |
| } |
| |
| uint32_t muteWaitMs = 0; |
| audio_devices_t device = outputDesc->device(); |
| bool shouldMute = outputDesc->isActive() && (AudioSystem::popCount(device) >= 2); |
| // temporary mute output if device selection changes to avoid volume bursts due to |
| // different per device volumes |
| bool tempMute = outputDesc->isActive() && (device != prevDevice); |
| |
| for (size_t i = 0; i < NUM_STRATEGIES; i++) { |
| audio_devices_t curDevice = getDeviceForStrategy((routing_strategy)i, false /*fromCache*/); |
| bool mute = shouldMute && (curDevice & device) && (curDevice != device); |
| bool doMute = false; |
| |
| if (mute && !outputDesc->mStrategyMutedByDevice[i]) { |
| doMute = true; |
| outputDesc->mStrategyMutedByDevice[i] = true; |
| } else if (!mute && outputDesc->mStrategyMutedByDevice[i]){ |
| doMute = true; |
| outputDesc->mStrategyMutedByDevice[i] = false; |
| } |
| if (doMute || tempMute) { |
| for (size_t j = 0; j < mOutputs.size(); j++) { |
| AudioOutputDescriptor *desc = mOutputs.valueAt(j); |
| // skip output if it does not share any device with current output |
| if ((desc->supportedDevices() & outputDesc->supportedDevices()) |
| == AUDIO_DEVICE_NONE) { |
| continue; |
| } |
| audio_io_handle_t curOutput = mOutputs.keyAt(j); |
| ALOGVV("checkDeviceMuteStrategies() %s strategy %d (curDevice %04x) on output %d", |
| mute ? "muting" : "unmuting", i, curDevice, curOutput); |
| setStrategyMute((routing_strategy)i, mute, curOutput, mute ? 0 : delayMs); |
| if (desc->isStrategyActive((routing_strategy)i)) { |
| // do tempMute only for current output |
| if (tempMute && (desc == outputDesc)) { |
| setStrategyMute((routing_strategy)i, true, curOutput); |
| setStrategyMute((routing_strategy)i, false, curOutput, |
| desc->latency() * 2, device); |
| } |
| if ((tempMute && (desc == outputDesc)) || mute) { |
| if (muteWaitMs < desc->latency()) { |
| muteWaitMs = desc->latency(); |
| } |
| } |
| } |
| } |
| } |
| } |
| |
| // FIXME: should not need to double latency if volume could be applied immediately by the |
| // audioflinger mixer. We must account for the delay between now and the next time |
| // the audioflinger thread for this output will process a buffer (which corresponds to |
| // one buffer size, usually 1/2 or 1/4 of the latency). |
| muteWaitMs *= 2; |
| // wait for the PCM output buffers to empty before proceeding with the rest of the command |
| if (muteWaitMs > delayMs) { |
| muteWaitMs -= delayMs; |
| usleep(muteWaitMs * 1000); |
| return muteWaitMs; |
| } |
| return 0; |
| } |
| |
| uint32_t AudioPolicyManagerBase::setOutputDevice(audio_io_handle_t output, |
| audio_devices_t device, |
| bool force, |
| int delayMs) |
| { |
| ALOGV("setOutputDevice() output %d device %04x delayMs %d", output, device, delayMs); |
| AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output); |
| AudioParameter param; |
| uint32_t muteWaitMs; |
| |
| if (outputDesc->isDuplicated()) { |
| muteWaitMs = setOutputDevice(outputDesc->mOutput1->mId, device, force, delayMs); |
| muteWaitMs += setOutputDevice(outputDesc->mOutput2->mId, device, force, delayMs); |
| return muteWaitMs; |
| } |
| // no need to proceed if new device is not AUDIO_DEVICE_NONE and not supported by current |
| // output profile |
| if ((device != AUDIO_DEVICE_NONE) && |
| ((device & outputDesc->mProfile->mSupportedDevices) == 0)) { |
| return 0; |
| } |
| |
| // filter devices according to output selected |
| device = (audio_devices_t)(device & outputDesc->mProfile->mSupportedDevices); |
| |
| audio_devices_t prevDevice = outputDesc->mDevice; |
| |
| ALOGV("setOutputDevice() prevDevice %04x", prevDevice); |
| |
| if (device != AUDIO_DEVICE_NONE) { |
| outputDesc->mDevice = device; |
| |
| // Force routing if previously asked for this output |
| if (outputDesc->mForceRouting) { |
| ALOGV("Force routing to current device as previous device was null for this output"); |
| force = true; |
| |
| // Request consumed. Reset mForceRouting to false |
| outputDesc->mForceRouting = false; |
| } |
| } |
| else { |
| // Device is null and does not reflect the routing. Save the necessity to force |
| // re-routing upon next attempt to select a non-null device for this output |
| outputDesc->mForceRouting = true; |
| } |
| |
| muteWaitMs = checkDeviceMuteStrategies(outputDesc, prevDevice, delayMs); |
| |
| // Do not change the routing if: |
| // - the requested device is AUDIO_DEVICE_NONE |
| // - 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 == AUDIO_DEVICE_NONE || device == prevDevice) && !force) { |
| ALOGV("setOutputDevice() setting same device %04x or null device for output %d", device, output); |
| return muteWaitMs; |
| } |
| |
| ALOGV("setOutputDevice() changing device"); |
| // do the routing |
| param.addInt(String8(AudioParameter::keyRouting), (int)device); |
| mpClientInterface->setParameters(output, param.toString(), delayMs); |
| |
| // update stream volumes according to new device |
| applyStreamVolumes(output, device, delayMs); |
| |
| return muteWaitMs; |
| } |
| |
| AudioPolicyManagerBase::IOProfile *AudioPolicyManagerBase::getInputProfile(audio_devices_t device, |
| uint32_t samplingRate, |
| audio_format_t format, |
| audio_channel_mask_t channelMask) |
| { |
| // Choose an input profile based on the requested capture parameters: select the first available |
| // profile supporting all requested parameters. |
| for (size_t i = 0; i < mHwModules.size(); i++) |
| { |
| if (mHwModules[i]->mHandle == 0) { |
| continue; |
| } |
| for (size_t j = 0; j < mHwModules[i]->mInputProfiles.size(); j++) |
| { |
| IOProfile *profile = mHwModules[i]->mInputProfiles[j]; |
| // profile->log(); |
| if (profile->isCompatibleProfile(device, samplingRate, format, |
| channelMask, AUDIO_OUTPUT_FLAG_NONE)) { |
| return profile; |
| } |
| } |
| } |
| return NULL; |
| } |
| |
| audio_devices_t AudioPolicyManagerBase::getDeviceForInputSource(int inputSource) |
| { |
| uint32_t device = AUDIO_DEVICE_NONE; |
| |
| switch (inputSource) { |
| case AUDIO_SOURCE_VOICE_UPLINK: |
| if (mAvailableInputDevices & AUDIO_DEVICE_IN_VOICE_CALL) { |
| device = AUDIO_DEVICE_IN_VOICE_CALL; |
| break; |
| } |
| // FALL THROUGH |
| |
| case AUDIO_SOURCE_DEFAULT: |
| case AUDIO_SOURCE_MIC: |
| if (mAvailableInputDevices & AUDIO_DEVICE_IN_BLUETOOTH_A2DP) { |
