/* * Copyright (C) 2009 The Android Open Source Project * * Licensed under the Apache License, Version 2.0 (the "License"); * you may not use this file except in compliance with the License. * You may obtain a copy of the License at * * http://www.apache.org/licenses/LICENSE-2.0 * * Unless required by applicable law or agreed to in writing, software * distributed under the License is distributed on an "AS IS" BASIS, * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. * See the License for the specific language governing permissions and * limitations under the License. */ #define LOG_TAG "APM::AudioPolicyManager" //#define LOG_NDEBUG 0 //#define VERY_VERBOSE_LOGGING #ifdef VERY_VERBOSE_LOGGING #define ALOGVV ALOGV #else #define ALOGVV(a...) do { } while(0) #endif #include #include #include #include #include #include #include #include #include #include #include #include "AudioPolicyManager.h" #include "audio_policy_conf.h" #include #include namespace android { // ---------------------------------------------------------------------------- // AudioPolicyInterface implementation // ---------------------------------------------------------------------------- status_t AudioPolicyManager::setDeviceConnectionState(audio_devices_t device, audio_policy_dev_state_t state, const char *device_address, const char *device_name) { return setDeviceConnectionStateInt(device, state, device_address, device_name); } status_t AudioPolicyManager::setDeviceConnectionStateInt(audio_devices_t device, audio_policy_dev_state_t state, const char *device_address, const char *device_name) { ALOGV("setDeviceConnectionStateInt() device: 0x%X, state %d, address %s name %s", - device, state, device_address, device_name); // connect/disconnect only 1 device at a time if (!audio_is_output_device(device) && !audio_is_input_device(device)) return BAD_VALUE; sp devDesc = mHwModules.getDeviceDescriptor(device, device_address, device_name); // handle output devices if (audio_is_output_device(device)) { SortedVector outputs; ssize_t index = mAvailableOutputDevices.indexOf(devDesc); // save a copy of the opened output descriptors before any output is opened or closed // by checkOutputsForDevice(). This will be needed by checkOutputForAllStrategies() mPreviousOutputs = mOutputs; switch (state) { // handle output device connection case AUDIO_POLICY_DEVICE_STATE_AVAILABLE: { if (index >= 0) { ALOGW("setDeviceConnectionState() device already connected: %x", device); return INVALID_OPERATION; } ALOGV("setDeviceConnectionState() connecting device %x", device); // register new device as available index = mAvailableOutputDevices.add(devDesc); if (index >= 0) { sp module = mHwModules.getModuleForDevice(device); if (module == 0) { ALOGD("setDeviceConnectionState() could not find HW module for device %08x", device); mAvailableOutputDevices.remove(devDesc); return INVALID_OPERATION; } mAvailableOutputDevices[index]->attach(module); } else { return NO_MEMORY; } if (checkOutputsForDevice(devDesc, state, outputs, devDesc->mAddress) != NO_ERROR) { mAvailableOutputDevices.remove(devDesc); return INVALID_OPERATION; } // Propagate device availability to Engine mEngine->setDeviceConnectionState(devDesc, state); // outputs should never be empty here ALOG_ASSERT(outputs.size() != 0, "setDeviceConnectionState():" "checkOutputsForDevice() returned no outputs but status OK"); ALOGV("setDeviceConnectionState() checkOutputsForDevice() returned %zu outputs", outputs.size()); // Send connect to HALs AudioParameter param = AudioParameter(devDesc->mAddress); param.addInt(String8(AUDIO_PARAMETER_DEVICE_CONNECT), device); mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString()); } break; // handle output device disconnection case AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE: { if (index < 0) { ALOGW("setDeviceConnectionState() device not connected: %x", device); return INVALID_OPERATION; } ALOGV("setDeviceConnectionState() disconnecting output device %x", device); // Send Disconnect to HALs AudioParameter param = AudioParameter(devDesc->mAddress); param.addInt(String8(AUDIO_PARAMETER_DEVICE_DISCONNECT), device); mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString()); // remove device from available output devices mAvailableOutputDevices.remove(devDesc); checkOutputsForDevice(devDesc, state, outputs, devDesc->mAddress); // Propagate device availability to Engine mEngine->setDeviceConnectionState(devDesc, state); } break; default: ALOGE("setDeviceConnectionState() invalid state: %x", state); return BAD_VALUE; } // checkA2dpSuspend must run before checkOutputForAllStrategies so that A2DP // output is suspended before any tracks are moved to it checkA2dpSuspend(); checkOutputForAllStrategies(); // outputs must be closed after checkOutputForAllStrategies() is executed if (!outputs.isEmpty()) { for (size_t i = 0; i < outputs.size(); i++) { sp desc = mOutputs.valueFor(outputs[i]); // close unused outputs after device disconnection or direct outputs that have been // opened by checkOutputsForDevice() to query dynamic parameters if ((state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) || (((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) && (desc->mDirectOpenCount == 0))) { closeOutput(outputs[i]); } } // check again after closing A2DP output to reset mA2dpSuspended if needed checkA2dpSuspend(); } updateDevicesAndOutputs(); if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL) { audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/); updateCallRouting(newDevice); } for (size_t i = 0; i < mOutputs.size(); i++) { sp desc = mOutputs.valueAt(i); if ((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) || (desc != mPrimaryOutput)) { audio_devices_t newDevice = getNewOutputDevice(desc, true /*fromCache*/); // do not force device change on duplicated output because if device is 0, it will // also force a device 0 for the two outputs it is duplicated to which may override // a valid device selection on those outputs. bool force = !desc->isDuplicated() && (!device_distinguishes_on_address(device) // always force when disconnecting (a non-duplicated device) || (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE)); setOutputDevice(desc, newDevice, force, 0); } } mpClientInterface->onAudioPortListUpdate(); return NO_ERROR; } // end if is output device // handle input devices if (audio_is_input_device(device)) { SortedVector inputs; ssize_t index = mAvailableInputDevices.indexOf(devDesc); switch (state) { // handle input device connection case AUDIO_POLICY_DEVICE_STATE_AVAILABLE: { if (index >= 0) { ALOGW("setDeviceConnectionState() device already connected: %d", device); return INVALID_OPERATION; } sp module = mHwModules.getModuleForDevice(device); if (module == NULL) { ALOGW("setDeviceConnectionState(): could not find HW module for device %08x", device); return INVALID_OPERATION; } if (checkInputsForDevice(device, state, inputs, devDesc->mAddress) != NO_ERROR) { return INVALID_OPERATION; } index = mAvailableInputDevices.add(devDesc); if (index >= 0) { mAvailableInputDevices[index]->attach(module); } else { return NO_MEMORY; } // Set connect to HALs AudioParameter param = AudioParameter(devDesc->mAddress); param.addInt(String8(AUDIO_PARAMETER_DEVICE_CONNECT), device); mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString()); // Propagate device availability to Engine mEngine->setDeviceConnectionState(devDesc, state); } break; // handle input device disconnection case AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE: { if (index < 0) { ALOGW("setDeviceConnectionState() device not connected: %d", device); return INVALID_OPERATION; } ALOGV("setDeviceConnectionState() disconnecting input device %x", device); // Set Disconnect to HALs AudioParameter param = AudioParameter(devDesc->mAddress); param.addInt(String8(AUDIO_PARAMETER_DEVICE_DISCONNECT), device); mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString()); checkInputsForDevice(device, state, inputs, devDesc->mAddress); mAvailableInputDevices.remove(devDesc); // Propagate device availability to Engine mEngine->setDeviceConnectionState(devDesc, state); } break; default: ALOGE("setDeviceConnectionState() invalid state: %x", state); return BAD_VALUE; } closeAllInputs(); if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL) { audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/); updateCallRouting(newDevice); } mpClientInterface->onAudioPortListUpdate(); return NO_ERROR; } // end if is input device ALOGW("setDeviceConnectionState() invalid device: %x", device); return BAD_VALUE; } audio_policy_dev_state_t AudioPolicyManager::getDeviceConnectionState(audio_devices_t device, const char *device_address) { sp devDesc = mHwModules.getDeviceDescriptor(device, device_address, ""); DeviceVector *deviceVector; if (audio_is_output_device(device)) { deviceVector = &mAvailableOutputDevices; } else if (audio_is_input_device(device)) { deviceVector = &mAvailableInputDevices; } else { ALOGW("getDeviceConnectionState() invalid device type %08x", device); return AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE; } return deviceVector->getDeviceConnectionState(devDesc); } void AudioPolicyManager::updateCallRouting(audio_devices_t rxDevice, int delayMs) { bool createTxPatch = false; struct audio_patch patch; patch.num_sources = 1; patch.num_sinks = 1; status_t status; audio_patch_handle_t afPatchHandle; DeviceVector deviceList; audio_devices_t txDevice = getDeviceAndMixForInputSource(AUDIO_SOURCE_VOICE_COMMUNICATION); ALOGV("updateCallRouting device rxDevice %08x txDevice %08x", rxDevice, txDevice); // release existing RX patch if any if (mCallRxPatch != 0) { mpClientInterface->releaseAudioPatch(mCallRxPatch->mAfPatchHandle, 0); mCallRxPatch.clear(); } // release TX patch if any if (mCallTxPatch != 0) { mpClientInterface->releaseAudioPatch(mCallTxPatch->mAfPatchHandle, 0); mCallTxPatch.clear(); } // If the RX device is on the primary HW module, then use legacy routing method for voice calls // via setOutputDevice() on primary output. // Otherwise, create two audio patches for TX and RX path. if (availablePrimaryOutputDevices() & rxDevice) { setOutputDevice(mPrimaryOutput, rxDevice, true, delayMs); // If the TX device is also on the primary HW module, setOutputDevice() will take care // of it due to legacy implementation. If not, create a patch. if ((availablePrimaryInputDevices() & txDevice & ~AUDIO_DEVICE_BIT_IN) == AUDIO_DEVICE_NONE) { createTxPatch = true; } } else { // create RX path audio patch deviceList = mAvailableOutputDevices.getDevicesFromType(rxDevice); ALOG_ASSERT(!deviceList.isEmpty(), "updateCallRouting() selected device not in output device list"); sp rxSinkDeviceDesc = deviceList.itemAt(0); deviceList = mAvailableInputDevices.getDevicesFromType(AUDIO_DEVICE_IN_TELEPHONY_RX); ALOG_ASSERT(!deviceList.isEmpty(), "updateCallRouting() no telephony RX device"); sp rxSourceDeviceDesc = deviceList.itemAt(0); rxSourceDeviceDesc->toAudioPortConfig(&patch.sources[0]); rxSinkDeviceDesc->toAudioPortConfig(&patch.sinks[0]); // request to reuse existing output stream if one is already opened to reach the RX device SortedVector outputs = getOutputsForDevice(rxDevice, mOutputs); audio_io_handle_t output = selectOutput(outputs, AUDIO_OUTPUT_FLAG_NONE, AUDIO_FORMAT_INVALID); if (output != AUDIO_IO_HANDLE_NONE) { sp outputDesc = mOutputs.valueFor(output); ALOG_ASSERT(!outputDesc->isDuplicated(), "updateCallRouting() RX device output is duplicated"); outputDesc->toAudioPortConfig(&patch.sources[1]); patch.sources[1].ext.mix.usecase.stream = AUDIO_STREAM_PATCH; patch.num_sources = 2; } afPatchHandle = AUDIO_PATCH_HANDLE_NONE; status = mpClientInterface->createAudioPatch(&patch, &afPatchHandle, 0); ALOGW_IF(status != NO_ERROR, "updateCallRouting() error %d creating RX audio patch", status); if (status == NO_ERROR) { mCallRxPatch = new AudioPatch(&patch, mUidCached); mCallRxPatch->mAfPatchHandle = afPatchHandle; mCallRxPatch->mUid = mUidCached; } createTxPatch = true; } if (createTxPatch) { struct audio_patch patch; patch.num_sources = 1; patch.num_sinks = 1; deviceList = mAvailableInputDevices.getDevicesFromType(txDevice); ALOG_ASSERT(!deviceList.isEmpty(), "updateCallRouting() selected device not in input device list"); sp txSourceDeviceDesc = deviceList.itemAt(0); txSourceDeviceDesc->toAudioPortConfig(&patch.sources[0]); deviceList = mAvailableOutputDevices.getDevicesFromType(AUDIO_DEVICE_OUT_TELEPHONY_TX); ALOG_ASSERT(!deviceList.isEmpty(), "updateCallRouting() no telephony TX device"); sp txSinkDeviceDesc = deviceList.itemAt(0); txSinkDeviceDesc->toAudioPortConfig(&patch.sinks[0]); SortedVector outputs = getOutputsForDevice(AUDIO_DEVICE_OUT_TELEPHONY_TX, mOutputs); audio_io_handle_t output = selectOutput(outputs, AUDIO_OUTPUT_FLAG_NONE, AUDIO_FORMAT_INVALID); // request to reuse existing output stream if one is already opened to reach the TX // path output device if (output != AUDIO_IO_HANDLE_NONE) { sp outputDesc = mOutputs.valueFor(output); ALOG_ASSERT(!outputDesc->isDuplicated(), "updateCallRouting() RX device output is duplicated"); outputDesc->toAudioPortConfig(&patch.sources[1]); patch.sources[1].ext.mix.usecase.stream = AUDIO_STREAM_PATCH; patch.num_sources = 2; } afPatchHandle = AUDIO_PATCH_HANDLE_NONE; status = mpClientInterface->createAudioPatch(&patch, &afPatchHandle, 0); ALOGW_IF(status != NO_ERROR, "setPhoneState() error %d creating TX audio patch", status); if (status == NO_ERROR) { mCallTxPatch = new AudioPatch(&patch, mUidCached); mCallTxPatch->mAfPatchHandle = afPatchHandle; mCallTxPatch->mUid = mUidCached; } } } void AudioPolicyManager::setPhoneState(audio_mode_t state) { ALOGV("setPhoneState() state %d", state); // store previous phone state for management of sonification strategy below int oldState = mEngine->getPhoneState(); if (mEngine->setPhoneState(state) != NO_ERROR) { ALOGW("setPhoneState() invalid or same state %d", state); return; } /// Opens: can these line be executed after the switch of volume curves??? // if leaving call state, handle special case of active streams // pertaining to sonification strategy see handleIncallSonification() if (isInCall()) { ALOGV("setPhoneState() in call state management: new state is %d", state); for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) { if (stream == AUDIO_STREAM_PATCH) { continue; } handleIncallSonification((audio_stream_type_t)stream, false, true); } // force reevaluating accessibility routing when call starts mpClientInterface->invalidateStream(AUDIO_STREAM_ACCESSIBILITY); } /** * Switching to or from incall state or switching between telephony and VoIP lead to force * routing command. */ bool force = ((is_state_in_call(oldState) != is_state_in_call(state)) || (is_state_in_call(state) && (state != oldState))); // check for device and output changes triggered by new phone state checkA2dpSuspend(); checkOutputForAllStrategies(); updateDevicesAndOutputs(); sp hwOutputDesc = mPrimaryOutput; int delayMs = 0; if (isStateInCall(state)) { nsecs_t sysTime = systemTime(); for (size_t i = 0; i < mOutputs.size(); i++) { sp desc = mOutputs.valueAt(i); // mute media and sonification strategies and delay device switch by the largest // latency of any output where either strategy is active. // This avoid sending the ring tone or music tail into the earpiece or headset. if ((isStrategyActive(desc, STRATEGY_MEDIA, SONIFICATION_HEADSET_MUSIC_DELAY, sysTime) || isStrategyActive(desc, STRATEGY_SONIFICATION, SONIFICATION_HEADSET_MUSIC_DELAY, sysTime)) && (delayMs < (int)desc->latency()*2)) { delayMs = desc->latency()*2; } setStrategyMute(STRATEGY_MEDIA, true, desc); setStrategyMute(STRATEGY_MEDIA, false, desc, MUTE_TIME_MS, getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/)); setStrategyMute(STRATEGY_SONIFICATION, true, desc); setStrategyMute(STRATEGY_SONIFICATION, false, desc, MUTE_TIME_MS, getDeviceForStrategy(STRATEGY_SONIFICATION, true /*fromCache*/)); } } // Note that despite the fact that getNewOutputDevice() is called on the primary output, // the device returned is not necessarily reachable via this output audio_devices_t rxDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/); // force routing command to audio hardware when ending call // even if no device change is needed if (isStateInCall(oldState) && rxDevice == AUDIO_DEVICE_NONE) { rxDevice = hwOutputDesc->device(); } if (state == AUDIO_MODE_IN_CALL) { updateCallRouting(rxDevice, delayMs); } else if (oldState == AUDIO_MODE_IN_CALL) { if (mCallRxPatch != 0) { mpClientInterface->releaseAudioPatch(mCallRxPatch->mAfPatchHandle, 0); mCallRxPatch.clear(); } if (mCallTxPatch != 0) { mpClientInterface->releaseAudioPatch(mCallTxPatch->mAfPatchHandle, 0); mCallTxPatch.clear(); } setOutputDevice(mPrimaryOutput, rxDevice, force, 0); } else { setOutputDevice(mPrimaryOutput, rxDevice, force, 0); } // if entering in call state, handle special case of active streams // pertaining to sonification strategy see handleIncallSonification() if (isStateInCall(state)) { ALOGV("setPhoneState() in call state management: new state is %d", state); for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) { if (stream == AUDIO_STREAM_PATCH) { continue; } handleIncallSonification((audio_stream_type_t)stream, true, true); } } // Flag that ringtone volume must be limited to music volume until we exit MODE_RINGTONE if (state == AUDIO_MODE_RINGTONE && isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY)) { mLimitRingtoneVolume = true; } else { mLimitRingtoneVolume = false; } } audio_mode_t AudioPolicyManager::getPhoneState() { return mEngine->getPhoneState(); } void AudioPolicyManager::setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config) { ALOGV("setForceUse() usage %d, config %d, mPhoneState %d", usage, config, mEngine->getPhoneState()); if (mEngine->setForceUse(usage, config) != NO_ERROR) { ALOGW("setForceUse() could not set force cfg %d for usage %d", config, usage); return; } bool forceVolumeReeval = (usage == AUDIO_POLICY_FORCE_FOR_COMMUNICATION) || (usage == AUDIO_POLICY_FORCE_FOR_DOCK) || (usage == AUDIO_POLICY_FORCE_FOR_SYSTEM); // check for device and output changes triggered by new force usage checkA2dpSuspend(); checkOutputForAllStrategies(); updateDevicesAndOutputs(); if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL) { audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, true /*fromCache*/); updateCallRouting(newDevice); } for (size_t i = 0; i < mOutputs.size(); i++) { sp outputDesc = mOutputs.valueAt(i); audio_devices_t newDevice = getNewOutputDevice(outputDesc, true /*fromCache*/); if ((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) || (outputDesc != mPrimaryOutput)) { setOutputDevice(outputDesc, newDevice, (newDevice != AUDIO_DEVICE_NONE)); } if (forceVolumeReeval && (newDevice != AUDIO_DEVICE_NONE)) { applyStreamVolumes(outputDesc, newDevice, 0, true); } } audio_io_handle_t activeInput = mInputs.getActiveInput(); if (activeInput != 0) { setInputDevice(activeInput, getNewInputDevice(activeInput)); } } void AudioPolicyManager::setSystemProperty(const char* property, const char* value) { ALOGV("setSystemProperty() property %s, value %s", property, value); } // Find a direct output profile compatible with the parameters passed, even if the input flags do // not explicitly request a direct output sp AudioPolicyManager::getProfileForDirectOutput( audio_devices_t device, uint32_t samplingRate, audio_format_t format, audio_channel_mask_t channelMask, audio_output_flags_t flags) { for (size_t i = 0; i < mHwModules.size(); i++) { if (mHwModules[i]->mHandle == 0) { continue; } for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++) { sp profile = mHwModules[i]->mOutputProfiles[j]; bool found = profile->isCompatibleProfile(device, String8(""), samplingRate, NULL /*updatedSamplingRate*/, format, NULL /*updatedFormat*/, channelMask, NULL /*updatedChannelMask*/, flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD ? AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD : AUDIO_OUTPUT_FLAG_DIRECT); if (found && (mAvailableOutputDevices.types() & profile->mSupportedDevices.types())) { return profile; } } } return 0; } audio_io_handle_t AudioPolicyManager::getOutput(audio_stream_type_t stream, uint32_t samplingRate, audio_format_t format, audio_channel_mask_t channelMask, audio_output_flags_t flags, const audio_offload_info_t *offloadInfo) { routing_strategy strategy = getStrategy(stream); audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/); ALOGV("getOutput() device %d, stream %d, samplingRate %d, format %x, channelMask %x, flags %x", device, stream, samplingRate, format, channelMask, flags); return getOutputForDevice(device, AUDIO_SESSION_ALLOCATE, stream, samplingRate,format, channelMask, flags, offloadInfo); } status_t AudioPolicyManager::getOutputForAttr(const audio_attributes_t *attr, audio_io_handle_t *output, audio_session_t session, audio_stream_type_t *stream, uint32_t samplingRate, audio_format_t format, audio_channel_mask_t channelMask, audio_output_flags_t flags, audio_port_handle_t selectedDeviceId, const audio_offload_info_t *offloadInfo) { audio_attributes_t attributes; if (attr != NULL) { if (!isValidAttributes(attr)) { ALOGE("getOutputForAttr() invalid attributes: usage=%d content=%d flags=0x%x tags=[%s]", attr->usage, attr->content_type, attr->flags, attr->tags); return BAD_VALUE; } attributes = *attr; } else { if (*stream < AUDIO_STREAM_MIN || *stream >= AUDIO_STREAM_PUBLIC_CNT) { ALOGE("getOutputForAttr(): invalid stream type"); return BAD_VALUE; } stream_type_to_audio_attributes(*stream, &attributes); } sp desc; if (mPolicyMixes.getOutputForAttr(attributes, desc) == NO_ERROR) { ALOG_ASSERT(desc != 0, "Invalid desc returned by getOutputForAttr"); if (!audio_is_linear_pcm(format)) { return BAD_VALUE; } *stream = streamTypefromAttributesInt(&attributes); *output = desc->mIoHandle; ALOGV("getOutputForAttr() returns output %d", *output); return NO_ERROR; } if (attributes.usage == AUDIO_USAGE_VIRTUAL_SOURCE) { ALOGW("getOutputForAttr() no policy mix found for usage AUDIO_USAGE_VIRTUAL_SOURCE"); return BAD_VALUE; } ALOGV("getOutputForAttr() usage=%d, content=%d, tag=%s flags=%08x", attributes.usage, attributes.content_type, attributes.tags, attributes.flags); routing_strategy strategy = (routing_strategy) getStrategyForAttr(&attributes); audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/); if ((attributes.flags & AUDIO_FLAG_HW_AV_SYNC) != 0) { flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_HW_AV_SYNC); } ALOGV("getOutputForAttr() device 0x%x, samplingRate %d, format %x, channelMask %x, flags %x", device, samplingRate, format, channelMask, flags); *stream = streamTypefromAttributesInt(&attributes); *output = getOutputForDevice(device, session, *stream, samplingRate, format, channelMask, flags, offloadInfo); if (*output == AUDIO_IO_HANDLE_NONE) { return INVALID_OPERATION; } // Explicit routing? sp deviceDesc; for (size_t i = 0; i < mAvailableOutputDevices.size(); i++) { if (mAvailableOutputDevices[i]->getId() == selectedDeviceId) { deviceDesc = mAvailableOutputDevices[i]; break; } } mOutputRoutes.addRoute(session, *stream, SessionRoute::SOURCE_TYPE_NA, deviceDesc); return NO_ERROR; } audio_io_handle_t AudioPolicyManager::getOutputForDevice( audio_devices_t device, audio_session_t session __unused, audio_stream_type_t stream, uint32_t samplingRate, audio_format_t format, audio_channel_mask_t channelMask, audio_output_flags_t flags, const audio_offload_info_t *offloadInfo) { audio_io_handle_t output = AUDIO_IO_HANDLE_NONE; uint32_t latency = 0; status_t status; #ifdef AUDIO_POLICY_TEST if (mCurOutput != 0) { ALOGV("getOutput() test output mCurOutput %d, samplingRate %d, format %d, channelMask %x, mDirectOutput %d", mCurOutput, mTestSamplingRate, mTestFormat, mTestChannels, mDirectOutput); if (mTestOutputs[mCurOutput] == 0) { ALOGV("getOutput() opening test output"); sp outputDesc = new SwAudioOutputDescriptor(NULL, mpClientInterface); outputDesc->mDevice = mTestDevice; outputDesc->mLatency = mTestLatencyMs; outputDesc->mFlags = (audio_output_flags_t)(mDirectOutput ? AUDIO_OUTPUT_FLAG_DIRECT : 0); outputDesc->mRefCount[stream] = 0; audio_config_t config = AUDIO_CONFIG_INITIALIZER; config.sample_rate = mTestSamplingRate; config.channel_mask = mTestChannels; config.format = mTestFormat; if (offloadInfo != NULL) { config.offload_info = *offloadInfo; } status = mpClientInterface->openOutput(0, &mTestOutputs[mCurOutput], &config, &outputDesc->mDevice, String8(""), &outputDesc->mLatency, outputDesc->mFlags); if (status == NO_ERROR) { outputDesc->mSamplingRate = config.sample_rate; outputDesc->mFormat = config.format; outputDesc->mChannelMask = config.channel_mask; AudioParameter outputCmd = AudioParameter(); outputCmd.addInt(String8("set_id"),mCurOutput); mpClientInterface->setParameters(mTestOutputs[mCurOutput],outputCmd.toString()); addOutput(mTestOutputs[mCurOutput], outputDesc); } } return mTestOutputs[mCurOutput]; } #endif //AUDIO_POLICY_TEST // open a direct output if required by specified parameters //force direct flag if offload flag is set: offloading implies a direct output stream // and all common behaviors are driven by checking only the direct flag // this should normally be set appropriately in the policy configuration file if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) { flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT); } if ((flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) != 0) { flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT); } // only allow deep buffering for music stream type if (stream != AUDIO_STREAM_MUSIC) { flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER); } sp profile; // skip direct output selection if the request can obviously be attached to a mixed output // and not explicitly requested if (((flags & AUDIO_OUTPUT_FLAG_DIRECT) == 0) && audio_is_linear_pcm(format) && samplingRate <= MAX_MIXER_SAMPLING_RATE && audio_channel_count_from_out_mask(channelMask) <= 2) { goto non_direct_output; } // Do not allow offloading if one non offloadable effect is enabled. This prevents from // creating an offloaded track and tearing it down immediately after start when audioflinger // detects there is an active non offloadable effect. // FIXME: We should check the audio session here but we do not have it in this context. // This may prevent offloading in rare situations where effects are left active by apps // in the background. if (((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) == 0) || !mEffects.isNonOffloadableEffectEnabled()) { profile = getProfileForDirectOutput(device, samplingRate, format, channelMask, (audio_output_flags_t)flags); } if (profile != 0) { sp outputDesc = NULL; for (size_t i = 0; i < mOutputs.size(); i++) { sp desc = mOutputs.valueAt(i); if (!desc->isDuplicated() && (profile == desc->mProfile)) { outputDesc = desc; // reuse direct output if currently open and configured with same parameters if ((samplingRate == outputDesc->mSamplingRate) && (format == outputDesc->mFormat) && (channelMask == outputDesc->mChannelMask)) { outputDesc->mDirectOpenCount++; ALOGV("getOutput() reusing direct output %d", mOutputs.keyAt(i)); return mOutputs.keyAt(i); } } } // close direct output if currently open and configured with different parameters if (outputDesc != NULL) { closeOutput(outputDesc->mIoHandle); } outputDesc = new SwAudioOutputDescriptor(profile, mpClientInterface); outputDesc->mDevice = device; outputDesc->mLatency = 0; outputDesc->mFlags =(audio_output_flags_t) (outputDesc->mFlags | flags); audio_config_t config = AUDIO_CONFIG_INITIALIZER; config.sample_rate = samplingRate; config.channel_mask = channelMask; config.format = format; if (offloadInfo != NULL) { config.offload_info = *offloadInfo; } status = mpClientInterface->openOutput(profile->getModuleHandle(), &output, &config, &outputDesc->mDevice, String8(""), &outputDesc->mLatency, outputDesc->mFlags); // only accept an output with the requested parameters if (status != NO_ERROR || (samplingRate != 0 && samplingRate != config.sample_rate) || (format != AUDIO_FORMAT_DEFAULT && format != config.format) || (channelMask != 0 && channelMask != config.channel_mask)) { ALOGV("getOutput() failed opening direct output: output %d samplingRate %d %d," "format %d %d, channelMask %04x %04x", output, samplingRate, outputDesc->mSamplingRate, format, outputDesc->mFormat, channelMask, outputDesc->mChannelMask); if (output != AUDIO_IO_HANDLE_NONE) { mpClientInterface->closeOutput(output); } // fall back to mixer output if possible when the direct output could not be open if (audio_is_linear_pcm(format) && samplingRate <= MAX_MIXER_SAMPLING_RATE) { goto non_direct_output; } // fall back to mixer output if possible when the direct output could not be open if (audio_is_linear_pcm(format) && samplingRate <= MAX_MIXER_SAMPLING_RATE) { goto non_direct_output; } return AUDIO_IO_HANDLE_NONE; } outputDesc->mSamplingRate = config.sample_rate; outputDesc->mChannelMask = config.channel_mask; outputDesc->mFormat = config.format; outputDesc->mRefCount[stream] = 0; outputDesc->mStopTime[stream] = 0; outputDesc->mDirectOpenCount = 1; audio_io_handle_t srcOutput = getOutputForEffect(); addOutput(output, outputDesc); audio_io_handle_t dstOutput = getOutputForEffect(); if (dstOutput == output) { mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, srcOutput, dstOutput); } mPreviousOutputs = mOutputs; ALOGV("getOutput() returns new direct output %d", output); mpClientInterface->onAudioPortListUpdate(); return output; } non_direct_output: // ignoring channel mask due to downmix capability in mixer // open a non direct output // for non direct outputs, only PCM is supported if (audio_is_linear_pcm(format)) { // get which output is suitable for the specified stream. The actual // routing change will happen when startOutput() will be called SortedVector outputs = getOutputsForDevice(device, mOutputs); // at this stage we should ignore the DIRECT flag as no direct output could be found earlier flags = (audio_output_flags_t)(flags & ~AUDIO_OUTPUT_FLAG_DIRECT); output = selectOutput(outputs, flags, format); } ALOGW_IF((output == 0), "getOutput() could not find output for stream %d, samplingRate %d," "format %d, channels %x, flags %x", stream, samplingRate, format, channelMask, flags); ALOGV(" getOutputForDevice() returns output %d", output); return output; } audio_io_handle_t AudioPolicyManager::selectOutput(const SortedVector& outputs, audio_output_flags_t flags, audio_format_t format) { // select one output among several that provide a path to a particular device or set of // devices (the list was previously build by getOutputsForDevice()). // The priority is as follows: // 1: the output with the highest number of requested policy flags // 2: the primary output // 3: the first output in the list if (outputs.size() == 0) { return 0; } if (outputs.size() == 1) { return outputs[0]; } int maxCommonFlags = 0; audio_io_handle_t outputFlags = 0; audio_io_handle_t outputPrimary = 0; for (size_t i = 0; i < outputs.size(); i++) { sp outputDesc = mOutputs.valueFor(outputs[i]); if (!outputDesc->isDuplicated()) { // if a valid format is specified, skip output if not compatible if (format != AUDIO_FORMAT_INVALID) { if (outputDesc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) { if (format != outputDesc->mFormat) { continue; } } else if (!audio_is_linear_pcm(format)) { continue; } } int commonFlags = popcount(outputDesc->mProfile->mFlags & flags); if (commonFlags > maxCommonFlags) { outputFlags = outputs[i]; maxCommonFlags = commonFlags; ALOGV("selectOutput() commonFlags for output %d, %04x", outputs[i], commonFlags); } if (outputDesc->mProfile->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY) { outputPrimary = outputs[i]; } } } if (outputFlags != 0) { return outputFlags; } if (outputPrimary != 0) { return outputPrimary; } return outputs[0]; } status_t AudioPolicyManager::startOutput(audio_io_handle_t output, audio_stream_type_t stream, audio_session_t session) { ALOGV("startOutput() output %d, stream %d, session %d", output, stream, session); ssize_t index = mOutputs.indexOfKey(output); if (index < 0) { ALOGW("startOutput() unknown output %d", output); return BAD_VALUE; } sp outputDesc = mOutputs.valueAt(index); audio_devices_t newDevice; if (outputDesc->mPolicyMix != NULL) { newDevice = AUDIO_DEVICE_OUT_REMOTE_SUBMIX; } else if (mOutputRoutes.hasRouteChanged(session)) { newDevice = getNewOutputDevice(outputDesc, false /*fromCache*/); } else { newDevice = AUDIO_DEVICE_NONE; } uint32_t delayMs = 0; // Routing? mOutputRoutes.incRouteActivity(session); status_t status = startSource(outputDesc, stream, newDevice, &delayMs); if (status != NO_ERROR) { mOutputRoutes.decRouteActivity(session); } // Automatically enable the remote submix input when output is started on a re routing mix // of type MIX_TYPE_RECORDERS if (audio_is_remote_submix_device(newDevice) && outputDesc->mPolicyMix != NULL && outputDesc->mPolicyMix->mMixType == MIX_TYPE_RECORDERS) { setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX, AUDIO_POLICY_DEVICE_STATE_AVAILABLE, outputDesc->mPolicyMix->mRegistrationId, "remote-submix"); } if (delayMs != 0) { usleep(delayMs * 1000); } return status; } status_t AudioPolicyManager::startSource(sp outputDesc, audio_stream_type_t stream, audio_devices_t device, uint32_t *delayMs) { // cannot start playback of STREAM_TTS if any other output is being used uint32_t beaconMuteLatency = 0; *delayMs = 0; if (stream == AUDIO_STREAM_TTS) { ALOGV("\t found BEACON stream"); if (mOutputs.isAnyOutputActive(AUDIO_STREAM_TTS /*streamToIgnore*/)) { return INVALID_OPERATION; } else { beaconMuteLatency = handleEventForBeacon(STARTING_BEACON); } } else { // some playback other than beacon starts beaconMuteLatency = handleEventForBeacon(STARTING_OUTPUT); } // 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 || device != AUDIO_DEVICE_NONE) { // starting an output being rerouted? if (device == AUDIO_DEVICE_NONE) { device = getNewOutputDevice(outputDesc, false /*fromCache*/); } routing_strategy strategy = getStrategy(stream); bool shouldWait = (strategy == STRATEGY_SONIFICATION) || (strategy == STRATEGY_SONIFICATION_RESPECTFUL) || (beaconMuteLatency > 0); uint32_t waitMs = beaconMuteLatency; bool force = false; for (size_t i = 0; i < mOutputs.size(); i++) { sp desc = mOutputs.valueAt(i); if (desc != outputDesc) { // force a device change if any other output is managed by the same hw // module and has a current device selection that differs from selected device. // In this case, the audio