| device = AUDIO_DEVICE_IN_BLUETOOTH_A2DP; |
| break; |
| } |
| // FALL THROUGH |
| |
| case AUDIO_SOURCE_VOICE_RECOGNITION: |
| case AUDIO_SOURCE_HOTWORD: |
| case AUDIO_SOURCE_VOICE_COMMUNICATION: |
| if (mForceUse[AudioSystem::FOR_RECORD] == AudioSystem::FORCE_BT_SCO && |
| mAvailableInputDevices & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) { |
| device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET; |
| } else if (mAvailableInputDevices & AUDIO_DEVICE_IN_WIRED_HEADSET) { |
| device = AUDIO_DEVICE_IN_WIRED_HEADSET; |
| } else if (mAvailableInputDevices & AUDIO_DEVICE_IN_USB_DEVICE) { |
| device = AUDIO_DEVICE_IN_USB_DEVICE; |
| } else if (mAvailableInputDevices & AUDIO_DEVICE_IN_BUILTIN_MIC) { |
| device = AUDIO_DEVICE_IN_BUILTIN_MIC; |
| } |
| break; |
| case AUDIO_SOURCE_CAMCORDER: |
| if (mAvailableInputDevices & AUDIO_DEVICE_IN_BACK_MIC) { |
| device = AUDIO_DEVICE_IN_BACK_MIC; |
| } else if (mAvailableInputDevices & AUDIO_DEVICE_IN_BUILTIN_MIC) { |
| device = AUDIO_DEVICE_IN_BUILTIN_MIC; |
| } |
| break; |
| case AUDIO_SOURCE_VOICE_DOWNLINK: |
| case AUDIO_SOURCE_VOICE_CALL: |
| if (mAvailableInputDevices & AUDIO_DEVICE_IN_VOICE_CALL) { |
| device = AUDIO_DEVICE_IN_VOICE_CALL; |
| } |
| break; |
| case AUDIO_SOURCE_REMOTE_SUBMIX: |
| if (mAvailableInputDevices & AUDIO_DEVICE_IN_REMOTE_SUBMIX) { |
| device = AUDIO_DEVICE_IN_REMOTE_SUBMIX; |
| } |
| break; |
| default: |
| ALOGW("getDeviceForInputSource() invalid input source %d", inputSource); |
| break; |
| } |
| ALOGV("getDeviceForInputSource()input source %d, device %08x", inputSource, device); |
| return device; |
| } |
| |
| bool AudioPolicyManagerBase::isVirtualInputDevice(audio_devices_t device) |
| { |
| if ((device & AUDIO_DEVICE_BIT_IN) != 0) { |
| device &= ~AUDIO_DEVICE_BIT_IN; |
| if ((popcount(device) == 1) && ((device & ~APM_AUDIO_IN_DEVICE_VIRTUAL_ALL) == 0)) |
| return true; |
| } |
| return false; |
| } |
| |
| audio_io_handle_t AudioPolicyManagerBase::getActiveInput(bool ignoreVirtualInputs) |
| { |
| for (size_t i = 0; i < mInputs.size(); i++) { |
| const AudioInputDescriptor * input_descriptor = mInputs.valueAt(i); |
| if ((input_descriptor->mRefCount > 0) |
| && (!ignoreVirtualInputs || !isVirtualInputDevice(input_descriptor->mDevice))) { |
| return mInputs.keyAt(i); |
| } |
| } |
| return 0; |
| } |
| |
| |
| audio_devices_t AudioPolicyManagerBase::getDeviceForVolume(audio_devices_t device) |
| { |
| if (device == AUDIO_DEVICE_NONE) { |
| // this happens when forcing a route update and no track is active on an output. |
| // In this case the returned category is not important. |
| device = AUDIO_DEVICE_OUT_SPEAKER; |
| } else if (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) { |
| device = AUDIO_DEVICE_OUT_SPEAKER; |
| } else { |
| device = (audio_devices_t)(device & AUDIO_DEVICE_OUT_ALL_A2DP); |
| } |
| } |
| |
| ALOGW_IF(AudioSystem::popCount(device) != 1, |
| "getDeviceForVolume() invalid device combination: %08x", |
| device); |
| |
| return device; |
| } |
| |
| AudioPolicyManagerBase::device_category AudioPolicyManagerBase::getDeviceCategory(audio_devices_t device) |
| { |
| switch(getDeviceForVolume(device)) { |
| case AUDIO_DEVICE_OUT_EARPIECE: |
| return DEVICE_CATEGORY_EARPIECE; |
| case AUDIO_DEVICE_OUT_WIRED_HEADSET: |
| case AUDIO_DEVICE_OUT_WIRED_HEADPHONE: |
| case AUDIO_DEVICE_OUT_BLUETOOTH_SCO: |
| case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET: |
| case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP: |
| case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES: |
| return DEVICE_CATEGORY_HEADSET; |
| case AUDIO_DEVICE_OUT_SPEAKER: |
| case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT: |
| case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER: |
| case AUDIO_DEVICE_OUT_AUX_DIGITAL: |
| case AUDIO_DEVICE_OUT_USB_ACCESSORY: |
| case AUDIO_DEVICE_OUT_USB_DEVICE: |
| case AUDIO_DEVICE_OUT_REMOTE_SUBMIX: |
| default: |
| return DEVICE_CATEGORY_SPEAKER; |
| } |
| } |
| |
| float AudioPolicyManagerBase::volIndexToAmpl(audio_devices_t device, const StreamDescriptor& streamDesc, |
| int indexInUi) |
| { |
| device_category deviceCategory = getDeviceCategory(device); |
| const VolumeCurvePoint *curve = streamDesc.mVolumeCurve[deviceCategory]; |
| |
| // the volume index in the UI is relative to the min and max volume indices for this stream type |
| int nbSteps = 1 + curve[VOLMAX].mIndex - |
| curve[VOLMIN].mIndex; |
| int volIdx = (nbSteps * (indexInUi - streamDesc.mIndexMin)) / |
| (streamDesc.mIndexMax - streamDesc.mIndexMin); |
| |
| // find what part of the curve this index volume belongs to, or if it's out of bounds |
| int segment = 0; |
| if (volIdx < curve[VOLMIN].mIndex) { // out of bounds |
| return 0.0f; |
| } else if (volIdx < curve[VOLKNEE1].mIndex) { |
| segment = 0; |
| } else if (volIdx < curve[VOLKNEE2].mIndex) { |
| segment = 1; |
| } else if (volIdx <= curve[VOLMAX].mIndex) { |
| segment = 2; |
| } else { // out of bounds |
| return 1.0f; |
| } |
| |
| // linear interpolation in the attenuation table in dB |
| float decibels = curve[segment].mDBAttenuation + |
| ((float)(volIdx - curve[segment].mIndex)) * |
| ( (curve[segment+1].mDBAttenuation - |
| curve[segment].mDBAttenuation) / |
| ((float)(curve[segment+1].mIndex - |
| curve[segment].mIndex)) ); |
| |
| float amplification = exp( decibels * 0.115129f); // exp( dB * ln(10) / 20 ) |
| |
| ALOGVV("VOLUME vol index=[%d %d %d], dB=[%.1f %.1f %.1f] ampl=%.5f", |
| curve[segment].mIndex, volIdx, |
| curve[segment+1].mIndex, |
| curve[segment].mDBAttenuation, |
| decibels, |
| curve[segment+1].mDBAttenuation, |
| amplification); |
| |
| return amplification; |
| } |
| |
| const 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::sSpeakerSonificationVolumeCurveDrc[AudioPolicyManagerBase::VOLCNT] = { |
| {1, -35.7f}, {33, -26.1f}, {66, -13.2f}, {100, 0.0f} |
| }; |
| |
| // AUDIO_STREAM_SYSTEM, AUDIO_STREAM_ENFORCED_AUDIBLE and AUDIO_STREAM_DTMF volume tracks |
| // AUDIO_STREAM_RING on phones and AUDIO_STREAM_MUSIC on tablets. |
| // AUDIO_STREAM_DTMF tracks AUDIO_STREAM_VOICE_CALL while in call (See AudioService.java). |
| // The range is constrained between -24dB and -6dB over speaker and -30dB and -18dB over headset. |
| |