HAL must receive the new device selection so that it can // change the device currently selected by the other active output. if (outputDesc->sharesHwModuleWith(desc) && desc->device() != device) { force = true; } // wait for audio on other active outputs to be presented when starting // a notification so that audio focus effect can propagate, or that a mute/unmute // event occurred for beacon uint32_t latency = desc->latency(); if (shouldWait && desc->isActive(latency * 2) && (waitMs < latency)) { waitMs = latency; } } } uint32_t muteWaitMs = setOutputDevice(outputDesc, device, 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.valueFor(stream).getVolumeIndex(device), outputDesc, device); // update the outputs if starting an output with a stream that can affect notification // routing handleNotificationRoutingForStream(stream); // force reevaluating accessibility routing when ringtone or alarm starts if (strategy == STRATEGY_SONIFICATION) { mpClientInterface->invalidateStream(AUDIO_STREAM_ACCESSIBILITY); } } return NO_ERROR; } status_t AudioPolicyManager::stopOutput(audio_io_handle_t output, audio_stream_type_t stream, audio_session_t session) { ALOGV("stopOutput() output %d, stream %d, session %d", output, stream, session); ssize_t index = mOutputs.indexOfKey(output); if (index < 0) { ALOGW("stopOutput() unknown output %d", output); return BAD_VALUE; } sp outputDesc = mOutputs.valueAt(index); if (outputDesc->mRefCount[stream] == 1) { // Automatically disable the remote submix input when output is stopped on a // re routing mix of type MIX_TYPE_RECORDERS if (audio_is_remote_submix_device(outputDesc->mDevice) && outputDesc->mPolicyMix != NULL && outputDesc->mPolicyMix->mMixType == MIX_TYPE_RECORDERS) { setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX, AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE, outputDesc->mPolicyMix->mRegistrationId, "remote-submix"); } } // Routing? if (outputDesc->mRefCount[stream] > 0) { mOutputRoutes.decRouteActivity(session); } return stopSource(outputDesc, stream); } status_t AudioPolicyManager::stopSource(sp outputDesc, audio_stream_type_t stream) { // always handle stream stop, check which stream type is stopping handleEventForBeacon(stream == AUDIO_STREAM_TTS ? STOPPING_BEACON : STOPPING_OUTPUT); // handle special case for sonification while in call if (isInCall()) { handleIncallSonification(stream, false, false); } if (outputDesc->mRefCount[stream] > 0) { // decrement usage count of this stream on the output outputDesc->changeRefCount(stream, -1); // store time at which the stream was stopped - see isStreamActive() if (outputDesc->mRefCount[stream] == 0) { outputDesc->mStopTime[stream] = systemTime(); audio_devices_t newDevice = getNewOutputDevice(outputDesc, 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(outputDesc, newDevice, false, outputDesc->latency()*2); // force restoring the device selection on other active outputs if it differs from the // one being selected for this output for (size_t i = 0; i < mOutputs.size(); i++) { audio_io_handle_t curOutput = mOutputs.keyAt(i); sp desc = mOutputs.valueAt(i); if (desc != outputDesc && desc->isActive() && outputDesc->sharesHwModuleWith(desc) && (newDevice != desc->device())) { setOutputDevice(desc, getNewOutputDevice(desc, false /*fromCache*/), true, outputDesc->latency()*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"); return INVALID_OPERATION; } } void AudioPolicyManager::releaseOutput(audio_io_handle_t output, audio_stream_type_t stream __unused, audio_session_t session __unused) { ALOGV("releaseOutput() %d", output); ssize_t index = mOutputs.indexOfKey(output); if (index < 0) { ALOGW("releaseOutput() releasing unknown output %d", output); return; } #ifdef AUDIO_POLICY_TEST int testIndex = testOutputIndex(output); if (testIndex != 0) { sp outputDesc = mOutputs.valueAt(index); if (outputDesc->isActive()) { mpClientInterface->closeOutput(output); removeOutput(output); mTestOutputs[testIndex] = 0; } return; } #endif //AUDIO_POLICY_TEST // Routing mOutputRoutes.removeRoute(session); sp desc = mOutputs.valueAt(index); if (desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) { if (desc->mDirectOpenCount <= 0) { ALOGW("releaseOutput() invalid open count %d for output %d", desc->mDirectOpenCount, output); return; } if (--desc->mDirectOpenCount == 0) { closeOutput(output); // If effects where present on the output, audioflinger moved them to the primary // output by default: move them back to the appropriate output. audio_io_handle_t dstOutput = getOutputForEffect(); if (dstOutput != mPrimaryOutput->mIoHandle) { mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, mPrimaryOutput->mIoHandle, dstOutput); } mpClientInterface->onAudioPortListUpdate(); } } } status_t AudioPolicyManager::getInputForAttr(const audio_attributes_t *attr, audio_io_handle_t *input, audio_session_t session, uint32_t samplingRate, audio_format_t format, audio_channel_mask_t channelMask, audio_input_flags_t flags, audio_port_handle_t selectedDeviceId, input_type_t *inputType) { ALOGV("getInputForAttr() source %d, samplingRate %d, format %d, channelMask %x," "session %d, flags %#x", attr->source, samplingRate, format, channelMask, session, flags); *input = AUDIO_IO_HANDLE_NONE; *inputType = API_INPUT_INVALID; audio_devices_t device; // handle legacy remote submix case where the address was not always specified String8 address = String8(""); bool isSoundTrigger = false; audio_source_t inputSource = attr->source; audio_source_t halInputSource; AudioMix *policyMix = NULL; if (inputSource == AUDIO_SOURCE_DEFAULT) { inputSource = AUDIO_SOURCE_MIC; } halInputSource = inputSource; // Explicit routing? sp deviceDesc; for (size_t i = 0; i < mAvailableInputDevices.size(); i++) { if (mAvailableInputDevices[i]->getId() == selectedDeviceId) { deviceDesc = mAvailableInputDevices[i]; break; } } mInputRoutes.addRoute(session, SessionRoute::STREAM_TYPE_NA, inputSource, deviceDesc); if (inputSource == AUDIO_SOURCE_REMOTE_SUBMIX && strncmp(attr->tags, "addr=", strlen("addr=")) == 0) { status_t ret = mPolicyMixes.getInputMixForAttr(*attr, &policyMix); if (ret != NO_ERROR) { return ret; } *inputType = API_INPUT_MIX_EXT_POLICY_REROUTE; device = AUDIO_DEVICE_IN_REMOTE_SUBMIX; address = String8(attr->tags + strlen("addr=")); } else { device = getDeviceAndMixForInputSource(inputSource, &policyMix); if (device == AUDIO_DEVICE_NONE) { ALOGW("getInputForAttr() could not find device for source %d", inputSource); return BAD_VALUE; } if (policyMix != NULL) { address = policyMix->mRegistrationId; if (policyMix->mMixType == MIX_TYPE_RECORDERS) { // there is an external policy, but this input is attached to a mix of recorders, // meaning it receives audio injected into the framework, so the recorder doesn't // know about it and is therefore considered "legacy" *inputType = API_INPUT_LEGACY; } else { // recording a mix of players defined by an external policy, we're rerouting for // an external policy *inputType = API_INPUT_MIX_EXT_POLICY_REROUTE; } } else if (audio_is_remote_submix_device(device)) { address = String8("0"); *inputType = API_INPUT_MIX_CAPTURE; } else { *inputType = API_INPUT_LEGACY; } // adapt channel selection to input source switch (inputSource) { case AUDIO_SOURCE_VOICE_UPLINK: channelMask = AUDIO_CHANNEL_IN_VOICE_UPLINK; break; case AUDIO_SOURCE_VOICE_DOWNLINK: channelMask = AUDIO_CHANNEL_IN_VOICE_DNLINK; break; case AUDIO_SOURCE_VOICE_CALL: channelMask = AUDIO_CHANNEL_IN_VOICE_UPLINK | AUDIO_CHANNEL_IN_VOICE_DNLINK; break; default: break; } if (inputSource == AUDIO_SOURCE_HOTWORD) { ssize_t index = mSoundTriggerSessions.indexOfKey(session); if (index >= 0) { *input = mSoundTriggerSessions.valueFor(session); isSoundTrigger = true; flags = (audio_input_flags_t)(flags | AUDIO_INPUT_FLAG_HW_HOTWORD); ALOGV("SoundTrigger capture on session %d input %d", session, *input); } else { halInputSource = AUDIO_SOURCE_VOICE_RECOGNITION; } } } // find a compatible input profile (not necessarily identical in parameters) sp profile; // samplingRate and flags may be updated by getInputProfile uint32_t profileSamplingRate = samplingRate; audio_format_t profileFormat = format; audio_channel_mask_t profileChannelMask = channelMask; audio_input_flags_t profileFlags = flags; for (;;) { profile = getInputProfile(device, address, profileSamplingRate, profileFormat, profileChannelMask, profileFlags); if (profile != 0) { break; // success } else if (profileFlags != AUDIO_INPUT_FLAG_NONE) { profileFlags = AUDIO_INPUT_FLAG_NONE; // retry } else { // fail ALOGW("getInputForAttr() could not find profile for device 0x%X, samplingRate %u," "format %#x, channelMask 0x%X, flags %#x", device, samplingRate, format, channelMask, flags); return BAD_VALUE; } } if (profile->getModuleHandle() == 0) { ALOGE("getInputForAttr(): HW module %s not opened", profile->getModuleName()); return NO_INIT; } audio_config_t config = AUDIO_CONFIG_INITIALIZER; config.sample_rate = profileSamplingRate; config.channel_mask = profileChannelMask; config.format = profileFormat; status_t status = mpClientInterface->openInput(profile->getModuleHandle(), input, &config, &device, address, halInputSource, profileFlags); // only accept input with the exact requested set of parameters if (status != NO_ERROR || *input == AUDIO_IO_HANDLE_NONE || (profileSamplingRate != config.sample_rate) || (profileFormat != config.format) || (profileChannelMask != config.channel_mask)) { ALOGW("getInputForAttr() failed opening input: samplingRate %d, format %d," " channelMask %x", samplingRate, format, channelMask); if (*input != AUDIO_IO_HANDLE_NONE) { mpClientInterface->closeInput(*input); } return BAD_VALUE; } sp inputDesc = new AudioInputDescriptor(profile); inputDesc->mInputSource = inputSource; inputDesc->mRefCount = 0; inputDesc->mOpenRefCount = 1; inputDesc->mSamplingRate = profileSamplingRate; inputDesc->mFormat = profileFormat; inputDesc->mChannelMask = profileChannelMask; inputDesc->mDevice = device; inputDesc->mSessions.add(session); inputDesc->mIsSoundTrigger = isSoundTrigger; inputDesc->mPolicyMix = policyMix; ALOGV("getInputForAttr() returns input type = %d", *inputType); addInput(*input, inputDesc); mpClientInterface->onAudioPortListUpdate(); return NO_ERROR; } status_t AudioPolicyManager::startInput(audio_io_handle_t input, audio_session_t session) { ALOGV("startInput() input %d", input); ssize_t index = mInputs.indexOfKey(input); if (index < 0) { ALOGW("startInput() unknown input %d", input); return BAD_VALUE; } sp inputDesc = mInputs.valueAt(index); index = inputDesc->mSessions.indexOf(session); if (index < 0) { ALOGW("startInput() unknown session %d on input %d", session, input); return BAD_VALUE; } // virtual input devices are compatible with other input devices if (!is_virtual_input_device(inputDesc->mDevice)) { // for a non-virtual input device, check if there is another (non-virtual) active input audio_io_handle_t activeInput = mInputs.getActiveInput(); if (activeInput != 0 && activeInput != input) { // If the already active input uses AUDIO_SOURCE_HOTWORD then it is closed, // otherwise the active input continues and the new input cannot be started. sp activeDesc = mInputs.valueFor(activeInput); if (activeDesc->mInputSource == AUDIO_SOURCE_HOTWORD) { ALOGW("startInput(%d) preempting low-priority input %d", input, activeInput); stopInput(activeInput, activeDesc->mSessions.itemAt(0)); releaseInput(activeInput, activeDesc->mSessions.itemAt(0)); } else { ALOGE("startInput(%d) failed: other input %d already started", input, activeInput); return INVALID_OPERATION; } } } if (inputDesc->mRefCount == 0 || mInputRoutes.hasRouteChanged(session)) { // if input maps to a dynamic policy with an activity listener, notify of state change if ((inputDesc->mPolicyMix != NULL) && ((inputDesc->mPolicyMix->mFlags & MIX_FLAG_NOTIFY_ACTIVITY) != 0)) { mpClientInterface->onDynamicPolicyMixStateUpdate(inputDesc->mPolicyMix->mRegistrationId, MIX_STATE_MIXING); } // Routing? mInputRoutes.incRouteActivity(session); if (mInputs.activeInputsCount() == 0) { SoundTrigger::setCaptureState(true); } setInputDevice(input, getNewInputDevice(input), true /* force */); // automatically enable the remote submix output when input is started if not // used by a policy mix of type MIX_TYPE_RECORDERS // For remote submix (a virtual device), we open only one input per capture request. if (audio_is_remote_submix_device(inputDesc->mDevice)) { String8 address = String8(""); if (inputDesc->mPolicyMix == NULL) { address = String8("0"); } else if (inputDesc->mPolicyMix->mMixType == MIX_TYPE_PLAYERS) { address = inputDesc->mPolicyMix->mRegistrationId; } if (address != "") { setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, AUDIO_POLICY_DEVICE_STATE_AVAILABLE, address, "remote-submix"); } } } ALOGV("AudioPolicyManager::startInput() input source = %d", inputDesc->mInputSource); inputDesc->mRefCount++; return NO_ERROR; } status_t AudioPolicyManager::stopInput(audio_io_handle_t input, audio_session_t session) { ALOGV("stopInput() input %d", input); ssize_t index = mInputs.indexOfKey(input); if (index < 0) { ALOGW("stopInput() unknown input %d", input); return BAD_VALUE; } sp inputDesc = mInputs.valueAt(index); index = inputDesc->mSessions.indexOf(session); if (index < 0) { ALOGW("stopInput() unknown session %d on input %d", session, input); return BAD_VALUE; } if (inputDesc->mRefCount == 0) { ALOGW("stopInput() input %d already stopped", input); return INVALID_OPERATION; } inputDesc->mRefCount--; // Routing? mInputRoutes.decRouteActivity(session); if (inputDesc->mRefCount == 0) { // if input maps to a dynamic policy with an activity listener, notify of state change if ((inputDesc->mPolicyMix != NULL) && ((inputDesc->mPolicyMix->mFlags & MIX_FLAG_NOTIFY_ACTIVITY) != 0)) { mpClientInterface->onDynamicPolicyMixStateUpdate(inputDesc->mPolicyMix->mRegistrationId, MIX_STATE_IDLE); } // automatically disable the remote submix output when input is stopped if not // used by a policy mix of type MIX_TYPE_RECORDERS if (audio_is_remote_submix_device(inputDesc->mDevice)) { String8 address = String8(""); if (inputDesc->mPolicyMix == NULL) { address = String8("0"); } else if (inputDesc->mPolicyMix->mMixType == MIX_TYPE_PLAYERS) { address = inputDesc->mPolicyMix->mRegistrationId; } if (address != "") { setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE, address, "remote-submix"); } } resetInputDevice(input); if (mInputs.activeInputsCount() == 0) { SoundTrigger::setCaptureState(false); } } return NO_ERROR; } void AudioPolicyManager::releaseInput(audio_io_handle_t input, audio_session_t session) { ALOGV("releaseInput() %d", input); ssize_t index = mInputs.indexOfKey(input); if (index < 0) { ALOGW("releaseInput() releasing unknown input %d", input); return; } // Routing mInputRoutes.removeRoute(session); sp inputDesc = mInputs.valueAt(index); ALOG_ASSERT(inputDesc != 0); index = inputDesc->mSessions.indexOf(session); if (index < 0) { ALOGW("releaseInput() unknown session %d on input %d", session, input); return; } inputDesc->mSessions.remove(session); if (inputDesc->mOpenRefCount == 0) { ALOGW("releaseInput() invalid open ref count %d", inputDesc->mOpenRefCount); return; } inputDesc->mOpenRefCount--; if (inputDesc->mOpenRefCount > 0) { ALOGV("releaseInput() exit > 0"); return; } closeInput(input); mpClientInterface->onAudioPortListUpdate(); ALOGV("releaseInput() exit"); } void AudioPolicyManager::closeAllInputs() { bool patchRemoved = false; for(size_t input_index = 0; input_index < mInputs.size(); input_index++) { sp inputDesc = mInputs.valueAt(input_index); ssize_t patch_index = mAudioPatches.indexOfKey(inputDesc->mPatchHandle); if (patch_index >= 0) { sp patchDesc = mAudioPatches.valueAt(patch_index); status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0); mAudioPatches.removeItemsAt(patch_index); patchRemoved = true; } mpClientInterface->closeInput(mInputs.keyAt(input_index)); } mInputs.clear(); nextAudioPortGeneration(); if (patchRemoved) { mpClientInterface->onAudioPatchListUpdate(); } } void AudioPolicyManager::initStreamVolume(audio_stream_type_t stream, int indexMin, int indexMax) { ALOGV("initStreamVolume() stream %d, min %d, max %d", stream , indexMin, indexMax); mEngine->initStreamVolume(stream, indexMin, indexMax); //FIXME: AUDIO_STREAM_ACCESSIBILITY volume follows AUDIO_STREAM_MUSIC for now if (stream == AUDIO_STREAM_MUSIC) { mEngine->initStreamVolume(AUDIO_STREAM_ACCESSIBILITY, indexMin, indexMax); } } status_t AudioPolicyManager::setStreamVolumeIndex(audio_stream_type_t stream, int index, audio_devices_t device) { if ((index < mStreams.valueFor(stream).getVolumeIndexMin()) || (index > mStreams.valueFor(stream).getVolumeIndexMax())) { return BAD_VALUE; } if (!audio_is_output_device(device)) { return BAD_VALUE; } // Force max volume if stream cannot be muted if (!mStreams.canBeMuted(stream)) index = mStreams.valueFor(stream).getVolumeIndexMax(); 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.clearCurrentVolumeIndex(stream); } mStreams.addCurrentVolumeIndex(stream, device, index); // update volume on all outputs whose current device is also selected by the same // strategy as the device specified by the caller audio_devices_t strategyDevice = getDeviceForStrategy(getStrategy(stream), true /*fromCache*/); //FIXME: AUDIO_STREAM_ACCESSIBILITY volume follows AUDIO_STREAM_MUSIC for now audio_devices_t accessibilityDevice = AUDIO_DEVICE_NONE; if (stream == AUDIO_STREAM_MUSIC) { mStreams.addCurrentVolumeIndex(AUDIO_STREAM_ACCESSIBILITY, device, index); accessibilityDevice = getDeviceForStrategy(STRATEGY_ACCESSIBILITY, true /*fromCache*/); } if ((device != AUDIO_DEVICE_OUT_DEFAULT) && (device & (strategyDevice | accessibilityDevice)) == 0) { return NO_ERROR; } status_t status = NO_ERROR; for (size_t i = 0; i < mOutputs.size(); i++) { sp desc = mOutputs.valueAt(i); audio_devices_t curDevice = Volume::getDeviceForVolume(desc->device()); if ((device == AUDIO_DEVICE_OUT_DEFAULT) || ((curDevice & strategyDevice) != 0)) { status_t volStatus = checkAndSetVolume(stream, index, desc, curDevice); if (volStatus != NO_ERROR) { status = volStatus; } } if ((device == AUDIO_DEVICE_OUT_DEFAULT) || ((curDevice & accessibilityDevice) != 0)) { status_t volStatus = checkAndSetVolume(AUDIO_STREAM_ACCESSIBILITY, index, desc, curDevice); } } return status; } status_t AudioPolicyManager::getStreamVolumeIndex(audio_stream_type_t stream, int *index, audio_devices_t device) { if (index == NULL) { return BAD_VALUE; } if (!audio_is_output_device(device)) { return BAD_VALUE; } // if device is AUDIO_DEVICE_OUT_DEFAULT, return volume for device corresponding to // the strategy the stream belongs to. if (device == AUDIO_DEVICE_OUT_DEFAULT) { device = getDeviceForStrategy(getStrategy(stream), true /*fromCache*/); } device = Volume::getDeviceForVolume(device); *index = mStreams.valueFor(stream).getVolumeIndex(device); ALOGV("getStreamVolumeIndex() stream %d device %08x index %d", stream, device, *index); return NO_ERROR; } audio_io_handle_t AudioPolicyManager::selectOutputForEffects( const SortedVector& outputs) { // select one output among several suitable for global effects. // The priority is as follows: // 1: An offloaded output. If the effect ends up not being offloadable, // AudioFlinger will invalidate the track and the offloaded output // will be closed causing the effect to be moved to a PCM output. // 2: A deep buffer output // 3: the first output in the list if (outputs.size() == 0) { return 0; } audio_io_handle_t outputOffloaded = 0; audio_io_handle_t outputDeepBuffer = 0; for (size_t i = 0; i < outputs.size(); i++) { sp desc = mOutputs.valueFor(outputs[i]); ALOGV("selectOutputForEffects outputs[%zu] flags %x", i, desc->mFlags); if ((desc->mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) { outputOffloaded = outputs[i]; } if ((desc->mFlags & AUDIO_OUTPUT_FLAG_DEEP_BUFFER) != 0) { outputDeepBuffer = outputs[i]; } } ALOGV("selectOutputForEffects outputOffloaded %d outputDeepBuffer %d", outputOffloaded, outputDeepBuffer); if (outputOffloaded != 0) { return outputOffloaded; } if (outputDeepBuffer != 0) { return outputDeepBuffer; } return outputs[0]; } audio_io_handle_t AudioPolicyManager::getOutputForEffect(const effect_descriptor_t *desc) { // apply simple rule where global effects are attached to the same output as MUSIC streams routing_strategy strategy = getStrategy(AUDIO_STREAM_MUSIC); audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/); SortedVector dstOutputs = getOutputsForDevice(device, mOutputs); audio_io_handle_t output = selectOutputForEffects(dstOutputs); ALOGV("getOutputForEffect() got output %d for fx %s flags %x", output, (desc == NULL) ? "unspecified" : desc->name, (desc == NULL) ? 0 : desc->flags); return output; } status_t AudioPolicyManager::registerEffect(const effect_descriptor_t *desc, audio_io_handle_t io, uint32_t strategy, int session, int id) { ssize_t index = mOutputs.indexOfKey(io); if (index < 0) { index = mInputs.indexOfKey(io); if (index < 0) { ALOGW("registerEffect() unknown io %d", io); return INVALID_OPERATION; } } return mEffects.registerEffect(desc, io, strategy, session, id); } bool AudioPolicyManager::isStreamActive(audio_stream_type_t stream, uint32_t inPastMs) const { return mOutputs.isStreamActive(stream, inPastMs); } bool AudioPolicyManager::isStreamActiveRemotely(audio_stream_type_t stream, uint32_t inPastMs) const { return mOutputs.isStreamActiveRemotely(stream, inPastMs); } bool AudioPolicyManager::isSourceActive(audio_source_t source) const { for (size_t i = 0; i < mInputs.size(); i++) { const sp inputDescriptor = mInputs.valueAt(i); if (inputDescriptor->mRefCount == 0) { continue; } if (inputDescriptor->mInputSource == (int)source) { return true; } // AUDIO_SOURCE_HOTWORD is equivalent to AUDIO_SOURCE_VOICE_RECOGNITION only if it // corresponds to an active capture triggered by a hardware hotword recognition if ((source == AUDIO_SOURCE_VOICE_RECOGNITION) && (inputDescriptor->mInputSource == AUDIO_SOURCE_HOTWORD)) { // FIXME: we should not assume that the first session is the active one and keep // activity count per session. Same in startInput(). ssize_t index = mSoundTriggerSessions.indexOfKey(inputDescriptor->mSessions.itemAt(0)); if (index >= 0) { return true; } } } return false; } // Register a list of custom mixes with their attributes and format. // When a mix is registered, corresponding input and output profiles are // added to the remote submix hw module. The profile contains only the // parameters (sampling rate, format...) specified by the mix. // The corresponding input remote submix device is also connected. // // When a remote submix device is connected, the address is checked to select the // appropriate profile and the corresponding input or output stream is opened. // // When capture starts, getInputForAttr() will: // - 1 look for a mix matching the address passed in attribtutes tags if any // - 2 if none found, getDeviceForInputSource() will: // - 2.1 look for a mix matching the attributes source // - 2.2 if none found, default to device selection by policy rules // At this time, the corresponding output remote submix device is also connected // and active playback use cases can be transferred to this mix if needed when reconnecting // after AudioTracks are invalidated // // When playback starts, getOutputForAttr() will: // - 1 look for a mix matching the address passed in attribtutes tags if any // - 2 if none found, look for a mix matching the attributes usage // - 3 if none found, default to device and output selection by policy rules. status_t AudioPolicyManager::registerPolicyMixes(Vector mixes) { sp module; for (size_t i = 0; i < mHwModules.size(); i++) { if (strcmp(AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX, mHwModules[i]->mName) == 0 && mHwModules[i]->mHandle != 0) { module = mHwModules[i]; break; } } if (module == 0) { return INVALID_OPERATION; } ALOGV("registerPolicyMixes() num mixes %d", mixes.size()); for (size_t i = 0; i < mixes.size(); i++) { String8 address = mixes[i].mRegistrationId; if (mPolicyMixes.registerMix(address, mixes[i]) != NO_ERROR) { continue; } audio_config_t outputConfig = mixes[i].mFormat; audio_config_t inputConfig = mixes[i].mFormat; // NOTE: audio flinger mixer does not support mono output: configure remote submix HAL in // stereo and let audio flinger do the channel conversion if needed. outputConfig.channel_mask = AUDIO_CHANNEL_OUT_STEREO; inputConfig.channel_mask = AUDIO_CHANNEL_IN_STEREO; module->addOutputProfile(address, &outputConfig, AUDIO_DEVICE_OUT_REMOTE_SUBMIX, address); module->addInputProfile(address, &inputConfig, AUDIO_DEVICE_IN_REMOTE_SUBMIX, address); if (mixes[i].mMixType == MIX_TYPE_PLAYERS) { setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX, AUDIO_POLICY_DEVICE_STATE_AVAILABLE, address.string(), "remote-submix"); } else { setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, AUDIO_POLICY_DEVICE_STATE_AVAILABLE, address.string(), "remote-submix"); } } return NO_ERROR; } status_t AudioPolicyManager::unregisterPolicyMixes(Vector mixes) { sp module; for (size_t i = 0; i < mHwModules.size(); i++) { if (strcmp(AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX, mHwModules[i]->mName) == 0 && mHwModules[i]->mHandle != 0) { module = mHwModules[i]; break; } } if (module == 0) { return INVALID_OPERATION; } ALOGV("unregisterPolicyMixes() num mixes %d", mixes.size()); for (size_t i = 0; i < mixes.size(); i++) { String8 address = mixes[i].mRegistrationId; if (mPolicyMixes.unregisterMix(address) != NO_ERROR) { continue; } if (getDeviceConnectionState(AUDIO_DEVICE_IN_REMOTE_SUBMIX, address.string()) == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) { setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX, AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE, address.string(), "remote-submix"); } if (getDeviceConnectionState(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, address.string()) == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) { setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE, address.string(), "remote-submix"); } module->removeOutputProfile(address); module->removeInputProfile(address); } return NO_ERROR; } status_t AudioPolicyManager::dump(int fd) { const size_t SIZE = 256; char buffer[SIZE]; String8 result; snprintf(buffer, SIZE, "\nAudioPolicyManager Dump: %p\n", this); result.append(buffer); snprintf(buffer, SIZE, " Primary Output: %d\n", mPrimaryOutput->mIoHandle); result.append(buffer); snprintf(buffer, SIZE, " Phone state: %d\n", mEngine->getPhoneState()); result.append(buffer); snprintf(buffer, SIZE, " Force use for communications %d\n", mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_COMMUNICATION)); result.append(buffer); snprintf(buffer, SIZE, " Force use for media %d\n", mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_MEDIA)); result.append(buffer); snprintf(buffer, SIZE, " Force use for record %d\n", mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_RECORD)); result.append(buffer); snprintf(buffer, SIZE, " Force use for dock %d\n", mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_DOCK)); result.append(buffer); snprintf(buffer, SIZE, " Force use for system %d\n", mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM)); result.append(buffer); snprintf(buffer, SIZE, " Force use for hdmi system audio %d\n", mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_HDMI_SYSTEM_AUDIO)); result.append(buffer); write(fd, result.string(), result.size()); mAvailableOutputDevices.dump(fd, String8("output")); mAvailableInputDevices.dump(fd, String8("input")); mHwModules.dump(fd); mOutputs.dump(fd); mInputs.dump(fd); mStreams.dump(fd); mEffects.dump(fd); mAudioPatches.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 AudioPolicyManager::isOffloadSupported(const audio_offload_info_t& offloadInfo) { ALOGV("isOffloadSupported: SR=%u, CM=0x%x, Format=0x%x, StreamType=%d," " BitRate=%u, duration=%" PRId64 " us, has_video=%d", offloadInfo.sample_rate, offloadInfo.channel_mask, offloadInfo.format, offloadInfo.stream_type, offloadInfo.bit_rate, offloadInfo.duration_us, offloadInfo.has_video); // Check if offload has been disabled char propValue[PROPERTY_VALUE_MAX]; if (property_get("audio.offload.disable", propValue, "0")) { if (atoi(propValue) != 0) { ALOGV("offload disabled by audio.offload.disable=%s", propValue ); return false; } } // Check if stream type is music, then only allow offload as of now. if (offloadInfo.stream_type != AUDIO_STREAM_MUSIC) { ALOGV("isOffloadSupported: stream_type != MUSIC, returning false"); return false; } //TODO: enable audio offloading with video when ready if (offloadInfo.has_video) { ALOGV("isOffloadSupported: has_video == true, returning false"); return false; } //If duration is less than minimum value defined in property, return false if (property_get("audio.offload.min.duration.secs", propValue, NULL)) { if (offloadInfo.duration_us < (atoi(propValue) * 1000000 )) { ALOGV("Offload denied by duration < audio.offload.min.duration.secs(=%s)", propValue); return false; } } else if (offloadInfo.duration_us < OFFLOAD_DEFAULT_MIN_DURATION_SECS * 1000000) { ALOGV("Offload denied by duration < default min(=%u)", OFFLOAD_DEFAULT_MIN_DURATION_SECS); return false; } // Do not allow offloading if one non offloadable effect is enabled. This prevents from // creating an offloaded track and tearing it down immediately after start when audioflinger // detects there is an active non offloadable effect. // FIXME: We should check the audio session here but we do not have it in this context. // This may prevent offloading in rare situations where effects are left active by apps // in the background. if (mEffects.isNonOffloadableEffectEnabled()) { return false; } // See if there is a profile to support this. // AUDIO_DEVICE_NONE sp profile = getProfileForDirectOutput(AUDIO_DEVICE_NONE /*ignore device */, offloadInfo.sample_rate, offloadInfo.format, offloadInfo.channel_mask, AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD); ALOGV("isOffloadSupported() profile %sfound", profile != 0 ? "" : "NOT "); return (profile != 0); } status_t AudioPolicyManager::listAudioPorts(audio_port_role_t role, audio_port_type_t type, unsigned int *num_ports, struct audio_port *ports, unsigned int *generation) { if (num_ports == NULL || (*num_ports != 0 && ports == NULL) || generation == NULL) { return BAD_VALUE; } ALOGV("listAudioPorts() role %d type %d num_ports %d ports %p", role, type, *num_ports, ports); if (ports == NULL) { *num_ports = 0; } size_t portsWritten = 0; size_t portsMax = *num_ports; *num_ports = 0; if (type == AUDIO_PORT_TYPE_NONE || type == AUDIO_PORT_TYPE_DEVICE) { if (role == AUDIO_PORT_ROLE_SINK || role == AUDIO_PORT_ROLE_NONE) { for (size_t i = 0; i < mAvailableOutputDevices.size() && portsWritten < portsMax; i++) { mAvailableOutputDevices[i]->toAudioPort(&ports[portsWritten++]); } *num_ports += mAvailableOutputDevices.size(); } if (role == AUDIO_PORT_ROLE_SOURCE || role == AUDIO_PORT_ROLE_NONE) { for (size_t i = 0; i < mAvailableInputDevices.size() && portsWritten < portsMax; i++) { mAvailableInputDevices[i]->toAudioPort(&ports[portsWritten++]); } *num_ports += mAvailableInputDevices.size(); } } if (type == AUDIO_PORT_TYPE_NONE || type == AUDIO_PORT_TYPE_MIX) { if (role == AUDIO_PORT_ROLE_SINK || role == AUDIO_PORT_ROLE_NONE) { for (size_t i = 0; i < mInputs.size() && portsWritten < portsMax; i++) { mInputs[i]->toAudioPort(&ports[portsWritten++]); } *num_ports += mInputs.size(); } if (role == AUDIO_PORT_ROLE_SOURCE || role == AUDIO_PORT_ROLE_NONE) { size_t numOutputs = 0; for (size_t i = 0; i < mOutputs.size(); i++) { if (!mOutputs[i]->isDuplicated()) { numOutputs++; if (portsWritten < portsMax) { mOutputs[i]->toAudioPort(&ports[portsWritten++]); } } } *num_ports += numOutputs; } } *generation = curAudioPortGeneration(); ALOGV("listAudioPorts() got %zu ports needed %d", portsWritten, *num_ports); return NO_ERROR; } status_t AudioPolicyManager::getAudioPort(struct audio_port *port __unused) { return NO_ERROR; } status_t AudioPolicyManager::createAudioPatch(const struct audio_patch *patch, audio_patch_handle_t *handle, uid_t uid) { ALOGV("createAudioPatch()"); if (handle == NULL || patch == NULL) { return BAD_VALUE; } ALOGV("createAudioPatch() num sources %d num sinks %d", patch->num_sources, patch->num_sinks); if (patch->num_sources == 0 || patch->num_sources > AUDIO_PATCH_PORTS_MAX || patch->num_sinks == 0 || patch->num_sinks > AUDIO_PATCH_PORTS_MAX) { return BAD_VALUE; } // only one source per audio patch supported for now if (patch->num_sources > 1) { return INVALID_OPERATION; } if (patch->sources[0].role != AUDIO_PORT_ROLE_SOURCE) { return INVALID_OPERATION; } for (size_t i = 0; i < patch->num_sinks; i++) { if (patch->sinks[i].role != AUDIO_PORT_ROLE_SINK) { return INVALID_OPERATION; } } sp patchDesc; ssize_t index = mAudioPatches.indexOfKey(*handle); ALOGV("createAudioPatch source id %d role %d type %d", patch->sources[0].id, patch->sources[0].role, patch->sources[0].type); #if LOG_NDEBUG == 0 for (size_t i = 0; i < patch->num_sinks; i++) { ALOGV("createAudioPatch sink %d: id %d role %d type %d", i, patch->sinks[i].id, patch->sinks[i].role, patch->sinks[i].type); } #endif if (index >= 0) { patchDesc = mAudioPatches.valueAt(index); ALOGV("createAudioPatch() mUidCached %d patchDesc->mUid %d uid %d", mUidCached, patchDesc->mUid, uid); if (patchDesc->mUid != mUidCached && uid != patchDesc->mUid) { return INVALID_OPERATION; } } else { *handle = 0; } if (patch->sources[0].type == AUDIO_PORT_TYPE_MIX) { sp outputDesc = mOutputs.getOutputFromId(patch->sources[0].id); if (outputDesc == NULL) { ALOGV("createAudioPatch() output not found for id %d", patch->sources[0].id); return BAD_VALUE; } ALOG_ASSERT(!outputDesc->isDuplicated(),"duplicated output %d in source in ports", outputDesc->mIoHandle); if (patchDesc != 0) { if (patchDesc->mPatch.sources[0].id != patch->sources[0].id) { ALOGV("createAudioPatch() source id differs for patch current id %d new id %d", patchDesc->mPatch.sources[0].id, patch->sources[0].id); return BAD_VALUE; } } DeviceVector devices; for (size_t i = 0; i < patch->num_sinks; i++) { // Only support mix to devices connection // TODO add support for mix to mix connection if (patch->sinks[i].type != AUDIO_PORT_TYPE_DEVICE) { ALOGV("createAudioPatch() source mix but sink is not a device"); return INVALID_OPERATION; } sp devDesc = mAvailableOutputDevices.getDeviceFromId(patch->sinks[i].id); if (devDesc == 0) { ALOGV("createAudioPatch() out device not found for id %d", patch->sinks[i].id); return