| const AudioPolicyManagerBase::VolumeCurvePoint |
| AudioPolicyManagerBase::sDefaultSystemVolumeCurve[AudioPolicyManagerBase::VOLCNT] = { |
| {1, -24.0f}, {33, -18.0f}, {66, -12.0f}, {100, -6.0f} |
| }; |
| |
| const AudioPolicyManagerBase::VolumeCurvePoint |
| AudioPolicyManagerBase::sDefaultSystemVolumeCurveDrc[AudioPolicyManagerBase::VOLCNT] = { |
| {1, -34.0f}, {33, -24.0f}, {66, -15.0f}, {100, -6.0f} |
| }; |
| |
| const AudioPolicyManagerBase::VolumeCurvePoint |
| AudioPolicyManagerBase::sHeadsetSystemVolumeCurve[AudioPolicyManagerBase::VOLCNT] = { |
| {1, -30.0f}, {33, -26.0f}, {66, -22.0f}, {100, -18.0f} |
| }; |
| |
| const AudioPolicyManagerBase::VolumeCurvePoint |
| AudioPolicyManagerBase::sDefaultVoiceVolumeCurve[AudioPolicyManagerBase::VOLCNT] = { |
| {0, -42.0f}, {33, -28.0f}, {66, -14.0f}, {100, 0.0f} |
| }; |
| |
| const AudioPolicyManagerBase::VolumeCurvePoint |
| AudioPolicyManagerBase::sSpeakerVoiceVolumeCurve[AudioPolicyManagerBase::VOLCNT] = { |
| {0, -24.0f}, {33, -16.0f}, {66, -8.0f}, {100, 0.0f} |
| }; |
| |
| const AudioPolicyManagerBase::VolumeCurvePoint |
| *AudioPolicyManagerBase::sVolumeProfiles[AudioSystem::NUM_STREAM_TYPES] |
| [AudioPolicyManagerBase::DEVICE_CATEGORY_CNT] = { |
| { // AUDIO_STREAM_VOICE_CALL |
| sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_HEADSET |
| sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_SPEAKER |
| sDefaultVoiceVolumeCurve // DEVICE_CATEGORY_EARPIECE |
| }, |
| { // AUDIO_STREAM_SYSTEM |
| sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET |
| sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER |
| sDefaultSystemVolumeCurve // DEVICE_CATEGORY_EARPIECE |
| }, |
| { // AUDIO_STREAM_RING |
| sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET |
| sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER |
| sDefaultVolumeCurve // DEVICE_CATEGORY_EARPIECE |
| }, |
| { // AUDIO_STREAM_MUSIC |
| sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_HEADSET |
| sSpeakerMediaVolumeCurve, // DEVICE_CATEGORY_SPEAKER |
| sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EARPIECE |
| }, |
| { // AUDIO_STREAM_ALARM |
| sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET |
| sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER |
| sDefaultVolumeCurve // DEVICE_CATEGORY_EARPIECE |
| }, |
| { // AUDIO_STREAM_NOTIFICATION |
| sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET |
| sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER |
| sDefaultVolumeCurve // DEVICE_CATEGORY_EARPIECE |
| }, |
| { // AUDIO_STREAM_BLUETOOTH_SCO |
| sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_HEADSET |
| sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_SPEAKER |
| sDefaultVoiceVolumeCurve // DEVICE_CATEGORY_EARPIECE |
| }, |
| { // AUDIO_STREAM_ENFORCED_AUDIBLE |
| sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET |
| sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER |
| sDefaultSystemVolumeCurve // DEVICE_CATEGORY_EARPIECE |
| }, |
| { // AUDIO_STREAM_DTMF |
| sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET |
| sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER |
| sDefaultSystemVolumeCurve // DEVICE_CATEGORY_EARPIECE |
| }, |
| { // AUDIO_STREAM_TTS |
| sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_HEADSET |
| sSpeakerMediaVolumeCurve, // DEVICE_CATEGORY_SPEAKER |
| sDefaultMediaVolumeCurve // 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[i][j]; |
| } |
| } |
| |
| // Check availability of DRC on speaker path: if available, override some of the speaker curves |
| if (mSpeakerDrcEnabled) { |
| mStreams[AUDIO_STREAM_SYSTEM].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] = |
| sDefaultSystemVolumeCurveDrc; |
| mStreams[AUDIO_STREAM_RING].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] = |
| sSpeakerSonificationVolumeCurveDrc; |
| mStreams[AUDIO_STREAM_ALARM].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] = |
| sSpeakerSonificationVolumeCurveDrc; |
| mStreams[AUDIO_STREAM_NOTIFICATION].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] = |
| sSpeakerSonificationVolumeCurveDrc; |
| } |
| } |
| |
| float AudioPolicyManagerBase::computeVolume(int stream, |
| int index, |
| audio_io_handle_t output, |
| audio_devices_t device) |
| { |
| float volume = 1.0; |
| AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output); |
| StreamDescriptor &streamDesc = mStreams[stream]; |
| |
| if (device == AUDIO_DEVICE_NONE) { |
| 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 == AUDIO_DEVICE_OUT_AUX_DIGITAL || |
| device == AUDIO_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. |
| const routing_strategy stream_strategy = getStrategy((AudioSystem::stream_type)stream); |
| if ((device & (AUDIO_DEVICE_OUT_BLUETOOTH_A2DP | |
| AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES | |
| AUDIO_DEVICE_OUT_WIRED_HEADSET | |
| AUDIO_DEVICE_OUT_WIRED_HEADPHONE)) && |
| ((stream_strategy == STRATEGY_SONIFICATION) |
| || (stream_strategy == STRATEGY_SONIFICATION_RESPECTFUL) |
| || (stream == AudioSystem::SYSTEM) |
| || ((stream_strategy == STRATEGY_ENFORCED_AUDIBLE) && |
| (mForceUse[AudioSystem::FOR_SYSTEM] == AudioSystem::FORCE_NONE))) && |
| streamDesc.mCanBeMuted) { |
| volume *= SONIFICATION_HEADSET_VOLUME_FACTOR; |
| // when the phone is ringing we must consider that music could have been paused just before |
| // by the music application and behave as if music was active if the last music track was |
| // just stopped |
| if (isStreamActive(AudioSystem::MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY) || |
| mLimitRingtoneVolume) { |
| audio_devices_t musicDevice = getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/); |
| float musicVol = computeVolume(AudioSystem::MUSIC, |
| mStreams[AudioSystem::MUSIC].getVolumeIndex(musicDevice), |
| output, |
| musicDevice); |
| float minVol = (musicVol > SONIFICATION_HEADSET_VOLUME_MIN) ? |
| musicVol : SONIFICATION_HEADSET_VOLUME_MIN; |
| if (volume > minVol) { |
| volume = minVol; |
| ALOGV("computeVolume limiting volume to %f musicVol %f", minVol, musicVol); |
| } |
| } |
| } |
| |
| return volume; |
| } |
| |
| status_t AudioPolicyManagerBase::checkAndSetVolume(int stream, |
| int index, |
| audio_io_handle_t output, |
| audio_devices_t device, |
| int delayMs, |