BAD_VALUE; } if (!outputDesc->mProfile->isCompatibleProfile(devDesc->type(), devDesc->mAddress, patch->sources[0].sample_rate, NULL, // updatedSamplingRate patch->sources[0].format, NULL, // updatedFormat patch->sources[0].channel_mask, NULL, // updatedChannelMask AUDIO_OUTPUT_FLAG_NONE /*FIXME*/)) { ALOGV("createAudioPatch() profile not supported for device %08x", devDesc->type()); return INVALID_OPERATION; } devices.add(devDesc); } if (devices.size() == 0) { return INVALID_OPERATION; } // TODO: reconfigure output format and channels here ALOGV("createAudioPatch() setting device %08x on output %d", devices.types(), outputDesc->mIoHandle); setOutputDevice(outputDesc, devices.types(), true, 0, handle); index = mAudioPatches.indexOfKey(*handle); if (index >= 0) { if (patchDesc != 0 && patchDesc != mAudioPatches.valueAt(index)) { ALOGW("createAudioPatch() setOutputDevice() did not reuse the patch provided"); } patchDesc = mAudioPatches.valueAt(index); patchDesc->mUid = uid; ALOGV("createAudioPatch() success"); } else { ALOGW("createAudioPatch() setOutputDevice() failed to create a patch"); return INVALID_OPERATION; } } else if (patch->sources[0].type == AUDIO_PORT_TYPE_DEVICE) { if (patch->sinks[0].type == AUDIO_PORT_TYPE_MIX) { // input device to input mix connection // only one sink supported when connecting an input device to a mix if (patch->num_sinks > 1) { return INVALID_OPERATION; } sp inputDesc = mInputs.getInputFromId(patch->sinks[0].id); if (inputDesc == NULL) { return BAD_VALUE; } if (patchDesc != 0) { if (patchDesc->mPatch.sinks[0].id != patch->sinks[0].id) { return BAD_VALUE; } } sp devDesc = mAvailableInputDevices.getDeviceFromId(patch->sources[0].id); if (devDesc == 0) { return BAD_VALUE; } if (!inputDesc->mProfile->isCompatibleProfile(devDesc->type(), devDesc->mAddress, patch->sinks[0].sample_rate, NULL, /*updatedSampleRate*/ patch->sinks[0].format, NULL, /*updatedFormat*/ patch->sinks[0].channel_mask, NULL, /*updatedChannelMask*/ // FIXME for the parameter type, // and the NONE (audio_output_flags_t) AUDIO_INPUT_FLAG_NONE)) { return INVALID_OPERATION; } // TODO: reconfigure output format and channels here ALOGV("createAudioPatch() setting device %08x on output %d", devDesc->type(), inputDesc->mIoHandle); setInputDevice(inputDesc->mIoHandle, devDesc->type(), true, handle); index = mAudioPatches.indexOfKey(*handle); if (index >= 0) { if (patchDesc != 0 && patchDesc != mAudioPatches.valueAt(index)) { ALOGW("createAudioPatch() setInputDevice() did not reuse the patch provided"); } patchDesc = mAudioPatches.valueAt(index); patchDesc->mUid = uid; ALOGV("createAudioPatch() success"); } else { ALOGW("createAudioPatch() setInputDevice() failed to create a patch"); return INVALID_OPERATION; } } else if (patch->sinks[0].type == AUDIO_PORT_TYPE_DEVICE) { // device to device connection if (patchDesc != 0) { if (patchDesc->mPatch.sources[0].id != patch->sources[0].id) { return BAD_VALUE; } } sp srcDeviceDesc = mAvailableInputDevices.getDeviceFromId(patch->sources[0].id); if (srcDeviceDesc == 0) { return BAD_VALUE; } //update source and sink with our own data as the data passed in the patch may // be incomplete. struct audio_patch newPatch = *patch; srcDeviceDesc->toAudioPortConfig(&newPatch.sources[0], &patch->sources[0]); for (size_t i = 0; i < patch->num_sinks; i++) { if (patch->sinks[i].type != AUDIO_PORT_TYPE_DEVICE) { ALOGV("createAudioPatch() source device but one sink is not a device"); return INVALID_OPERATION; } sp sinkDeviceDesc = mAvailableOutputDevices.getDeviceFromId(patch->sinks[i].id); if (sinkDeviceDesc == 0) { return BAD_VALUE; } sinkDeviceDesc->toAudioPortConfig(&newPatch.sinks[i], &patch->sinks[i]); // create a software bridge in PatchPanel if: // - source and sink devices are on differnt HW modules OR // - audio HAL version is < 3.0 if ((srcDeviceDesc->getModuleHandle() != sinkDeviceDesc->getModuleHandle()) || (srcDeviceDesc->mModule->mHalVersion < AUDIO_DEVICE_API_VERSION_3_0)) { // support only one sink device for now to simplify output selection logic if (patch->num_sinks > 1) { return INVALID_OPERATION; } SortedVector outputs = getOutputsForDevice(sinkDeviceDesc->type(), mOutputs); // if the sink device is reachable via an opened output stream, request to go via // this output stream by adding a second source to the patch description audio_io_handle_t output = selectOutput(outputs, AUDIO_OUTPUT_FLAG_NONE, AUDIO_FORMAT_INVALID); if (output != AUDIO_IO_HANDLE_NONE) { sp outputDesc = mOutputs.valueFor(output); if (outputDesc->isDuplicated()) { return INVALID_OPERATION; } outputDesc->toAudioPortConfig(&newPatch.sources[1], &patch->sources[0]); newPatch.sources[1].ext.mix.usecase.stream = AUDIO_STREAM_PATCH; newPatch.num_sources = 2; } } } // TODO: check from routing capabilities in config file and other conflicting patches audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE; if (index >= 0) { afPatchHandle = patchDesc->mAfPatchHandle; } status_t status = mpClientInterface->createAudioPatch(&newPatch, &afPatchHandle, 0); ALOGV("createAudioPatch() patch panel returned %d patchHandle %d", status, afPatchHandle); if (status == NO_ERROR) { if (index < 0) { patchDesc = new AudioPatch(&newPatch, uid); addAudioPatch(patchDesc->mHandle, patchDesc); } else { patchDesc->mPatch = newPatch; } patchDesc->mAfPatchHandle = afPatchHandle; *handle = patchDesc->mHandle; nextAudioPortGeneration(); mpClientInterface->onAudioPatchListUpdate(); } else { ALOGW("createAudioPatch() patch panel could not connect device patch, error %d", status); return INVALID_OPERATION; } } else { return BAD_VALUE; } } else { return BAD_VALUE; } return NO_ERROR; } status_t AudioPolicyManager::releaseAudioPatch(audio_patch_handle_t handle, uid_t uid) { ALOGV("releaseAudioPatch() patch %d", handle); ssize_t index = mAudioPatches.indexOfKey(handle); if (index < 0) { return BAD_VALUE; } sp patchDesc = mAudioPatches.valueAt(index); ALOGV("releaseAudioPatch() mUidCached %d patchDesc->mUid %d uid %d", mUidCached, patchDesc->mUid, uid); if (patchDesc->mUid != mUidCached && uid != patchDesc->mUid) { return INVALID_OPERATION; } struct audio_patch *patch = &patchDesc->mPatch; patchDesc->mUid = mUidCached; if (patch->sources[0].type == AUDIO_PORT_TYPE_MIX) { sp outputDesc = mOutputs.getOutputFromId(patch->sources[0].id); if (outputDesc == NULL) { ALOGV("releaseAudioPatch() output not found for id %d", patch->sources[0].id); return BAD_VALUE; } setOutputDevice(outputDesc, getNewOutputDevice(outputDesc, true /*fromCache*/), true, 0, NULL); } else if (patch->sources[0].type == AUDIO_PORT_TYPE_DEVICE) { if (patch->sinks[0].type == AUDIO_PORT_TYPE_MIX) { sp inputDesc = mInputs.getInputFromId(patch->sinks[0].id); if (inputDesc == NULL) { ALOGV("releaseAudioPatch() input not found for id %d", patch->sinks[0].id); return BAD_VALUE; } setInputDevice(inputDesc->mIoHandle, getNewInputDevice(inputDesc->mIoHandle), true, NULL); } else if (patch->sinks[0].type == AUDIO_PORT_TYPE_DEVICE) { audio_patch_handle_t afPatchHandle = patchDesc->mAfPatchHandle; status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0); ALOGV("releaseAudioPatch() patch panel returned %d patchHandle %d", status, patchDesc->mAfPatchHandle); removeAudioPatch(patchDesc->mHandle); nextAudioPortGeneration(); mpClientInterface->onAudioPatchListUpdate(); } else { return BAD_VALUE; } } else { return BAD_VALUE; } return NO_ERROR; } status_t AudioPolicyManager::listAudioPatches(unsigned int *num_patches, struct audio_patch *patches, unsigned int *generation) { if (generation == NULL) { return BAD_VALUE; } *generation = curAudioPortGeneration(); return mAudioPatches.listAudioPatches(num_patches, patches); } status_t AudioPolicyManager::setAudioPortConfig(const struct audio_port_config *config) { ALOGV("setAudioPortConfig()"); if (config == NULL) { return BAD_VALUE; } ALOGV("setAudioPortConfig() on port handle %d", config->id); // Only support gain configuration for now if (config->config_mask != AUDIO_PORT_CONFIG_GAIN) { return INVALID_OPERATION; } sp audioPortConfig; if (config->type == AUDIO_PORT_TYPE_MIX) { if (config->role == AUDIO_PORT_ROLE_SOURCE) { sp outputDesc = mOutputs.getOutputFromId(config->id); if (outputDesc == NULL) { return BAD_VALUE; } ALOG_ASSERT(!outputDesc->isDuplicated(), "setAudioPortConfig() called on duplicated output %d", outputDesc->mIoHandle); audioPortConfig = outputDesc; } else if (config->role == AUDIO_PORT_ROLE_SINK) { sp inputDesc = mInputs.getInputFromId(config->id); if (inputDesc == NULL) { return BAD_VALUE; } audioPortConfig = inputDesc; } else { return BAD_VALUE; } } else if (config->type == AUDIO_PORT_TYPE_DEVICE) { sp deviceDesc; if (config->role == AUDIO_PORT_ROLE_SOURCE) { deviceDesc = mAvailableInputDevices.getDeviceFromId(config->id); } else if (config->role == AUDIO_PORT_ROLE_SINK) { deviceDesc = mAvailableOutputDevices.getDeviceFromId(config->id); } else { return BAD_VALUE; } if (deviceDesc == NULL) { return BAD_VALUE; } audioPortConfig = deviceDesc; } else { return BAD_VALUE; } struct audio_port_config backupConfig; status_t status = audioPortConfig->applyAudioPortConfig(config, &backupConfig); if (status == NO_ERROR) { struct audio_port_config newConfig; audioPortConfig->toAudioPortConfig(&newConfig, config); status = mpClientInterface->setAudioPortConfig(&newConfig, 0); } if (status != NO_ERROR) { audioPortConfig->applyAudioPortConfig(&backupConfig); } return status; } void AudioPolicyManager::clearAudioPatches(uid_t uid) { for (ssize_t i = (ssize_t)mAudioPatches.size() - 1; i >= 0; i--) { sp patchDesc = mAudioPatches.valueAt(i); if (patchDesc->mUid == uid) { releaseAudioPatch(mAudioPatches.keyAt(i), uid); } } } status_t AudioPolicyManager::acquireSoundTriggerSession(audio_session_t *session, audio_io_handle_t *ioHandle, audio_devices_t *device) { *session = (audio_session_t)mpClientInterface->newAudioUniqueId(); *ioHandle = (audio_io_handle_t)mpClientInterface->newAudioUniqueId(); *device = getDeviceAndMixForInputSource(AUDIO_SOURCE_HOTWORD); return mSoundTriggerSessions.acquireSession(*session, *ioHandle); } status_t AudioPolicyManager::startAudioSource(const struct audio_port_config *source __unused, const audio_attributes_t *attributes __unused, audio_io_handle_t *handle __unused) { return INVALID_OPERATION; } status_t AudioPolicyManager::stopAudioSource(audio_io_handle_t handle __unused) { return INVALID_OPERATION; } // ---------------------------------------------------------------------------- // AudioPolicyManager // ---------------------------------------------------------------------------- uint32_t AudioPolicyManager::nextAudioPortGeneration() { return android_atomic_inc(&mAudioPortGeneration); } AudioPolicyManager::AudioPolicyManager(AudioPolicyClientInterface *clientInterface) : #ifdef AUDIO_POLICY_TEST Thread(false), #endif //AUDIO_POLICY_TEST mLimitRingtoneVolume(false), mLastVoiceVolume(-1.0f), mA2dpSuspended(false), mSpeakerDrcEnabled(false), mAudioPortGeneration(1), mBeaconMuteRefCount(0), mBeaconPlayingRefCount(0), mBeaconMuted(false) { audio_policy::EngineInstance *engineInstance = audio_policy::EngineInstance::getInstance(); if (!engineInstance) { ALOGE("%s: Could not get an instance of policy engine", __FUNCTION__); return; } // Retrieve the Policy Manager Interface mEngine = engineInstance->queryInterface(); if (mEngine == NULL) { ALOGE("%s: Failed to get Policy Engine Interface", __FUNCTION__); return; } mEngine->setObserver(this); status_t status = mEngine->initCheck(); ALOG_ASSERT(status == NO_ERROR, "Policy engine not initialized(err=%d)", status); mUidCached = getuid(); mpClientInterface = clientInterface; mDefaultOutputDevice = new DeviceDescriptor(String8("Speaker"), AUDIO_DEVICE_OUT_SPEAKER); if (ConfigParsingUtils::loadAudioPolicyConfig(AUDIO_POLICY_VENDOR_CONFIG_FILE, mHwModules, mAvailableInputDevices, mAvailableOutputDevices, mDefaultOutputDevice, mSpeakerDrcEnabled) != NO_ERROR) { if (ConfigParsingUtils::loadAudioPolicyConfig(AUDIO_POLICY_CONFIG_FILE, mHwModules, mAvailableInputDevices, mAvailableOutputDevices, mDefaultOutputDevice, mSpeakerDrcEnabled) != NO_ERROR) { ALOGE("could not load audio policy configuration file, setting defaults"); defaultAudioPolicyConfig(); } } // mAvailableOutputDevices and mAvailableInputDevices now contain all attached devices // must be done after reading the policy (since conditionned by Speaker Drc Enabling) mEngine->initializeVolumeCurves(mSpeakerDrcEnabled); // open all output streams needed to access attached devices audio_devices_t outputDeviceTypes = mAvailableOutputDevices.types(); audio_devices_t inputDeviceTypes = mAvailableInputDevices.types() & ~AUDIO_DEVICE_BIT_IN; for (size_t i = 0; i < mHwModules.size(); i++) { mHwModules[i]->mHandle = mpClientInterface->loadHwModule(mHwModules[i]->mName); if (mHwModules[i]->mHandle == 0) { ALOGW("could not open HW module %s", mHwModules[i]->mName); continue; } // open all output streams needed to access attached devices // except for direct output streams that are only opened when they are actually // required by an app. // This also validates mAvailableOutputDevices list for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++) { const sp outProfile = mHwModules[i]->mOutputProfiles[j]; if (outProfile->mSupportedDevices.isEmpty()) { ALOGW("Output profile contains no device on module %s", mHwModules[i]->mName); continue; } if ((outProfile->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) { continue; } audio_devices_t profileType = outProfile->mSupportedDevices.types(); if ((profileType & mDefaultOutputDevice->type()) != AUDIO_DEVICE_NONE) { profileType = mDefaultOutputDevice->type(); } else { // chose first device present in mSupportedDevices also part of // outputDeviceTypes for (size_t k = 0; k < outProfile->mSupportedDevices.size(); k++) { profileType = outProfile->mSupportedDevices[k]->type(); if ((profileType & outputDeviceTypes) != 0) { break; } } } if ((profileType & outputDeviceTypes) == 0) { continue; } sp outputDesc = new SwAudioOutputDescriptor(outProfile, mpClientInterface); outputDesc->mDevice = profileType; audio_config_t config = AUDIO_CONFIG_INITIALIZER; config.sample_rate = outputDesc->mSamplingRate; config.channel_mask = outputDesc->mChannelMask; config.format = outputDesc->mFormat; audio_io_handle_t output = AUDIO_IO_HANDLE_NONE; status_t status = mpClientInterface->openOutput(outProfile->getModuleHandle(), &output, &config, &outputDesc->mDevice, String8(""), &outputDesc->mLatency, outputDesc->mFlags); if (status != NO_ERROR) { ALOGW("Cannot open output stream for device %08x on hw module %s", outputDesc->mDevice, mHwModules[i]->mName); } else { outputDesc->mSamplingRate = config.sample_rate; outputDesc->mChannelMask = config.channel_mask; outputDesc->mFormat = config.format; for (size_t k = 0; k < outProfile->mSupportedDevices.size(); k++) { audio_devices_t type = outProfile->mSupportedDevices[k]->type(); ssize_t index = mAvailableOutputDevices.indexOf(outProfile->mSupportedDevices[k]); // give a valid ID to an attached device once confirmed it is reachable if (index >= 0 && !mAvailableOutputDevices[index]->isAttached()) { mAvailableOutputDevices[index]->attach(mHwModules[i]); } } if (mPrimaryOutput == 0 && outProfile->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY) { mPrimaryOutput = outputDesc; } addOutput(output, outputDesc); setOutputDevice(outputDesc, outputDesc->mDevice, true); } } // open input streams needed to access attached devices to validate // mAvailableInputDevices list for (size_t j = 0; j < mHwModules[i]->mInputProfiles.size(); j++) { const sp inProfile = mHwModules[i]->mInputProfiles[j]; if (inProfile->mSupportedDevices.isEmpty()) { ALOGW("Input profile contains no device on module %s", mHwModules[i]->mName); continue; } // chose first device present in mSupportedDevices also part of // inputDeviceTypes audio_devices_t profileType = AUDIO_DEVICE_NONE; for (size_t k = 0; k < inProfile->mSupportedDevices.size(); k++) { profileType = inProfile->mSupportedDevices[k]->type(); if (profileType & inputDeviceTypes) { break; } } if ((profileType & inputDeviceTypes) == 0) { continue; } sp inputDesc = new AudioInputDescriptor(inProfile); inputDesc->mInputSource = AUDIO_SOURCE_MIC; inputDesc->mDevice = profileType; // find the address DeviceVector inputDevices = mAvailableInputDevices.getDevicesFromType(profileType); // the inputs vector must be of size 1, but we don't want to crash here String8 address = inputDevices.size() > 0 ? inputDevices.itemAt(0)->mAddress : String8(""); ALOGV(" for input device 0x%x using address %s", profileType, address.string()); ALOGE_IF(inputDevices.size() == 0, "Input device list is empty!"); audio_config_t config = AUDIO_CONFIG_INITIALIZER; config.sample_rate = inputDesc->mSamplingRate; config.channel_mask = inputDesc->mChannelMask; config.format = inputDesc->mFormat; audio_io_handle_t input = AUDIO_IO_HANDLE_NONE; status_t status = mpClientInterface->openInput(inProfile->getModuleHandle(), &input, &config, &inputDesc->mDevice, address, AUDIO_SOURCE_MIC, AUDIO_INPUT_FLAG_NONE); if (status == NO_ERROR) { for (size_t k = 0; k < inProfile->mSupportedDevices.size(); k++) { audio_devices_t type = inProfile->mSupportedDevices[k]->type(); ssize_t index = mAvailableInputDevices.indexOf(inProfile->mSupportedDevices[k]); // give