| bool force) |
| { |
| |
| // do not change actual stream volume if the stream is muted |
| if (mOutputs.valueFor(output)->mMuteCount[stream] != 0) { |
| ALOGVV("checkAndSetVolume() stream %d muted count %d", |
| stream, mOutputs.valueFor(output)->mMuteCount[stream]); |
| return NO_ERROR; |
| } |
| |
| // do not change in call volume if bluetooth is connected and vice versa |
| if ((stream == AudioSystem::VOICE_CALL && mForceUse[AudioSystem::FOR_COMMUNICATION] == AudioSystem::FORCE_BT_SCO) || |
| (stream == AudioSystem::BLUETOOTH_SCO && mForceUse[AudioSystem::FOR_COMMUNICATION] != AudioSystem::FORCE_BT_SCO)) { |
| ALOGV("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; |
| ALOGVV("checkAndSetVolume() for output %d stream %d, volume %f, delay %d", output, stream, volume, delayMs); |
| // Force VOICE_CALL to track BLUETOOTH_SCO stream volume when bluetooth audio is |
| // enabled |
| if (stream == 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 == mPrimaryOutput) { |
| mpClientInterface->setVoiceVolume(voiceVolume, delayMs); |
| mLastVoiceVolume = voiceVolume; |
| } |
| } |
| |
| return NO_ERROR; |
| } |
| |
| void AudioPolicyManagerBase::applyStreamVolumes(audio_io_handle_t output, |
| audio_devices_t device, |
| int delayMs, |
| bool force) |
| { |
| ALOGVV("applyStreamVolumes() for output %d and device %x", output, device); |
| |
| for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) { |
| checkAndSetVolume(stream, |
| mStreams[stream].getVolumeIndex(device), |
| output, |
| device, |
| delayMs, |
| force); |
| } |
| } |
| |
| void AudioPolicyManagerBase::setStrategyMute(routing_strategy strategy, |
| bool on, |
| audio_io_handle_t output, |
| int delayMs, |
| audio_devices_t device) |
| { |
| ALOGVV("setStrategyMute() strategy %d, mute %d, output %d", strategy, on, output); |
| for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) { |
| if (getStrategy((AudioSystem::stream_type)stream) == strategy) { |
| setStreamMute(stream, on, output, delayMs, device); |
| } |
| } |
| } |
| |
| void AudioPolicyManagerBase::setStreamMute(int stream, |
| bool on, |
| audio_io_handle_t output, |
| int delayMs, |
| audio_devices_t device) |
| { |
| StreamDescriptor &streamDesc = mStreams[stream]; |
| AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output); |
| if (device == AUDIO_DEVICE_NONE) { |
| device = outputDesc->device(); |
| } |
| |
| ALOGVV("setStreamMute() stream %d, mute %d, output %d, mMuteCount %d device %04x", |
| stream, on, output, outputDesc->mMuteCount[stream], device); |
| |
| if (on) { |
| if (outputDesc->mMuteCount[stream] == 0) { |
| if (streamDesc.mCanBeMuted && |
| ((stream != AudioSystem::ENFORCED_AUDIBLE) || |
| (mForceUse[AudioSystem::FOR_SYSTEM] == AudioSystem::FORCE_NONE))) { |
| checkAndSetVolume(stream, 0, output, device, delayMs); |
| } |
| } |
| // increment mMuteCount after calling checkAndSetVolume() so that volume change is not ignored |
| outputDesc->mMuteCount[stream]++; |
| } else { |
| if (outputDesc->mMuteCount[stream] == 0) { |
| ALOGV("setStreamMute() unmuting non muted stream!"); |
| return; |
| } |
| if (--outputDesc->mMuteCount[stream] == 0) { |
| checkAndSetVolume(stream, |
| streamDesc.getVolumeIndex(device), |
| output, |
| device, |
| delayMs); |
| } |
| } |
| } |
| |
| void 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 |
| const routing_strategy stream_strategy = getStrategy((AudioSystem::stream_type)stream); |
| if ((stream_strategy == STRATEGY_SONIFICATION) || |
| ((stream_strategy == STRATEGY_SONIFICATION_RESPECTFUL))) { |
| AudioOutputDescriptor *outputDesc = mOutputs.valueFor(mPrimaryOutput); |
| ALOGV("handleIncallSonification() stream %d starting %d device %x stateChange %d", |
| stream, starting, outputDesc->mDevice, stateChange); |
| if (outputDesc->mRefCount[stream]) { |
| int muteCount = 1; |
| if (stateChange) { |
| muteCount = outputDesc->mRefCount[stream]; |
| } |
| if (AudioSystem::isLowVisibility((AudioSystem::stream_type)stream)) { |
| ALOGV("handleIncallSonification() low visibility, muteCount %d", muteCount); |
| for (int i = 0; i < muteCount; i++) { |
| setStreamMute(stream, starting, mPrimaryOutput); |
| } |
| } else { |
| ALOGV("handleIncallSonification() high visibility"); |
| if (outputDesc->device() & |
| getDeviceForStrategy(STRATEGY_PHONE, true /*fromCache*/)) { |
| ALOGV("handleIncallSonification() high visibility muted, muteCount %d", muteCount); |
| for (int i = 0; i < muteCount; i++) { |
| setStreamMute(stream, starting, mPrimaryOutput); |
| } |
| } |
| if (starting) { |
| mpClientInterface->startTone(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)); |
| } |
| |
| uint32_t AudioPolicyManagerBase::getMaxEffectsCpuLoad() |
| { |
| return MAX_EFFECTS_CPU_LOAD; |
| } |
| |
| uint32_t AudioPolicyManagerBase::getMaxEffectsMemory() |
| { |
| return MAX_EFFECTS_MEMORY; |
| } |
| |
| // --- AudioOutputDescriptor class implementation |
| |
| AudioPolicyManagerBase::AudioOutputDescriptor::AudioOutputDescriptor( |
| const IOProfile *profile) |
| : mId(0), mSamplingRate(0), mFormat(AUDIO_FORMAT_DEFAULT), |
| mChannelMask(0), mLatency(0), |
| mFlags((audio_output_flags_t)0), mDevice(AUDIO_DEVICE_NONE), |
| mOutput1(0), mOutput2(0), mProfile(profile), mDirectOpenCount(0), |
| mForceRouting(false) |
| { |
| // 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; |
| } |
| for (int i = 0; i < NUM_STRATEGIES; i++) { |
| mStrategyMutedByDevice[i] = false; |
| } |
| if (profile != NULL) { |
| mSamplingRate = profile->mSamplingRates[0]; |
| mFormat = profile->mFormats[0]; |
| mChannelMask = profile->mChannelMasks[0]; |
| mFlags = profile->mFlags; |
| } |
| } |
| |
| audio_devices_t AudioPolicyManagerBase::AudioOutputDescriptor::device() const |
| { |
| if (isDuplicated()) { |
| return (audio_devices_t)(mOutput1->mDevice | mOutput2->mDevice); |
| } else { |
| return mDevice; |
| } |
| } |
| |
| uint32_t AudioPolicyManagerBase::AudioOutputDescriptor::latency() |
| { |
| if (isDuplicated()) { |
| return (mOutput1->mLatency > mOutput2->mLatency) ? mOutput1->mLatency : mOutput2->mLatency; |
| } else { |
| return mLatency; |
| } |
| } |
| |
| bool AudioPolicyManagerBase::AudioOutputDescriptor::sharesHwModuleWith( |
| const AudioOutputDescriptor *outputDesc) |
| { |
| if (isDuplicated()) { |
| return mOutput1->sharesHwModuleWith(outputDesc) || mOutput2->sharesHwModuleWith(outputDesc); |
| } else if (outputDesc->isDuplicated()){ |
| return sharesHwModuleWith(outputDesc->mOutput1) || sharesHwModuleWith(outputDesc->mOutput2); |
| } else { |
| return (mProfile->mModule == outputDesc->mProfile->mModule); |
| } |
| } |
| |
| void 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) { |
| ALOGW("changeRefCount() invalid delta %d for stream %d, refCount %d", delta, stream, mRefCount[stream]); |
| mRefCount[stream] = 0; |
| return; |
| } |
| mRefCount[stream] += delta; |
| ALOGV("changeRefCount() stream %d, count %d", stream, mRefCount[stream]); |
| } |
| |
| audio_devices_t AudioPolicyManagerBase::AudioOutputDescriptor::supportedDevices() |
| { |
| if (isDuplicated()) { |
| return (audio_devices_t)(mOutput1->supportedDevices() | mOutput2->supportedDevices()); |
| } else { |
| return mProfile->mSupportedDevices ; |
| } |
| } |
| |
| bool AudioPolicyManagerBase::AudioOutputDescriptor::isActive(uint32_t inPastMs) const |
| { |
| return isStrategyActive(NUM_STRATEGIES, inPastMs); |
| } |
| |
| bool AudioPolicyManagerBase::AudioOutputDescriptor::isStrategyActive(routing_strategy strategy, |
| uint32_t inPastMs, |
| nsecs_t sysTime) const |
| { |
| if ((sysTime == 0) && (inPastMs != 0)) { |
| sysTime = systemTime(); |
| } |
| for (int i = 0; i < AudioSystem::NUM_STREAM_TYPES; i++) { |
| if (((getStrategy((AudioSystem::stream_type)i) == strategy) || |
| (NUM_STRATEGIES == strategy)) && |
| isStreamActive((AudioSystem::stream_type)i, inPastMs, sysTime)) { |
| return true; |
| } |
| } |
| return false; |
| } |
| |
| bool AudioPolicyManagerBase::AudioOutputDescriptor::isStreamActive(AudioSystem::stream_type stream, |
| uint32_t inPastMs, |
| nsecs_t sysTime) const |
| { |
| if (mRefCount[stream] != 0) { |
| return true; |
| } |
| if (inPastMs == 0) { |
| return false; |
| } |
| if (sysTime == 0) { |
| sysTime = systemTime(); |
| } |
| if (ns2ms(sysTime - mStopTime[stream]) < inPastMs) { |
| return true; |
| } |
| return false; |
| } |
| |
| |
| 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: %08x\n", mFormat); |
| result.append(buffer); |
| snprintf(buffer, SIZE, " Channels: %08x\n", mChannelMask); |
| result.append(buffer); |
| snprintf(buffer, SIZE, " Latency: %d\n", mLatency); |
| result.append(buffer); |
| snprintf(buffer, SIZE, " Flags %08x\n", mFlags); |
| result.append(buffer); |
| snprintf(buffer, SIZE, " Devices %08x\n", device()); |
| result.append(buffer); |
| snprintf(buffer, SIZE, " Stream volume refCount muteCount\n"); |
| result.append(buffer); |
| for (int i = 0; i < 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(const IOProfile *profile) |
| : mId(0), mSamplingRate(0), mFormat(AUDIO_FORMAT_DEFAULT), mChannelMask(0), |
| mDevice(AUDIO_DEVICE_NONE), mRefCount(0), |
| mInputSource(0), mProfile(profile) |
| { |
| if (profile != NULL) { |
| mSamplingRate = profile->mSamplingRates[0]; |
| mFormat = profile->mFormats[0]; |
| mChannelMask = profile->mChannelMasks[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", mChannelMask); |
| result.append(buffer); |
| snprintf(buffer, SIZE, " Devices %08x\n", mDevice); |
| result.append(buffer); |
| snprintf(buffer, SIZE, " Ref Count %d\n", mRefCount); |
| result.append(buffer); |
| write(fd, result.string(), result.size()); |
| |
| return NO_ERROR; |
| } |
| |
| // --- StreamDescriptor class implementation |
| |
| AudioPolicyManagerBase::StreamDescriptor::StreamDescriptor() |
| : mIndexMin(0), mIndexMax(1), mCanBeMuted(true) |
| { |
| mIndexCur.add(AUDIO_DEVICE_OUT_DEFAULT, 0); |
| } |
| |
| int AudioPolicyManagerBase::StreamDescriptor::getVolumeIndex(audio_devices_t device) |
| { |
| device = AudioPolicyManagerBase::getDeviceForVolume(device); |
| // there is always a valid entry for AUDIO_DEVICE_OUT_DEFAULT |
| if (mIndexCur.indexOfKey(device) < 0) { |
| device = AUDIO_DEVICE_OUT_DEFAULT; |
| } |
| return mIndexCur.valueFor(device); |
| } |
| |
| void AudioPolicyManagerBase::StreamDescriptor::dump(int fd) |
| { |
| const size_t SIZE = 256; |
| char buffer[SIZE]; |
| String8 result; |
| |
| snprintf(buffer, SIZE, "%s %02d %02d ", |
| mCanBeMuted ? "true " : "false", mIndexMin, mIndexMax); |
| result.append(buffer); |
| for (size_t i = 0; i < mIndexCur.size(); i++) { |
| snprintf(buffer, SIZE, "%04x : %02d, ", |
| mIndexCur.keyAt(i), |
| mIndexCur.valueAt(i)); |
| result.append(buffer); |
| } |
| result.append("\n"); |
| |
| write(fd, result.string(), result.size()); |
| } |
| |
| // --- EffectDescriptor class implementation |
| |
| status_t 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; |
| } |
| |
| // --- IOProfile class implementation |
| |
| AudioPolicyManagerBase::HwModule::HwModule(const char *name) |
| : mName(strndup(name, AUDIO_HARDWARE_MODULE_ID_MAX_LEN)), mHandle(0) |
| { |
| } |
| |
| AudioPolicyManagerBase::HwModule::~HwModule() |
| { |
| for (size_t i = 0; i < mOutputProfiles.size(); i++) { |
| delete mOutputProfiles[i]; |
| } |
| for (size_t i = 0; i < mInputProfiles.size(); i++) { |
| delete mInputProfiles[i]; |
| } |
| free((void *)mName); |
| } |
| |
| void AudioPolicyManagerBase::HwModule::dump(int fd) |
| { |
| const size_t SIZE = 256; |
| char buffer[SIZE]; |
| String8 result; |
| |
| snprintf(buffer, SIZE, " - name: %s\n", mName); |
| result.append(buffer); |
| snprintf(buffer, SIZE, " - handle: %d\n", mHandle); |
| result.append(buffer); |
| write(fd, result.string(), result.size()); |
| if (mOutputProfiles.size()) { |
| write(fd, " - outputs:\n", strlen(" - outputs:\n")); |
| for (size_t i = 0; i < mOutputProfiles.size(); i++) { |
| snprintf(buffer, SIZE, " output %zu:\n", i); |
| write(fd, buffer, strlen(buffer)); |
| mOutputProfiles[i]->dump(fd); |
| } |
| } |
| if (mInputProfiles.size()) { |
| write(fd, " - inputs:\n", strlen(" - inputs:\n")); |
| for (size_t i = 0; i < mInputProfiles.size(); i++) { |
| snprintf(buffer, SIZE, " input %zu:\n", i); |
| write(fd, buffer, strlen(buffer)); |
| mInputProfiles[i]->dump(fd); |
| } |
| } |
| } |
| |
| AudioPolicyManagerBase::IOProfile::IOProfile(HwModule *module) |
| : mFlags((audio_output_flags_t)0), mModule(module) |
| { |
| } |
| |
| AudioPolicyManagerBase::IOProfile::~IOProfile() |
| { |
| } |
| |