a valid ID to an attached device once confirmed it is reachable if (index >= 0 && !mAvailableInputDevices[index]->isAttached()) { mAvailableInputDevices[index]->attach(mHwModules[i]); } } mpClientInterface->closeInput(input); } else { ALOGW("Cannot open input stream for device %08x on hw module %s", inputDesc->mDevice, mHwModules[i]->mName); } } } // make sure all attached devices have been allocated a unique ID for (size_t i = 0; i < mAvailableOutputDevices.size();) { if (!mAvailableOutputDevices[i]->isAttached()) { ALOGW("Input device %08x unreachable", mAvailableOutputDevices[i]->type()); mAvailableOutputDevices.remove(mAvailableOutputDevices[i]); continue; } // The device is now validated and can be appended to the available devices of the engine mEngine->setDeviceConnectionState(mAvailableOutputDevices[i], AUDIO_POLICY_DEVICE_STATE_AVAILABLE); i++; } for (size_t i = 0; i < mAvailableInputDevices.size();) { if (!mAvailableInputDevices[i]->isAttached()) { ALOGW("Input device %08x unreachable", mAvailableInputDevices[i]->type()); mAvailableInputDevices.remove(mAvailableInputDevices[i]); continue; } // The device is now validated and can be appended to the available devices of the engine mEngine->setDeviceConnectionState(mAvailableInputDevices[i], AUDIO_POLICY_DEVICE_STATE_AVAILABLE); i++; } // make sure default device is reachable if (mAvailableOutputDevices.indexOf(mDefaultOutputDevice) < 0) { ALOGE("Default device %08x is unreachable", mDefaultOutputDevice->type()); } 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->mIoHandle, outputCmd.toString()); mTestDevice = AUDIO_DEVICE_OUT_SPEAKER; mTestSamplingRate = 44100; mTestFormat = AUDIO_FORMAT_PCM_16_BIT; mTestChannels = AUDIO_CHANNEL_OUT_STEREO; mTestLatencyMs = 0; mCurOutput = 0; mDirectOutput = false; for (int i = 0; i < NUM_TEST_OUTPUTS; i++) { mTestOutputs[i] = 0; } const size_t SIZE = 256; char buffer[SIZE]; snprintf(buffer, SIZE, "AudioPolicyManagerTest"); run(buffer, ANDROID_PRIORITY_AUDIO); } #endif //AUDIO_POLICY_TEST } AudioPolicyManager::~AudioPolicyManager() { #ifdef AUDIO_POLICY_TEST exit(); #endif //AUDIO_POLICY_TEST for (size_t i = 0; i < mOutputs.size(); i++) { mpClientInterface->closeOutput(mOutputs.keyAt(i)); } for (size_t i = 0; i < mInputs.size(); i++) { mpClientInterface->closeInput(mInputs.keyAt(i)); } mAvailableOutputDevices.clear(); mAvailableInputDevices.clear(); mOutputs.clear(); mInputs.clear(); mHwModules.clear(); } status_t AudioPolicyManager::initCheck() { return (mPrimaryOutput == 0) ? NO_INIT : NO_ERROR; } #ifdef AUDIO_POLICY_TEST bool AudioPolicyManager::threadLoop() { ALOGV("entering threadLoop()"); while (!exitPending()) { String8 command; int valueInt; String8 value; Mutex::Autolock _l(mLock); mWaitWorkCV.waitRelative(mLock, milliseconds(50)); command = mpClientInterface->getParameters(0, String8("test_cmd_policy")); AudioParameter param = AudioParameter(command); if (param.getInt(String8("test_cmd_policy"), valueInt) == NO_ERROR && valueInt != 0) { ALOGV("Test command %s received", command.string()); String8 target; if (param.get(String8("target"), target) != NO_ERROR) { target = "Manager"; } if (param.getInt(String8("test_cmd_policy_output"), valueInt) == NO_ERROR) { param.remove(String8("test_cmd_policy_output")); mCurOutput = valueInt; } if (param.get(String8("test_cmd_policy_direct"), value) == NO_ERROR) { param.remove(String8("test_cmd_policy_direct")); if (value == "false") { mDirectOutput = false; } else if (value == "true") { mDirectOutput = true; } } if (param.getInt(String8("test_cmd_policy_input"), valueInt) == NO_ERROR) { param.remove(String8("test_cmd_policy_input")); mTestInput = valueInt; } if (param.get(String8("test_cmd_policy_format"), value) == NO_ERROR) { param.remove(String8("test_cmd_policy_format")); int format = AUDIO_FORMAT_INVALID; if (value == "PCM 16 bits") { format = AUDIO_FORMAT_PCM_16_BIT; } else if (value == "PCM 8 bits") { format = AUDIO_FORMAT_PCM_8_BIT; } else if (value == "Compressed MP3") { format = AUDIO_FORMAT_MP3; } if (format != AUDIO_FORMAT_INVALID) { if (target == "Manager") { mTestFormat = format; } else if (mTestOutputs[mCurOutput] != 0) { AudioParameter outputParam = AudioParameter(); outputParam.addInt(String8("format"), format); mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString()); } } } if (param.get(String8("test_cmd_policy_channels"), value) == NO_ERROR) { param.remove(String8("test_cmd_policy_channels")); int channels = 0; if (value == "Channels Stereo") { channels = AUDIO_CHANNEL_OUT_STEREO; } else if (value == "Channels Mono") { channels = AUDIO_CHANNEL_OUT_MONO; } if (channels != 0) { if (target == "Manager") { mTestChannels = channels; } else if (mTestOutputs[mCurOutput] != 0) { AudioParameter outputParam = AudioParameter(); outputParam.addInt(String8("channels"), channels); mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString()); } } } if (param.getInt(String8("test_cmd_policy_sampleRate"), valueInt) == NO_ERROR) { param.remove(String8("test_cmd_policy_sampleRate")); if (valueInt >= 0 && valueInt <= 96000) { int samplingRate = valueInt; if (target == "Manager") { mTestSamplingRate = samplingRate; } else if (mTestOutputs[mCurOutput] != 0) { AudioParameter outputParam = AudioParameter(); outputParam.addInt(String8("sampling_rate"), samplingRate); mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString()); } } } if (param.get(String8("test_cmd_policy_reopen"), value) == NO_ERROR) { param.remove(String8("test_cmd_policy_reopen")); mpClientInterface->closeOutput(mpClientInterface->closeOutput(mPrimaryOutput);); audio_module_handle_t moduleHandle = mPrimaryOutput->getModuleHandle(); removeOutput(mPrimaryOutput->mIoHandle); sp outputDesc = new AudioOutputDescriptor(NULL, mpClientInterface); outputDesc->mDevice = AUDIO_DEVICE_OUT_SPEAKER; audio_config_t config = AUDIO_CONFIG_INITIALIZER; config.sample_rate = outputDesc->mSamplingRate; config.channel_mask = outputDesc->mChannelMask; config.format = outputDesc->mFormat; audio_io_handle_t handle; status_t status = mpClientInterface->openOutput(moduleHandle, &handle, &config, &outputDesc->mDevice, String8(""), &outputDesc->mLatency, outputDesc->mFlags); if (status != NO_ERROR) { ALOGE("Failed to reopen hardware output stream, " "samplingRate: %d, format %d, channels %d", outputDesc->mSamplingRate, outputDesc->mFormat, outputDesc->mChannelMask); } else { outputDesc->mSamplingRate = config.sample_rate; outputDesc->mChannelMask = config.channel_mask; outputDesc->mFormat = config.format; mPrimaryOutput = outputDesc; AudioParameter outputCmd = AudioParameter(); outputCmd.addInt(String8("set_id"), 0); mpClientInterface->setParameters(handle, outputCmd.toString()); addOutput(handle, outputDesc); } } mpClientInterface->setParameters(0, String8("test_cmd_policy=")); } } return false; } void AudioPolicyManager::exit() { { AutoMutex _l(mLock); requestExit(); mWaitWorkCV.signal(); } requestExitAndWait(); } int AudioPolicyManager::testOutputIndex(audio_io_handle_t output) { for (int i = 0; i < NUM_TEST_OUTPUTS; i++) { if (output == mTestOutputs[i]) return i; } return 0; } #endif //AUDIO_POLICY_TEST // --- void AudioPolicyManager::addOutput(audio_io_handle_t output, sp outputDesc) { outputDesc->setIoHandle(output); mOutputs.add(output, outputDesc); nextAudioPortGeneration(); } void AudioPolicyManager::removeOutput(audio_io_handle_t output) { mOutputs.removeItem(output); } void AudioPolicyManager::addInput(audio_io_handle_t input, sp inputDesc) { inputDesc->setIoHandle(input); mInputs.add(input, inputDesc); nextAudioPortGeneration(); } void AudioPolicyManager::findIoHandlesByAddress(sp desc /*in*/, const audio_devices_t device /*in*/, const String8 address /*in*/, SortedVector& outputs /*out*/) { sp devDesc = desc->mProfile->mSupportedDevices.getDevice(device, address); if (devDesc != 0) { ALOGV("findIoHandlesByAddress(): adding opened output %d on same address %s", desc->mIoHandle, address.string()); outputs.add(desc->mIoHandle); } } status_t AudioPolicyManager::checkOutputsForDevice(const sp devDesc, audio_policy_dev_state_t state, SortedVector& outputs, const String8 address) { audio_devices_t device = devDesc->type(); sp desc; // erase all current sample rates, formats and channel masks devDesc->clearCapabilities(); if (state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) { // first list already open outputs that can be routed to this device for (size_t i = 0; i < mOutputs.size(); i++) { desc = mOutputs.valueAt(i); if (!desc->isDuplicated() && (desc->supportedDevices() & device)) { if (!device_distinguishes_on_address(device)) { ALOGV("checkOutputsForDevice(): adding opened output %d", mOutputs.keyAt(i)); outputs.add(mOutputs.keyAt(i)); } else { ALOGV(" checking address match due to device 0x%x", device); findIoHandlesByAddress(desc, device, address, outputs); } } } // then look for output profiles that can be routed to this device SortedVector< sp > profiles; for (size_t i = 0; i < mHwModules.size(); i++) { if (mHwModules[i]->mHandle == 0) { continue; } for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++) { sp profile = mHwModules[i]->mOutputProfiles[j]; if (profile->mSupportedDevices.types() & device) { if (!device_distinguishes_on_address(device) || address == profile->mSupportedDevices[0]->mAddress) { profiles.add(profile); ALOGV("checkOutputsForDevice(): adding profile %zu from module %zu", j, i); } } } } ALOGV(" found %d profiles, %d outputs", profiles.size(), outputs.size()); if (profiles.isEmpty() && outputs.isEmpty()) { ALOGW("checkOutputsForDevice(): No output available for device %04x", device); return BAD_VALUE; } // open outputs for matching profiles if needed. Direct outputs are also opened to // query for dynamic parameters and will be closed later by setDeviceConnectionState() for (ssize_t profile_index = 0; profile_index < (ssize_t)profiles.size(); profile_index++) { sp profile = profiles[profile_index]; // nothing to do if one output is already opened for this profile size_t j; for (j = 0; j < outputs.size(); j++) { desc = mOutputs.valueFor(outputs.itemAt(j)); if (!desc->isDuplicated() && desc->mProfile == profile) { // matching profile: save the sample rates, format and channel masks supported // by the profile in our device descriptor devDesc->importAudioPort(profile); break; } } if (j != outputs.size()) { continue; } ALOGV("opening output for device %08x with params %s profile %p", device, address.string(), profile.get()); desc = new SwAudioOutputDescriptor(profile, mpClientInterface); desc->mDevice = device; audio_config_t config = AUDIO_CONFIG_INITIALIZER; config.sample_rate = desc->mSamplingRate; config.channel_mask = desc->mChannelMask; config.format = desc->mFormat; config.offload_info.sample_rate = desc->mSamplingRate; config.offload_info.channel_mask = desc->mChannelMask; config.offload_info.format = desc->mFormat; audio_io_handle_t output = AUDIO_IO_HANDLE_NONE; status_t status = mpClientInterface->openOutput(profile->getModuleHandle(), &output, &config, &desc->mDevice, address, &desc->mLatency, desc->mFlags); if (status == NO_ERROR) { desc->mSamplingRate = config.sample_rate; desc->mChannelMask = config.channel_mask; desc->mFormat = config.format; // Here is where the out_set_parameters() for card & device gets called if (!address.isEmpty()) { char *param = audio_device_address_to_parameter(device, address); mpClientInterface->setParameters(output, String8(param)); free(param); } // Here is where we step through and resolve any "dynamic" fields String8 reply; char *value; if (profile->mSamplingRates[0] == 0) { reply = mpClientInterface->getParameters(output, String8(AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES)); ALOGV("checkOutputsForDevice() supported sampling rates %s", reply.string()); value = strpbrk((char *)reply.string(), "="); if (value != NULL) { profile->loadSamplingRates(value + 1); } } if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) { reply = mpClientInterface->getParameters(output, String8(AUDIO_PARAMETER_STREAM_SUP_FORMATS)); ALOGV("checkOutputsForDevice() supported formats %s", reply.string()); value = strpbrk((char *)reply.string(), "="); if (value != NULL) { profile->loadFormats(value + 1); } } if (profile->mChannelMasks[0] == 0) { reply = mpClientInterface->getParameters(output, String8(AUDIO_PARAMETER_STREAM_SUP_CHANNELS)); ALOGV("checkOutputsForDevice() supported channel masks %s", reply.string()); value = strpbrk((char *)reply.string(), "="); if (value != NULL) { profile->loadOutChannels(value + 1); } } if (((profile->mSamplingRates[0] == 0) && (profile->mSamplingRates.size() < 2)) || ((profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) && (profile->mFormats.size() < 2)) || ((profile->mChannelMasks[0] == 0) && (profile->mChannelMasks.size() < 2))) { ALOGW("checkOutputsForDevice() missing param"); mpClientInterface->closeOutput(output); output = AUDIO_IO_HANDLE_NONE; } else if (profile->mSamplingRates[0] == 0 || profile->mFormats[0] == 0 || profile->mChannelMasks[0] == 0) { mpClientInterface->closeOutput(output); config.sample_rate = profile->pickSamplingRate(); config.channel_mask = profile->pickChannelMask(); config.format = profile->pickFormat(); config.offload_info.sample_rate = config.sample_rate; config.offload_info.channel_mask = config.channel_mask; config.offload_info.format = config.format; status = mpClientInterface->openOutput(profile->getModuleHandle(), &output, &config, &desc->mDevice, address, &desc->mLatency, desc->mFlags); if (status == NO_ERROR) { desc->mSamplingRate = config.sample_rate; desc->mChannelMask = config.channel_mask; desc->mFormat = config.format; } else { output = AUDIO_IO_HANDLE_NONE; } } if (output != AUDIO_IO_HANDLE_NONE) { addOutput(output, desc); if (device_distinguishes_on_address(device) && address != "0") { sp policyMix; if (mPolicyMixes.getAudioPolicyMix(address, policyMix) != NO_ERROR) { ALOGE("checkOutputsForDevice() cannot find policy for address %s", address.string()); } policyMix->setOutput(desc); desc->mPolicyMix = policyMix->getMix(); } else if ((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) == 0) { // no duplicated output for direct outputs and // outputs used by dynamic policy mixes audio_io_handle_t duplicatedOutput = AUDIO_IO_HANDLE_NONE; // set initial stream volume for device applyStreamVolumes(desc, 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->mIoHandle); if (duplicatedOutput != AUDIO_IO_HANDLE_NONE) { // add duplicated output descriptor sp dupOutputDesc = new SwAudioOutputDescriptor(NULL, mpClientInterface); dupOutputDesc->mOutput1 = mPrimaryOutput; dupOutputDesc->mOutput2 = desc; dupOutputDesc->mSamplingRate = desc->mSamplingRate; dupOutputDesc->mFormat = desc->mFormat; dupOutputDesc->mChannelMask = desc->mChannelMask; dupOutputDesc->mLatency = desc->mLatency; addOutput(duplicatedOutput, dupOutputDesc); applyStreamVolumes(dupOutputDesc, device, 0, true); } else { ALOGW("checkOutputsForDevice() could not open dup output for %d and %d", mPrimaryOutput->mIoHandle, output); mpClientInterface->closeOutput(output); removeOutput(output); nextAudioPortGeneration(); output = AUDIO_IO_HANDLE_NONE; } } } } else { output = AUDIO_IO_HANDLE_NONE; } if (output == AUDIO_IO_HANDLE_NONE) { ALOGW("checkOutputsForDevice() could not open output for device %x", device); profiles.removeAt(profile_index); profile_index--; } else { outputs.add(output); devDesc->importAudioPort(profile); if (device_distinguishes_on_address(device)) { ALOGV("checkOutputsForDevice(): setOutputDevice(dev=0x%x, addr=%s)", device, address.string()); setOutputDevice(desc, device, true/*force*/, 0/*delay*/, NULL/*patch handle*/, address.string()); } ALOGV("checkOutputsForDevice(): adding output %d", output); } } if (profiles.isEmpty()) { ALOGW("checkOutputsForDevice(): No output available for device %04x", device); return BAD_VALUE; } } else { // Disconnect // check if one opened output is not needed any more after disconnecting one device for (size_t i = 0; i < mOutputs.size(); i++) { desc = mOutputs.valueAt(i); if (!desc->isDuplicated()) { // exact match on device if (device_distinguishes_on_address(device) && (desc->supportedDevices() == device)) { findIoHandlesByAddress(desc, device, address, outputs); } else if (!