| // checks if the IO profile is compatible with specified parameters. |
| // Sampling rate, format and channel mask must be specified in order to |
| // get a valid a match |
| bool AudioPolicyManagerBase::IOProfile::isCompatibleProfile(audio_devices_t device, |
| uint32_t samplingRate, |
| audio_format_t format, |
| audio_channel_mask_t channelMask, |
| audio_output_flags_t flags) const |
| { |
| if (samplingRate == 0 || !audio_is_valid_format(format) || channelMask == 0) { |
| return false; |
| } |
| |
| if ((mSupportedDevices & device) != device) { |
| return false; |
| } |
| if ((mFlags & flags) != flags) { |
| return false; |
| } |
| size_t i; |
| for (i = 0; i < mSamplingRates.size(); i++) |
| { |
| if (mSamplingRates[i] == samplingRate) { |
| break; |
| } |
| } |
| if (i == mSamplingRates.size()) { |
| return false; |
| } |
| for (i = 0; i < mFormats.size(); i++) |
| { |
| if (mFormats[i] == format) { |
| break; |
| } |
| } |
| if (i == mFormats.size()) { |
| return false; |
| } |
| for (i = 0; i < mChannelMasks.size(); i++) |
| { |
| if (mChannelMasks[i] == channelMask) { |
| break; |
| } |
| } |
| if (i == mChannelMasks.size()) { |
| return false; |
| } |
| return true; |
| } |
| |
| void AudioPolicyManagerBase::IOProfile::dump(int fd) |
| { |
| const size_t SIZE = 256; |
| char buffer[SIZE]; |
| String8 result; |
| |
| snprintf(buffer, SIZE, " - sampling rates: "); |
| result.append(buffer); |
| for (size_t i = 0; i < mSamplingRates.size(); i++) { |
| snprintf(buffer, SIZE, "%d", mSamplingRates[i]); |
| result.append(buffer); |
| result.append(i == (mSamplingRates.size() - 1) ? "\n" : ", "); |
| } |
| |
| snprintf(buffer, SIZE, " - channel masks: "); |
| result.append(buffer); |
| for (size_t i = 0; i < mChannelMasks.size(); i++) { |
| snprintf(buffer, SIZE, "0x%04x", mChannelMasks[i]); |
| result.append(buffer); |
| result.append(i == (mChannelMasks.size() - 1) ? "\n" : ", "); |
| } |
| |
| snprintf(buffer, SIZE, " - formats: "); |
| result.append(buffer); |
| for (size_t i = 0; i < mFormats.size(); i++) { |
| snprintf(buffer, SIZE, "0x%08x", mFormats[i]); |
| result.append(buffer); |
| result.append(i == (mFormats.size() - 1) ? "\n" : ", "); |
| } |
| |
| snprintf(buffer, SIZE, " - devices: 0x%04x\n", mSupportedDevices); |
| result.append(buffer); |
| snprintf(buffer, SIZE, " - flags: 0x%04x\n", mFlags); |
| result.append(buffer); |
| |
| write(fd, result.string(), result.size()); |
| } |
| |
| void AudioPolicyManagerBase::IOProfile::log() |
| { |
| const size_t SIZE = 256; |
| char buffer[SIZE]; |
| String8 result; |
| |
| ALOGV(" - sampling rates: "); |
| for (size_t i = 0; i < mSamplingRates.size(); i++) { |
| ALOGV(" %d", mSamplingRates[i]); |
| } |
| |
| ALOGV(" - channel masks: "); |
| for (size_t i = 0; i < mChannelMasks.size(); i++) { |
| ALOGV(" 0x%04x", mChannelMasks[i]); |
| } |
| |
| ALOGV(" - formats: "); |
| for (size_t i = 0; i < mFormats.size(); i++) { |
| ALOGV(" 0x%08x", mFormats[i]); |
| } |
| |
| ALOGV(" - devices: 0x%04x\n", mSupportedDevices); |
| ALOGV(" - flags: 0x%04x\n", mFlags); |
| } |
| |
| // --- audio_policy.conf file parsing |
| |
| struct StringToEnum { |
| const char *name; |
| uint32_t value; |
| }; |
| |
| #define STRING_TO_ENUM(string) { #string, string } |
| #define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0])) |
| |
| const struct StringToEnum sDeviceNameToEnumTable[] = { |
| STRING_TO_ENUM(AUDIO_DEVICE_OUT_EARPIECE), |
| STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPEAKER), |
| STRING_TO_ENUM(AUDIO_DEVICE_OUT_WIRED_HEADSET), |
| STRING_TO_ENUM(AUDIO_DEVICE_OUT_WIRED_HEADPHONE), |
| STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_SCO), |
| STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_A2DP), |
| STRING_TO_ENUM(AUDIO_DEVICE_OUT_AUX_DIGITAL), |
| STRING_TO_ENUM(AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET), |
| STRING_TO_ENUM(AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET), |
| STRING_TO_ENUM(AUDIO_DEVICE_OUT_USB_DEVICE), |
| STRING_TO_ENUM(AUDIO_DEVICE_OUT_USB_ACCESSORY), |
| STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_USB), |
| STRING_TO_ENUM(AUDIO_DEVICE_OUT_REMOTE_SUBMIX), |
| STRING_TO_ENUM(AUDIO_DEVICE_IN_BUILTIN_MIC), |
| STRING_TO_ENUM(AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET), |
| STRING_TO_ENUM(AUDIO_DEVICE_IN_WIRED_HEADSET), |
| STRING_TO_ENUM(AUDIO_DEVICE_IN_AUX_DIGITAL), |
| STRING_TO_ENUM(AUDIO_DEVICE_IN_VOICE_CALL), |
| STRING_TO_ENUM(AUDIO_DEVICE_IN_BACK_MIC), |
| STRING_TO_ENUM(AUDIO_DEVICE_IN_REMOTE_SUBMIX), |
| STRING_TO_ENUM(AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET), |
| STRING_TO_ENUM(AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET), |
| STRING_TO_ENUM(AUDIO_DEVICE_IN_USB_ACCESSORY), |
| STRING_TO_ENUM(AUDIO_DEVICE_IN_USB_DEVICE), |
| STRING_TO_ENUM(AUDIO_DEVICE_IN_BLUETOOTH_A2DP), |
| }; |
| |
| const struct StringToEnum sFlagNameToEnumTable[] = { |
| STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_DIRECT), |
| STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_PRIMARY), |
| STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_FAST), |
| STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_DEEP_BUFFER), |
| STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD), |
| STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_NON_BLOCKING), |
| }; |
| |
| const struct StringToEnum sFormatNameToEnumTable[] = { |
| STRING_TO_ENUM(AUDIO_FORMAT_PCM_16_BIT), |
| STRING_TO_ENUM(AUDIO_FORMAT_PCM_8_BIT), |
| STRING_TO_ENUM(AUDIO_FORMAT_PCM_32_BIT), |
| STRING_TO_ENUM(AUDIO_FORMAT_PCM_8_24_BIT), |
| STRING_TO_ENUM(AUDIO_FORMAT_PCM_FLOAT), |
| STRING_TO_ENUM(AUDIO_FORMAT_PCM_24_BIT_PACKED), |
| STRING_TO_ENUM(AUDIO_FORMAT_MP3), |
| STRING_TO_ENUM(AUDIO_FORMAT_AAC), |
| STRING_TO_ENUM(AUDIO_FORMAT_VORBIS), |
| STRING_TO_ENUM(AUDIO_FORMAT_HE_AAC_V1), |
| STRING_TO_ENUM(AUDIO_FORMAT_HE_AAC_V2), |
| STRING_TO_ENUM(AUDIO_FORMAT_OPUS), |
| STRING_TO_ENUM(AUDIO_FORMAT_AC3), |
| STRING_TO_ENUM(AUDIO_FORMAT_E_AC3), |
| }; |
| |
| const struct StringToEnum sOutChannelsNameToEnumTable[] = { |
| STRING_TO_ENUM(AUDIO_CHANNEL_OUT_MONO), |
| STRING_TO_ENUM(AUDIO_CHANNEL_OUT_STEREO), |
| STRING_TO_ENUM(AUDIO_CHANNEL_OUT_5POINT1), |
| STRING_TO_ENUM(AUDIO_CHANNEL_OUT_7POINT1), |
| }; |
| |
| const struct StringToEnum sInChannelsNameToEnumTable[] = { |
| STRING_TO_ENUM(AUDIO_CHANNEL_IN_MONO), |
| STRING_TO_ENUM(AUDIO_CHANNEL_IN_STEREO), |
| STRING_TO_ENUM(AUDIO_CHANNEL_IN_FRONT_BACK), |
| }; |
| |
| |
| uint32_t AudioPolicyManagerBase::stringToEnum(const struct StringToEnum *table, |
| size_t size, |
| const char *name) |
| { |
| for (size_t i = 0; i < size; i++) { |
| if (strcmp(table[i].name, name) == 0) { |
| ALOGV("stringToEnum() found %s", table[i].name); |
| return table[i].value; |
| } |
| } |
| return 0; |
| } |
| |
| bool AudioPolicyManagerBase::stringToBool(const char *value) |
| { |
| return ((strcasecmp("true", value) == 0) || (strcmp("1", value) == 0)); |
| } |
| |
| audio_output_flags_t AudioPolicyManagerBase::parseFlagNames(char *name) |
| { |
| uint32_t flag = 0; |
| |
| // it is OK to cast name to non const here as we are not going to use it after |
| // strtok() modifies it |
| char *flagName = strtok(name, "|"); |
| while (flagName != NULL) { |
| if (strlen(flagName) != 0) { |
| flag |= stringToEnum(sFlagNameToEnumTable, |
| ARRAY_SIZE(sFlagNameToEnumTable), |
| flagName); |
| } |
| flagName = strtok(NULL, "|"); |
| } |
| //force direct flag if offload flag is set: offloading implies a direct output stream |
| // and all common behaviors are driven by checking only the direct flag |
| // this should normally be set appropriately in the policy configuration file |
| if ((flag & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) { |
| flag |= AUDIO_OUTPUT_FLAG_DIRECT; |
| } |
| |
| return (audio_output_flags_t)flag; |
| } |
| |
| audio_devices_t AudioPolicyManagerBase::parseDeviceNames(char *name) |
| { |
| uint32_t device = 0; |
| |
| char *devName = strtok(name, "|"); |
| while (devName != NULL) { |
| if (strlen(devName) != 0) { |
| device |= stringToEnum(sDeviceNameToEnumTable, |
| ARRAY_SIZE(sDeviceNameToEnumTable), |
| devName); |
| } |
| devName = strtok(NULL, "|"); |
| } |
| return device; |
| } |
| |
| void AudioPolicyManagerBase::loadSamplingRates(char *name, IOProfile *profile) |
| { |
| char *str = strtok(name, "|"); |
| |
| // by convention, "0' in the first entry in mSamplingRates indicates the supported sampling |
| // rates should be read from the output stream after it is opened for the first time |
| if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) { |
| profile->mSamplingRates.add(0); |
| return; |
| } |
| |
| while (str != NULL) { |
| uint32_t rate = atoi(str); |
| if (rate != 0) { |
| ALOGV("loadSamplingRates() adding rate %d", rate); |
| profile->mSamplingRates.add(rate); |
| } |
| str = strtok(NULL, "|"); |
| } |
| return; |
| } |
| |
| void AudioPolicyManagerBase::loadFormats(char *name, IOProfile *profile) |
| { |
| char *str = strtok(name, "|"); |
| |
| // by convention, "0' in the first entry in mFormats indicates the supported formats |
| // should be read from the output stream after it is opened for the first time |
| if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) { |
| profile->mFormats.add(AUDIO_FORMAT_DEFAULT); |
| return; |
| } |
| |
| while (str != NULL) { |
| audio_format_t format = (audio_format_t)stringToEnum(sFormatNameToEnumTable, |
| ARRAY_SIZE(sFormatNameToEnumTable), |
| str); |
| if (format != AUDIO_FORMAT_DEFAULT) { |
| profile->mFormats.add(format); |
| } |
| str = strtok(NULL, "|"); |
| } |
| return; |
| } |
| |
| void AudioPolicyManagerBase::loadInChannels(char *name, IOProfile *profile) |
| { |
| const char *str = strtok(name, "|"); |
| |
| ALOGV("loadInChannels() %s", name); |
| |
| if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) { |
| profile->mChannelMasks.add(0); |
| return; |
| } |
| |
| while (str != NULL) { |
| audio_channel_mask_t channelMask = |
| (audio_channel_mask_t)stringToEnum(sInChannelsNameToEnumTable, |
| ARRAY_SIZE(sInChannelsNameToEnumTable), |
| str); |
| if (channelMask != 0) { |
| ALOGV("loadInChannels() adding channelMask %04x", channelMask); |
| profile->mChannelMasks.add(channelMask); |
| } |
| str = strtok(NULL, "|"); |
| } |
| return; |
| } |
| |
| void AudioPolicyManagerBase::loadOutChannels(char *name, IOProfile *profile) |
| { |
| const char *str = strtok(name, "|"); |
| |
| ALOGV("loadOutChannels() %s", name); |
| |
| // by convention, "0' in the first entry in mChannelMasks indicates the supported channel |
| // masks should be read from the output stream after it is opened for the first time |
| if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) { |
| profile->mChannelMasks.add(0); |
| return; |
| } |
| |
| while (str != NULL) { |
| audio_channel_mask_t channelMask = |
| (audio_channel_mask_t)stringToEnum(sOutChannelsNameToEnumTable, |
| ARRAY_SIZE(sOutChannelsNameToEnumTable), |
| str); |
| if (channelMask != 0) { |
| profile->mChannelMasks.add(channelMask); |
| } |
| str = strtok(NULL, "|"); |
| } |
| return; |
| } |
| |
| status_t AudioPolicyManagerBase::loadInput(cnode *root, HwModule *module) |
| { |
| cnode *node = root->first_child; |
| |
| IOProfile *profile = new IOProfile(module); |
| |
| while (node) { |
| if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) { |
| loadSamplingRates((char *)node->value, profile); |
| } else if (strcmp(node->name, FORMATS_TAG) == 0) { |
| loadFormats((char *)node->value, profile); |
| } else if (strcmp(node->name, CHANNELS_TAG) == 0) { |
| loadInChannels((char *)node->value, profile); |
| } else if (strcmp(node->name, DEVICES_TAG) == 0) { |
| profile->mSupportedDevices = parseDeviceNames((char *)node->value); |
| } |
| node = node->next; |
| } |
| ALOGW_IF(profile->mSupportedDevices == AUDIO_DEVICE_NONE, |
| "loadInput() invalid supported devices"); |
| ALOGW_IF(profile->mChannelMasks.size() == 0, |
| "loadInput() invalid supported channel masks"); |
| ALOGW_IF(profile->mSamplingRates.size() == 0, |
| "loadInput() invalid supported sampling rates"); |
| ALOGW_IF(profile->mFormats.size() == 0, |
| "loadInput() invalid supported formats"); |
| if ((profile->mSupportedDevices != AUDIO_DEVICE_NONE) && |
| (profile->mChannelMasks.size() != 0) && |
| (profile->mSamplingRates.size() != 0) && |
| (profile->mFormats.size() != 0)) { |
| |
| ALOGV("loadInput() adding input mSupportedDevices 0x%X", profile->mSupportedDevices); |
| |
| module->mInputProfiles.add(profile); |
| return NO_ERROR; |
| } else { |
| delete profile; |
| return BAD_VALUE; |