(desc->supportedDevices() & mAvailableOutputDevices.types())) { ALOGV("checkOutputsForDevice(): disconnecting adding output %d", mOutputs.keyAt(i)); outputs.add(mOutputs.keyAt(i)); } } } // Clear any profiles associated with the disconnected device. for (size_t i = 0; i < mHwModules.size(); i++) { if (mHwModules[i]->mHandle == 0) { continue; } for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++) { sp profile = mHwModules[i]->mOutputProfiles[j]; if (profile->mSupportedDevices.types() & device) { ALOGV("checkOutputsForDevice(): " "clearing direct output profile %zu on module %zu", j, i); if (profile->mSamplingRates[0] == 0) { profile->mSamplingRates.clear(); profile->mSamplingRates.add(0); } if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) { profile->mFormats.clear(); profile->mFormats.add(AUDIO_FORMAT_DEFAULT); } if (profile->mChannelMasks[0] == 0) { profile->mChannelMasks.clear(); profile->mChannelMasks.add(0); } } } } } return NO_ERROR; } status_t AudioPolicyManager::checkInputsForDevice(audio_devices_t device, audio_policy_dev_state_t state, SortedVector& inputs, const String8 address) { sp desc; if (state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) { // first list already open inputs that can be routed to this device for (size_t input_index = 0; input_index < mInputs.size(); input_index++) { desc = mInputs.valueAt(input_index); if (desc->mProfile->mSupportedDevices.types() & (device & ~AUDIO_DEVICE_BIT_IN)) { ALOGV("checkInputsForDevice(): adding opened input %d", mInputs.keyAt(input_index)); inputs.add(mInputs.keyAt(input_index)); } } // then look for input profiles that can be routed to this device SortedVector< sp > profiles; for (size_t module_idx = 0; module_idx < mHwModules.size(); module_idx++) { if (mHwModules[module_idx]->mHandle == 0) { continue; } for (size_t profile_index = 0; profile_index < mHwModules[module_idx]->mInputProfiles.size(); profile_index++) { sp profile = mHwModules[module_idx]->mInputProfiles[profile_index]; if (profile->mSupportedDevices.types() & (device & ~AUDIO_DEVICE_BIT_IN)) { if (!device_distinguishes_on_address(device) || address == profile->mSupportedDevices[0]->mAddress) { profiles.add(profile); ALOGV("checkInputsForDevice(): adding profile %zu from module %zu", profile_index, module_idx); } } } } if (profiles.isEmpty() && inputs.isEmpty()) { ALOGW("checkInputsForDevice(): No input available for device 0x%X", device); return BAD_VALUE; } // open inputs for matching profiles if needed. Direct inputs are also opened to // query for dynamic parameters and will be closed later by setDeviceConnectionState() for (ssize_t profile_index = 0; profile_index < (ssize_t)profiles.size(); profile_index++) { sp profile = profiles[profile_index]; // nothing to do if one input is already opened for this profile size_t input_index; for (input_index = 0; input_index < mInputs.size(); input_index++) { desc = mInputs.valueAt(input_index); if (desc->mProfile == profile) { break; } } if (input_index != mInputs.size()) { continue; } ALOGV("opening input for device 0x%X with params %s", device, address.string()); desc = new AudioInputDescriptor(profile); desc->mDevice = device; audio_config_t config = AUDIO_CONFIG_INITIALIZER; config.sample_rate = desc->mSamplingRate; config.channel_mask = desc->mChannelMask; config.format = desc->mFormat; audio_io_handle_t input = AUDIO_IO_HANDLE_NONE; status_t status = mpClientInterface->openInput(profile->getModuleHandle(), &input, &config, &desc->mDevice, address, AUDIO_SOURCE_MIC, AUDIO_INPUT_FLAG_NONE /*FIXME*/); if (status == NO_ERROR) { desc->mSamplingRate = config.sample_rate; desc->mChannelMask = config.channel_mask; desc->mFormat = config.format; if (!address.isEmpty()) { char *param = audio_device_address_to_parameter(device, address); mpClientInterface->setParameters(input, String8(param)); free(param); } // Here is where we step through and resolve any "dynamic" fields String8 reply; char *value; if (profile->mSamplingRates[0] == 0) { reply = mpClientInterface->getParameters(input, String8(AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES)); ALOGV("checkInputsForDevice() direct input sup sampling rates %s", reply.string()); value = strpbrk((char *)reply.string(), "="); if (value != NULL) { profile->loadSamplingRates(value + 1); } } if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) { reply = mpClientInterface->getParameters(input, String8(AUDIO_PARAMETER_STREAM_SUP_FORMATS)); ALOGV("checkInputsForDevice() direct input sup formats %s", reply.string()); value = strpbrk((char *)reply.string(), "="); if (value != NULL) { profile->loadFormats(value + 1); } } if (profile->mChannelMasks[0] == 0) { reply = mpClientInterface->getParameters(input, String8(AUDIO_PARAMETER_STREAM_SUP_CHANNELS)); ALOGV("checkInputsForDevice() direct input sup channel masks %s", reply.string()); value = strpbrk((char *)reply.string(), "="); if (value != NULL) { profile->loadInChannels(value + 1); } } if (((profile->mSamplingRates[0] == 0) && (profile->mSamplingRates.size() < 2)) || ((profile->mFormats[0] == 0) && (profile->mFormats.size() < 2)) || ((profile->mChannelMasks[0] == 0) && (profile->mChannelMasks.size() < 2))) { ALOGW("checkInputsForDevice() direct input missing param"); mpClientInterface->closeInput(input); input = AUDIO_IO_HANDLE_NONE; } if (input != 0) { addInput(input, desc); } } // endif input != 0 if (input == AUDIO_IO_HANDLE_NONE) { ALOGW("checkInputsForDevice() could not open input for device 0x%X", device); profiles.removeAt(profile_index); profile_index--; } else { inputs.add(input); ALOGV("checkInputsForDevice(): adding input %d", input); } } // end scan profiles if (profiles.isEmpty()) { ALOGW("checkInputsForDevice(): No input available for device 0x%X", device); return BAD_VALUE; } } else { // Disconnect // check if one opened input is not needed any more after disconnecting one device for (size_t input_index = 0; input_index < mInputs.size(); input_index++) { desc = mInputs.valueAt(input_index); if (!(desc->mProfile->mSupportedDevices.types() & mAvailableInputDevices.types() & ~AUDIO_DEVICE_BIT_IN)) { ALOGV("checkInputsForDevice(): disconnecting adding input %d", mInputs.keyAt(input_index)); inputs.add(mInputs.keyAt(input_index)); } } // Clear any profiles associated with the disconnected device. for (size_t module_index = 0; module_index < mHwModules.size(); module_index++) { if (mHwModules[module_index]->mHandle == 0) { continue; } for (size_t profile_index = 0; profile_index < mHwModules[module_index]->mInputProfiles.size(); profile_index++) { sp profile = mHwModules[module_index]->mInputProfiles[profile_index]; if (profile->mSupportedDevices.types() & device & ~AUDIO_DEVICE_BIT_IN) { ALOGV("checkInputsForDevice(): clearing direct input profile %zu on module %zu", profile_index, module_index); if (profile->mSamplingRates[0] == 0) { profile->mSamplingRates.clear(); profile->mSamplingRates.add(0); } if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) { profile->mFormats.clear(); profile->mFormats.add(AUDIO_FORMAT_DEFAULT); } if (profile->mChannelMasks[0] == 0) { profile->mChannelMasks.clear(); profile->mChannelMasks.add(0); } } } } } // end disconnect return NO_ERROR; } void AudioPolicyManager::closeOutput(audio_io_handle_t output) { ALOGV("closeOutput(%d)", output); sp outputDesc = mOutputs.valueFor(output); if (outputDesc == NULL) { ALOGW("closeOutput() unknown output %d", output); return; } mPolicyMixes.closeOutput(outputDesc); // look for duplicated outputs connected to the output being removed. for (size_t i = 0; i < mOutputs.size(); i++) { sp dupOutputDesc = mOutputs.valueAt(i); if (dupOutputDesc->isDuplicated() && (dupOutputDesc->mOutput1 == outputDesc || dupOutputDesc->mOutput2 == outputDesc)) { sp outputDesc2; if (dupOutputDesc->mOutput1 == outputDesc) { outputDesc2 = dupOutputDesc->mOutput2; } else { outputDesc2 = dupOutputDesc->mOutput1; } // As all active tracks on duplicated output will be deleted, // and as they were also referenced on the other output, the reference // count for their stream type must be adjusted accordingly on // the other output. for (int j = 0; j < AUDIO_STREAM_CNT; j++) { int refCount = dupOutputDesc->mRefCount[j]; outputDesc2->changeRefCount((audio_stream_type_t)j,-refCount); } audio_io_handle_t duplicatedOutput = mOutputs.keyAt(i); ALOGV("closeOutput() closing also duplicated output %d", duplicatedOutput); mpClientInterface->closeOutput(duplicatedOutput); removeOutput(duplicatedOutput); } } nextAudioPortGeneration(); ssize_t index = mAudioPatches.indexOfKey(outputDesc->mPatchHandle); if (index >= 0) { sp patchDesc = mAudioPatches.valueAt(index); status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0); mAudioPatches.removeItemsAt(index); mpClientInterface->onAudioPatchListUpdate(); } AudioParameter param; param.add(String8("closing"), String8("true")); mpClientInterface->setParameters(output, param.toString()); mpClientInterface->closeOutput(output); removeOutput(output); mPreviousOutputs = mOutputs; } void AudioPolicyManager::closeInput(audio_io_handle_t input) { ALOGV("closeInput(%d)", input); sp inputDesc = mInputs.valueFor(input); if (inputDesc == NULL) { ALOGW("closeInput() unknown input %d", input); return; } nextAudioPortGeneration(); ssize_t index = mAudioPatches.indexOfKey(inputDesc->mPatchHandle); if (index >= 0) { sp patchDesc = mAudioPatches.valueAt(index); status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0); mAudioPatches.removeItemsAt(index); mpClientInterface->onAudioPatchListUpdate(); } mpClientInterface->closeInput(input); mInputs.removeItem(input); } SortedVector AudioPolicyManager::getOutputsForDevice( audio_devices_t device, SwAudioOutputCollection openOutputs) { SortedVector outputs; ALOGVV("getOutputsForDevice() device %04x", device); for (size_t i = 0; i < openOutputs.size(); i++) { ALOGVV("output %d isDuplicated=%d device=%04x", i, openOutputs.valueAt(i)->isDuplicated(), openOutputs.valueAt(i)->supportedDevices()); if ((device & openOutputs.valueAt(i)->supportedDevices()) == device) { ALOGVV("getOutputsForDevice() found output %d", openOutputs.keyAt(i)); outputs.add(openOutputs.keyAt(i)); } } return outputs; } bool AudioPolicyManager::vectorsEqual(SortedVector& outputs1, SortedVector& outputs2) { if (outputs1.size() != outputs2.size()) { return false; } for (size_t i = 0; i < outputs1.size(); i++) { if (outputs1[i] != outputs2[i]) { return false; } } return true; } void AudioPolicyManager::checkOutputForStrategy(routing_strategy strategy) { audio_devices_t oldDevice = getDeviceForStrategy(strategy, true /*fromCache*/); audio_devices_t newDevice = getDeviceForStrategy(strategy, false /*fromCache*/); SortedVector srcOutputs = getOutputsForDevice(oldDevice, mPreviousOutputs); SortedVector dstOutputs = getOutputsForDevice(newDevice, mOutputs); // also take into account external policy-related changes: add all outputs which are // associated with policies in the "before" and "after" output vectors ALOGVV("checkOutputForStrategy(): policy related outputs"); for (size_t i = 0 ; i < mPreviousOutputs.size() ; i++) { const sp desc = mPreviousOutputs.valueAt(i); if (desc != 0 && desc->mPolicyMix != NULL) { srcOutputs.add(desc->mIoHandle); ALOGVV(" previous outputs: adding %d", desc->mIoHandle); } } for (size_t i = 0 ; i < mOutputs.size() ; i++) { const sp desc = mOutputs.valueAt(i); if (desc != 0 && desc->mPolicyMix != NULL) { dstOutputs.add(desc->mIoHandle); ALOGVV(" new outputs: adding %d", desc->mIoHandle); } } if (!vectorsEqual(srcOutputs,dstOutputs)) { ALOGV("checkOutputForStrategy() strategy %d, moving from output %d to output %d", strategy, srcOutputs[0], dstOutputs[0]); // mute strategy while moving tracks from one output to another for (size_t i = 0; i < srcOutputs.size(); i++) { sp desc = mOutputs.valueFor(srcOutputs[i]); if (isStrategyActive(desc, strategy)) { setStrategyMute(strategy, true, desc); setStrategyMute(strategy, false, desc, MUTE_TIME_MS, newDevice); } } // Move effects associated to this strategy from previous output to new output if (strategy == STRATEGY_MEDIA) { audio_io_handle_t fxOutput = selectOutputForEffects(dstOutputs); SortedVector moved; for (size_t i = 0; i < mEffects.size(); i++) { sp effectDesc = mEffects.valueAt(i); if (effectDesc->mSession == AUDIO_SESSION_OUTPUT_MIX && effectDesc->mIo != fxOutput) { if (moved.indexOf(effectDesc->mIo) < 0) { ALOGV("checkOutputForStrategy() moving effect %d to output %d", mEffects.keyAt(i), fxOutput); mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, effectDesc->mIo, fxOutput); moved.add(effectDesc->mIo); } effectDesc->mIo = fxOutput; } } } // Move tracks associated to this strategy from previous output to new output for (int i = 0; i < AUDIO_STREAM_CNT; i++) { if (i == AUDIO_STREAM_PATCH) { continue; } if (getStrategy((audio_stream_type_t)i) == strategy) { mpClientInterface->invalidateStream((audio_stream_type_t)i); } } } } void AudioPolicyManager::checkOutputForAllStrategies() { if (mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) checkOutputForStrategy(STRATEGY_ENFORCED_AUDIBLE); checkOutputForStrategy(STRATEGY_PHONE); if (mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) != AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) checkOutputForStrategy(STRATEGY_ENFORCED_AUDIBLE); checkOutputForStrategy(STRATEGY_SONIFICATION); checkOutputForStrategy(STRATEGY_SONIFICATION_RESPECTFUL); checkOutputForStrategy(STRATEGY_ACCESSIBILITY); checkOutputForStrategy(STRATEGY_MEDIA); checkOutputForStrategy(STRATEGY_DTMF); checkOutputForStrategy(STRATEGY_REROUTING); } void AudioPolicyManager::checkA2dpSuspend() { audio_io_handle_t a2dpOutput = mOutputs.getA2dpOutput(); if (a2dpOutput == 0) { mA2dpSuspended = false; return; } bool isScoConnected = ((mAvailableInputDevices.types() & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET & ~AUDIO_DEVICE_BIT_IN) != 0) || ((mAvailableOutputDevices.types() & AUDIO_DEVICE_OUT_ALL_SCO) != 0); // suspend A2DP output if: // (NOT already suspended) && // ((SCO device is connected && // (forced usage for communication || for record is SCO))) || // (phone state is ringing || in call) // // restore A2DP output if: // (Already suspended) && // ((SCO device is NOT connected || // (forced usage NOT for communication && NOT for record is SCO))) && // (phone state is NOT ringing && NOT in call) // if (mA2dpSuspended) { if ((!isScoConnected || ((mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_COMMUNICATION) != AUDIO_POLICY_FORCE_BT_SCO) && (mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_RECORD) != AUDIO_POLICY_FORCE_BT_SCO))) && ((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) && (mEngine->getPhoneState() != AUDIO_MODE_RINGTONE))) { mpClientInterface->restoreOutput(a2dpOutput); mA2dpSuspended = false; } } else { if ((isScoConnected && ((mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_COMMUNICATION) == AUDIO_POLICY_FORCE_BT_SCO) || (mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_RECORD) == AUDIO_POLICY_FORCE_BT_SCO))) || ((mEngine->getPhoneState() == AUDIO_MODE_IN_CALL) || (mEngine->getPhoneState() == AUDIO_MODE_RINGTONE))) { mpClientInterface->suspendOutput(a2dpOutput); mA2dpSuspended = true; } } } audio_devices_t AudioPolicyManager::getNewOutputDevice(const sp& outputDesc, bool fromCache) { audio_devices_t device = AUDIO_DEVICE_NONE; ssize_t index = mAudioPatches.indexOfKey(outputDesc->mPatchHandle); if (index >= 0) { sp patchDesc = mAudioPatches.valueAt(index); if (patchDesc->mUid != mUidCached) { ALOGV("getNewOutputDevice() device %08x forced by patch %d", outputDesc->device(), outputDesc->mPatchHandle); return outputDesc->device(); } } // check the following by order of priority to request a routing change if necessary: // 1: the strategy enforced audible is active and enforced on the output: // use device for strategy enforced audible // 2: we are in call or the strategy phone is active on the output: // use device for strategy phone // 3: the strategy for enforced audible is active but not enforced on the output: // use the device for strategy enforced audible // 4: the strategy sonification is active on the output: // use device for strategy sonification // 5: the strategy "respectful" sonification is active on the output: // use device for strategy "respectful" sonification // 6: the strategy accessibility is active on the output: // use device for strategy accessibility // 7: the strategy media is active on the output: // use device for strategy media // 8: the strategy DTMF is active on the output: // use device for strategy DTMF // 9: the strategy for beacon, a.k.a. "transmitted through speaker" is active on the output: // use device for strategy t-t-s if (isStrategyActive(outputDesc, STRATEGY_ENFORCED_AUDIBLE) && mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) { device = getDeviceForStrategy(STRATEGY_ENFORCED_AUDIBLE, fromCache); } else if (isInCall() || isStrategyActive(outputDesc, STRATEGY_PHONE)) { device = getDeviceForStrategy(STRATEGY_PHONE, fromCache); } else if (isStrategyActive(outputDesc, STRATEGY_ENFORCED_AUDIBLE)) { device = getDeviceForStrategy(STRATEGY_ENFORCED_AUDIBLE, fromCache); } else if (isStrategyActive(outputDesc, STRATEGY_SONIFICATION)) { device = getDeviceForStrategy(STRATEGY_SONIFICATION, fromCache); } else if (isStrategyActive(outputDesc, STRATEGY_SONIFICATION_RESPECTFUL)) { device = getDeviceForStrategy(STRATEGY_SONIFICATION_RESPECTFUL, fromCache); } else if (isStrategyActive(outputDesc, STRATEGY_ACCESSIBILITY)) { device = getDeviceForStrategy(STRATEGY_ACCESSIBILITY, fromCache); } else if (isStrategyActive(outputDesc, STRATEGY_MEDIA)) { device = getDeviceForStrategy(STRATEGY_MEDIA, fromCache); } else if (isStrategyActive(outputDesc, STRATEGY_DTMF)) { device = getDeviceForStrategy(STRATEGY_DTMF, fromCache); } else if (isStrategyActive(outputDesc, STRATEGY_TRANSMITTED_THROUGH_SPEAKER)) { device = getDeviceForStrategy(STRATEGY_TRANSMITTED_THROUGH_SPEAKER, fromCache); } else if (isStrategyActive(outputDesc, STRATEGY_REROUTING)) { device = getDeviceForStrategy(STRATEGY_REROUTING, fromCache); } ALOGV("getNewOutputDevice() selected device %x", device); return device; } audio_devices_t AudioPolicyManager::getNewInputDevice(audio_io_handle_t input) { sp inputDesc = mInputs.valueFor(input); ssize_t index = mAudioPatches.indexOfKey(inputDesc->mPatchHandle); if (index >= 0) { sp patchDesc = mAudioPatches.valueAt(index); if (patchDesc->mUid != mUidCached) { ALOGV("getNewInputDevice() device %08x forced by patch %d", inputDesc->mDevice, inputDesc->mPatchHandle); return inputDesc->mDevice; } } audio_devices_t device = getDeviceAndMixForInputSource(inputDesc->mInputSource); return device; } uint32_t AudioPolicyManager::getStrategyForStream(audio_stream_type_t stream) { return (uint32_t)getStrategy(stream); } audio_devices_t AudioPolicyManager::getDevicesForStream(audio_stream_type_t stream) { // By checking the range of stream before calling getStrategy, we avoid // getStrategy's behavior for invalid streams. getStrategy would do a ALOGE // and then return STRATEGY_MEDIA, but we want to return the empty set. if (stream < (audio_stream_type_t) 0 || stream >= AUDIO_STREAM_PUBLIC_CNT) { return AUDIO_DEVICE_NONE; } audio_devices_t devices; routing_strategy strategy = getStrategy(stream); devices = getDeviceForStrategy(strategy, true /*fromCache*/); SortedVector outputs = getOutputsForDevice(devices, mOutputs); for (size_t i = 0; i < outputs.size(); i++) { sp outputDesc = mOutputs.valueFor(outputs[i]); if (isStrategyActive(outputDesc, strategy)) { devices = outputDesc->device(); break; } } /*Filter SPEAKER_SAFE out of results, as AudioService doesn't know about it and doesn't really need to.