| } |
| } |
| |
| status_t AudioPolicyManagerBase::loadOutput(cnode *root, HwModule *module) |
| { |
| cnode *node = root->first_child; |
| |
| IOProfile *profile = new IOProfile(module); |
| |
| while (node) { |
| if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) { |
| loadSamplingRates((char *)node->value, profile); |
| } else if (strcmp(node->name, FORMATS_TAG) == 0) { |
| loadFormats((char *)node->value, profile); |
| } else if (strcmp(node->name, CHANNELS_TAG) == 0) { |
| loadOutChannels((char *)node->value, profile); |
| } else if (strcmp(node->name, DEVICES_TAG) == 0) { |
| profile->mSupportedDevices = parseDeviceNames((char *)node->value); |
| } else if (strcmp(node->name, FLAGS_TAG) == 0) { |
| profile->mFlags = parseFlagNames((char *)node->value); |
| } |
| node = node->next; |
| } |
| ALOGW_IF(profile->mSupportedDevices == AUDIO_DEVICE_NONE, |
| "loadOutput() invalid supported devices"); |
| ALOGW_IF(profile->mChannelMasks.size() == 0, |
| "loadOutput() invalid supported channel masks"); |
| ALOGW_IF(profile->mSamplingRates.size() == 0, |
| "loadOutput() invalid supported sampling rates"); |
| ALOGW_IF(profile->mFormats.size() == 0, |
| "loadOutput() invalid supported formats"); |
| if ((profile->mSupportedDevices != AUDIO_DEVICE_NONE) && |
| (profile->mChannelMasks.size() != 0) && |
| (profile->mSamplingRates.size() != 0) && |
| (profile->mFormats.size() != 0)) { |
| |
| ALOGV("loadOutput() adding output mSupportedDevices %04x, mFlags %04x", |
| profile->mSupportedDevices, profile->mFlags); |
| |
| module->mOutputProfiles.add(profile); |
| return NO_ERROR; |
| } else { |
| delete profile; |
| return BAD_VALUE; |
| } |
| } |
| |
| void AudioPolicyManagerBase::loadHwModule(cnode *root) |
| { |
| cnode *node = config_find(root, OUTPUTS_TAG); |
| status_t status = NAME_NOT_FOUND; |
| |
| HwModule *module = new HwModule(root->name); |
| |
| if (node != NULL) { |
| if (strcmp(root->name, AUDIO_HARDWARE_MODULE_ID_A2DP) == 0) { |
| mHasA2dp = true; |
| } else if (strcmp(root->name, AUDIO_HARDWARE_MODULE_ID_USB) == 0) { |
| mHasUsb = true; |
| } else if (strcmp(root->name, AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX) == 0) { |
| mHasRemoteSubmix = true; |
| } |
| |
| node = node->first_child; |
| while (node) { |
| ALOGV("loadHwModule() loading output %s", node->name); |
| status_t tmpStatus = loadOutput(node, module); |
| if (status == NAME_NOT_FOUND || status == NO_ERROR) { |
| status = tmpStatus; |
| } |
| node = node->next; |
| } |
| } |
| node = config_find(root, INPUTS_TAG); |
| if (node != NULL) { |
| node = node->first_child; |
| while (node) { |
| ALOGV("loadHwModule() loading input %s", node->name); |
| status_t tmpStatus = loadInput(node, module); |
| if (status == NAME_NOT_FOUND || status == NO_ERROR) { |
| status = tmpStatus; |
| } |
| node = node->next; |
| } |
| } |
| if (status == NO_ERROR) { |
| mHwModules.add(module); |
| } else { |
| delete module; |
| } |
| } |
| |
| void AudioPolicyManagerBase::loadHwModules(cnode *root) |
| { |
| cnode *node = config_find(root, AUDIO_HW_MODULE_TAG); |
| if (node == NULL) { |
| return; |
| } |
| |
| node = node->first_child; |
| while (node) { |
| ALOGV("loadHwModules() loading module %s", node->name); |
| loadHwModule(node); |
| node = node->next; |
| } |
| } |
| |
| void AudioPolicyManagerBase::loadGlobalConfig(cnode *root) |
| { |
| cnode *node = config_find(root, GLOBAL_CONFIG_TAG); |
| if (node == NULL) { |
| return; |
| } |
| node = node->first_child; |
| while (node) { |
| if (strcmp(ATTACHED_OUTPUT_DEVICES_TAG, node->name) == 0) { |
| mAttachedOutputDevices = parseDeviceNames((char *)node->value); |
| ALOGW_IF(mAttachedOutputDevices == AUDIO_DEVICE_NONE, |
| "loadGlobalConfig() no attached output devices"); |
| ALOGV("loadGlobalConfig() mAttachedOutputDevices %04x", mAttachedOutputDevices); |
| } else if (strcmp(DEFAULT_OUTPUT_DEVICE_TAG, node->name) == 0) { |
| mDefaultOutputDevice = (audio_devices_t)stringToEnum(sDeviceNameToEnumTable, |
| ARRAY_SIZE(sDeviceNameToEnumTable), |
| (char *)node->value); |
| ALOGW_IF(mDefaultOutputDevice == AUDIO_DEVICE_NONE, |
| "loadGlobalConfig() default device not specified"); |
| ALOGV("loadGlobalConfig() mDefaultOutputDevice %04x", mDefaultOutputDevice); |
| } else if (strcmp(ATTACHED_INPUT_DEVICES_TAG, node->name) == 0) { |
| mAvailableInputDevices = parseDeviceNames((char *)node->value) & ~AUDIO_DEVICE_BIT_IN; |
| ALOGV("loadGlobalConfig() mAvailableInputDevices %04x", mAvailableInputDevices); |
| } else if (strcmp(SPEAKER_DRC_ENABLED_TAG, node->name) == 0) { |
| mSpeakerDrcEnabled = stringToBool((char *)node->value); |
| ALOGV("loadGlobalConfig() mSpeakerDrcEnabled = %d", mSpeakerDrcEnabled); |
| } |
| node = node->next; |
| } |
| } |
| |
| status_t AudioPolicyManagerBase::loadAudioPolicyConfig(const char *path) |
| { |
| cnode *root; |
| char *data; |
| |
| data = (char *)load_file(path, NULL); |
| if (data == NULL) { |
| return -ENODEV; |
| } |
| root = config_node("", ""); |
| config_load(root, data); |
| |
| loadGlobalConfig(root); |
| loadHwModules(root); |
| |
| config_free(root); |
| free(root); |
| free(data); |
| |
| ALOGI("loadAudioPolicyConfig() loaded %s\n", path); |
| |
| return NO_ERROR; |
| } |
| |
| void AudioPolicyManagerBase::defaultAudioPolicyConfig(void) |
| { |
| HwModule *module; |
| IOProfile *profile; |
| |
| mDefaultOutputDevice = AUDIO_DEVICE_OUT_SPEAKER; |
| mAttachedOutputDevices = AUDIO_DEVICE_OUT_SPEAKER; |
| mAvailableInputDevices = AUDIO_DEVICE_IN_BUILTIN_MIC & ~AUDIO_DEVICE_BIT_IN; |
| |
| module = new HwModule("primary"); |
| |
| profile = new IOProfile(module); |
| profile->mSamplingRates.add(44100); |
| profile->mFormats.add(AUDIO_FORMAT_PCM_16_BIT); |
| profile->mChannelMasks.add(AUDIO_CHANNEL_OUT_STEREO); |
| profile->mSupportedDevices = AUDIO_DEVICE_OUT_SPEAKER; |
| profile->mFlags = AUDIO_OUTPUT_FLAG_PRIMARY; |
| module->mOutputProfiles.add(profile); |
| |
| profile = new IOProfile(module); |
| profile->mSamplingRates.add(8000); |
| profile->mFormats.add(AUDIO_FORMAT_PCM_16_BIT); |
| profile->mChannelMasks.add(AUDIO_CHANNEL_IN_MONO); |
| profile->mSupportedDevices = AUDIO_DEVICE_IN_BUILTIN_MIC; |
| module->mInputProfiles.add(profile); |
| |
| mHwModules.add(module); |
| } |
| |
| }; // namespace android |