*/ if (devices & AUDIO_DEVICE_OUT_SPEAKER_SAFE) { devices |= AUDIO_DEVICE_OUT_SPEAKER; devices &= ~AUDIO_DEVICE_OUT_SPEAKER_SAFE; } return devices; } routing_strategy AudioPolicyManager::getStrategy(audio_stream_type_t stream) const { ALOG_ASSERT(stream != AUDIO_STREAM_PATCH,"getStrategy() called for AUDIO_STREAM_PATCH"); return mEngine->getStrategyForStream(stream); } uint32_t AudioPolicyManager::getStrategyForAttr(const audio_attributes_t *attr) { // flags to strategy mapping if ((attr->flags & AUDIO_FLAG_BEACON) == AUDIO_FLAG_BEACON) { return (uint32_t) STRATEGY_TRANSMITTED_THROUGH_SPEAKER; } if ((attr->flags & AUDIO_FLAG_AUDIBILITY_ENFORCED) == AUDIO_FLAG_AUDIBILITY_ENFORCED) { return (uint32_t) STRATEGY_ENFORCED_AUDIBLE; } // usage to strategy mapping return static_cast(mEngine->getStrategyForUsage(attr->usage)); } void AudioPolicyManager::handleNotificationRoutingForStream(audio_stream_type_t stream) { switch(stream) { case AUDIO_STREAM_MUSIC: checkOutputForStrategy(STRATEGY_SONIFICATION_RESPECTFUL); updateDevicesAndOutputs(); break; default: break; } } uint32_t AudioPolicyManager::handleEventForBeacon(int event) { switch(event) { case STARTING_OUTPUT: mBeaconMuteRefCount++; break; case STOPPING_OUTPUT: if (mBeaconMuteRefCount > 0) { mBeaconMuteRefCount--; } break; case STARTING_BEACON: mBeaconPlayingRefCount++; break; case STOPPING_BEACON: if (mBeaconPlayingRefCount > 0) { mBeaconPlayingRefCount--; } break; } if (mBeaconMuteRefCount > 0) { // any playback causes beacon to be muted return setBeaconMute(true); } else { // no other playback: unmute when beacon starts playing, mute when it stops return setBeaconMute(mBeaconPlayingRefCount == 0); } } uint32_t AudioPolicyManager::setBeaconMute(bool mute) { ALOGV("setBeaconMute(%d) mBeaconMuteRefCount=%d mBeaconPlayingRefCount=%d", mute, mBeaconMuteRefCount, mBeaconPlayingRefCount); // keep track of muted state to avoid repeating mute/unmute operations if (mBeaconMuted != mute) { // mute/unmute AUDIO_STREAM_TTS on all outputs ALOGV("\t muting %d", mute); uint32_t maxLatency = 0; for (size_t i = 0; i < mOutputs.size(); i++) { sp desc = mOutputs.valueAt(i); setStreamMute(AUDIO_STREAM_TTS, mute/*on*/, desc, 0 /*delay*/, AUDIO_DEVICE_NONE); const uint32_t latency = desc->latency() * 2; if (latency > maxLatency) { maxLatency = latency; } } mBeaconMuted = mute; return maxLatency; } return 0; } audio_devices_t AudioPolicyManager::getDeviceForStrategy(routing_strategy strategy, bool fromCache) { // Routing // see if we have an explicit route // scan the whole RouteMap, for each entry, convert the stream type to a strategy // (getStrategy(stream)). // if the strategy from the stream type in the RouteMap is the same as the argument above, // and activity count is non-zero // the device = the device from the descriptor in the RouteMap, and exit. for (size_t routeIndex = 0; routeIndex < mOutputRoutes.size(); routeIndex++) { sp route = mOutputRoutes.valueAt(routeIndex); routing_strategy strat = getStrategy(route->mStreamType); if (strat == strategy && route->mDeviceDescriptor != 0 /*&& route->mActivityCount != 0*/) { return route->mDeviceDescriptor->type(); } } if (fromCache) { ALOGVV("getDeviceForStrategy() from cache strategy %d, device %x", strategy, mDeviceForStrategy[strategy]); return mDeviceForStrategy[strategy]; } return mEngine->getDeviceForStrategy(strategy); } void AudioPolicyManager::updateDevicesAndOutputs() { for (int i = 0; i < NUM_STRATEGIES; i++) { mDeviceForStrategy[i] = getDeviceForStrategy((routing_strategy)i, false /*fromCache*/); } mPreviousOutputs = mOutputs; } uint32_t AudioPolicyManager::checkDeviceMuteStrategies(sp outputDesc, audio_devices_t prevDevice, uint32_t delayMs) { // mute/unmute strategies using an incompatible device combination // if muting, wait for the audio in pcm buffer to be drained before proceeding // if unmuting, unmute only after the specified delay if (outputDesc->isDuplicated()) { return 0; } uint32_t muteWaitMs = 0; audio_devices_t device = outputDesc->device(); bool shouldMute = outputDesc->isActive() && (popcount(device) >= 2); for (size_t i = 0; i < NUM_STRATEGIES; i++) { audio_devices_t curDevice = getDeviceForStrategy((routing_strategy)i, false /*fromCache*/); curDevice = curDevice & outputDesc->supportedDevices(); bool mute = shouldMute && (curDevice & device) && (curDevice != device); bool doMute = false; if (mute && !outputDesc->mStrategyMutedByDevice[i]) { doMute = true; outputDesc->mStrategyMutedByDevice[i] = true; } else if (!mute && outputDesc->mStrategyMutedByDevice[i]){ doMute = true; outputDesc->mStrategyMutedByDevice[i] = false; } if (doMute) { for (size_t j = 0; j < mOutputs.size(); j++) { sp desc = mOutputs.valueAt(j); // skip output if it does not share any device with current output if ((desc->supportedDevices() & outputDesc->supportedDevices()) == AUDIO_DEVICE_NONE) { continue; } ALOGVV("checkDeviceMuteStrategies() %s strategy %d (curDevice %04x)", mute ? "muting" : "unmuting", i, curDevice); setStrategyMute((routing_strategy)i, mute, desc, mute ? 0 : delayMs); if (isStrategyActive(desc, (routing_strategy)i)) { if (mute) { // FIXME: should not need to double latency if volume could be applied // immediately by the audioflinger mixer. We must account for the delay // between now and the next time the audioflinger thread for this output // will process a buffer (which corresponds to one buffer size, // usually 1/2 or 1/4 of the latency). if (muteWaitMs < desc->latency() * 2) { muteWaitMs = desc->latency() * 2; } } } } } } // temporary mute output if device selection changes to avoid volume bursts due to // different per device volumes if (outputDesc->isActive() && (device != prevDevice)) { if (muteWaitMs < outputDesc->latency() * 2) { muteWaitMs = outputDesc->latency() * 2; } for (size_t i = 0; i < NUM_STRATEGIES; i++) { if (isStrategyActive(outputDesc, (routing_strategy)i)) { setStrategyMute((routing_strategy)i, true, outputDesc); // do tempMute unmute after twice the mute wait time setStrategyMute((routing_strategy)i, false, outputDesc, muteWaitMs *2, device); } } } // wait for the PCM output buffers to empty before proceeding with the rest of the command if (muteWaitMs > delayMs) { muteWaitMs -= delayMs; usleep(muteWaitMs * 1000); return muteWaitMs; } return 0; } uint32_t AudioPolicyManager::setOutputDevice(const sp& outputDesc, audio_devices_t device, bool force, int delayMs, audio_patch_handle_t *patchHandle, const char* address) { ALOGV("setOutputDevice() device %04x delayMs %d", device, delayMs); AudioParameter param; uint32_t muteWaitMs; if (outputDesc->isDuplicated()) { muteWaitMs = setOutputDevice(outputDesc->subOutput1(), device, force, delayMs); muteWaitMs += setOutputDevice(outputDesc->subOutput2(), 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->supportedDevices()) == 0)) { return 0; } // filter devices according to output selected device = (audio_devices_t)(device & outputDesc->supportedDevices()); audio_devices_t prevDevice = outputDesc->mDevice; ALOGV("setOutputDevice() prevDevice 0x%04x", prevDevice); if (device != AUDIO_DEVICE_NONE) { outputDesc->mDevice = device; } muteWaitMs = checkDeviceMuteStrategies(outputDesc, prevDevice, delayMs); // Do not change the routing if: // the requested device is AUDIO_DEVICE_NONE // OR the requested device is the same as current device // AND force is not specified // AND the output is connected by a valid audio patch. // Doing this check here allows the caller to call setOutputDevice() without conditions if ((device == AUDIO_DEVICE_NONE || device == prevDevice) && !force && outputDesc->mPatchHandle != 0) { ALOGV("setOutputDevice() setting same device 0x%04x or null device", device); return muteWaitMs; } ALOGV("setOutputDevice() changing device"); // do the routing if (device == AUDIO_DEVICE_NONE) { resetOutputDevice(outputDesc, delayMs, NULL); } else { DeviceVector deviceList = (address == NULL) ? mAvailableOutputDevices.getDevicesFromType(device) : mAvailableOutputDevices.getDevicesFromTypeAddr(device, String8(address)); if (!deviceList.isEmpty()) { struct audio_patch patch; outputDesc->toAudioPortConfig(&patch.sources[0]); patch.num_sources = 1; patch.num_sinks = 0; for (size_t i = 0; i < deviceList.size() && i < AUDIO_PATCH_PORTS_MAX; i++) { deviceList.itemAt(i)->toAudioPortConfig(&patch.sinks[i]); patch.num_sinks++; } ssize_t index; if (patchHandle && *patchHandle != AUDIO_PATCH_HANDLE_NONE) { index = mAudioPatches.indexOfKey(*patchHandle); } else { index = mAudioPatches.indexOfKey(outputDesc->mPatchHandle); } sp< AudioPatch> patchDesc; audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE; if (index >= 0) { patchDesc = mAudioPatches.valueAt(index); afPatchHandle = patchDesc->mAfPatchHandle; } status_t status = mpClientInterface->createAudioPatch(&patch, &afPatchHandle, delayMs); ALOGV("setOutputDevice() createAudioPatch returned %d patchHandle %d" "num_sources %d num_sinks %d", status, afPatchHandle, patch.num_sources, patch.num_sinks); if (status == NO_ERROR) { if (index < 0) { patchDesc = new AudioPatch(&patch, mUidCached); addAudioPatch(patchDesc->mHandle, patchDesc); } else { patchDesc->mPatch = patch; } patchDesc->mAfPatchHandle = afPatchHandle; patchDesc->mUid = mUidCached; if (patchHandle) { *patchHandle = patchDesc->mHandle; } outputDesc->mPatchHandle = patchDesc->mHandle; nextAudioPortGeneration(); mpClientInterface->onAudioPatchListUpdate(); } } // inform all input as well for (size_t i = 0; i < mInputs.size(); i++) { const sp inputDescriptor = mInputs.valueAt(i); if (!is_virtual_input_device(inputDescriptor->mDevice)) { AudioParameter inputCmd = AudioParameter(); ALOGV("%s: inform input %d of device:%d", __func__, inputDescriptor->mIoHandle, device); inputCmd.addInt(String8(AudioParameter::keyRouting),device); mpClientInterface->setParameters(inputDescriptor->mIoHandle, inputCmd.toString(), delayMs); } } } // update stream volumes according to new device applyStreamVolumes(outputDesc, device, delayMs); return muteWaitMs; } status_t AudioPolicyManager::resetOutputDevice(const sp& outputDesc, int delayMs, audio_patch_handle_t *patchHandle) { ssize_t index; if (patchHandle) { index = mAudioPatches.indexOfKey(*patchHandle); } else { index = mAudioPatches.indexOfKey(outputDesc->mPatchHandle); } if (index < 0) { return INVALID_OPERATION; } sp< AudioPatch> patchDesc = mAudioPatches.valueAt(index); status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, delayMs); ALOGV("resetOutputDevice() releaseAudioPatch returned %d", status); outputDesc->mPatchHandle = 0; removeAudioPatch(patchDesc->mHandle); nextAudioPortGeneration(); mpClientInterface->onAudioPatchListUpdate(); return status; } status_t AudioPolicyManager::setInputDevice(audio_io_handle_t input, audio_devices_t device, bool force, audio_patch_handle_t *patchHandle) { status_t status = NO_ERROR; sp inputDesc = mInputs.valueFor(input); if ((device != AUDIO_DEVICE_NONE) && ((device != inputDesc->mDevice) || force)) { inputDesc->mDevice = device; DeviceVector deviceList = mAvailableInputDevices.getDevicesFromType(device); if (!deviceList.isEmpty()) { struct audio_patch patch; inputDesc->toAudioPortConfig(&patch.sinks[0]); // AUDIO_SOURCE_HOTWORD is for internal use only: // handled as AUDIO_SOURCE_VOICE_RECOGNITION by the audio HAL if (patch.sinks[0].ext.mix.usecase.source == AUDIO_SOURCE_HOTWORD && !inputDesc->mIsSoundTrigger) { patch.sinks[0].ext.mix.usecase.source = AUDIO_SOURCE_VOICE_RECOGNITION; } patch.num_sinks = 1; //only one input device for now deviceList.itemAt(0)->toAudioPortConfig(&patch.sources[0]); patch.num_sources = 1; ssize_t index; if (patchHandle && *patchHandle != AUDIO_PATCH_HANDLE_NONE) { index = mAudioPatches.indexOfKey(*patchHandle); } else { index = mAudioPatches.indexOfKey(inputDesc->mPatchHandle); } sp< AudioPatch> patchDesc; audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE; if (index >= 0) { patchDesc = mAudioPatches.valueAt(index); afPatchHandle = patchDesc->mAfPatchHandle; } status_t status = mpClientInterface->createAudioPatch(&patch, &afPatchHandle, 0); ALOGV("setInputDevice() createAudioPatch returned %d patchHandle %d", status, afPatchHandle); if (status == NO_ERROR) { if (index < 0) { patchDesc = new AudioPatch(&patch, mUidCached); addAudioPatch(patchDesc->mHandle, patchDesc); } else { patchDesc->mPatch = patch; } patchDesc->mAfPatchHandle = afPatchHandle; patchDesc->mUid = mUidCached; if (patchHandle) { *patchHandle = patchDesc->mHandle; } inputDesc->mPatchHandle = patchDesc->mHandle; nextAudioPortGeneration(); mpClientInterface->onAudioPatchListUpdate(); } } } return status; } status_t AudioPolicyManager::resetInputDevice(audio_io_handle_t input, audio_patch_handle_t *patchHandle) { sp inputDesc = mInputs.valueFor(input); ssize_t index; if (patchHandle) { index = mAudioPatches.indexOfKey(*patchHandle); } else { index = mAudioPatches.indexOfKey(inputDesc->mPatchHandle); } if (index < 0) { return INVALID_OPERATION; } sp< AudioPatch> patchDesc = mAudioPatches.valueAt(index); status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0); ALOGV("resetInputDevice() releaseAudioPatch returned %d", status); inputDesc->mPatchHandle = 0; removeAudioPatch(patchDesc->mHandle); nextAudioPortGeneration(); mpClientInterface->onAudioPatchListUpdate(); return status; } sp AudioPolicyManager::getInputProfile(audio_devices_t device, String8 address, uint32_t& samplingRate, audio_format_t& format, audio_channel_mask_t& channelMask, audio_input_flags_t flags) { // Choose an input profile based on the requested capture parameters: select the first available // profile supporting all requested parameters. // // TODO: perhaps isCompatibleProfile should return a "matching" score so we can return // the best matching profile, not the first one. for (size_t i = 0; i < mHwModules.size(); i++) { if (mHwModules[i]->mHandle == 0) { continue; } for (size_t j = 0; j < mHwModules[i]->mInputProfiles.size(); j++) { sp profile = mHwModules[i]->mInputProfiles[j]; // profile->log(); if (profile->isCompatibleProfile(device, address, samplingRate, &samplingRate /*updatedSamplingRate*/, format, &format /*updatedFormat*/, channelMask, &channelMask /*updatedChannelMask*/, (audio_output_flags_t) flags)) { return profile; } } } return NULL; } audio_devices_t AudioPolicyManager::getDeviceAndMixForInputSource(audio_source_t inputSource, AudioMix **policyMix) { audio_devices_t availableDeviceTypes = mAvailableInputDevices.types() & ~AUDIO_DEVICE_BIT_IN; audio_devices_t selectedDeviceFromMix = mPolicyMixes.getDeviceAndMixForInputSource(inputSource, availableDeviceTypes, policyMix); if (selectedDeviceFromMix != AUDIO_DEVICE_NONE) { return selectedDeviceFromMix; } return getDeviceForInputSource(inputSource); } audio_devices_t AudioPolicyManager::getDeviceForInputSource(audio_source_t inputSource) { for (size_t routeIndex = 0; routeIndex < mInputRoutes.size(); routeIndex++) { sp route = mInputRoutes.valueAt(routeIndex); if (inputSource == route->mSource && route->mDeviceDescriptor != 0 /*&& route->mActivityCount != 0*/) { return route->mDeviceDescriptor->type(); } } return mEngine->getDeviceForInputSource(inputSource); } float AudioPolicyManager::computeVolume(audio_stream_type_t stream, int index, audio_devices_t device) { float volumeDb = mEngine->volIndexToDb(Volume::getDeviceCategory(device), stream, index); // if a headset is connected, apply the following rules to ring tones and notifications // to avoid sound level bursts in user's ears: // - always attenuate ring tones and notifications volume by 6dB // - if music is playing, always limit the volume to current music volume, // with a minimum threshold at -36dB so that notification is always perceived. const routing_strategy stream_strategy = getStrategy(stream); if ((device & (AUDIO_DEVICE_OUT_BLUETOOTH_A2DP | AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES | AUDIO_DEVICE_OUT_WIRED_HEADSET | AUDIO_DEVICE_OUT_WIRED_HEADPHONE)) && ((stream_strategy == STRATEGY_SONIFICATION) || (stream_strategy == STRATEGY_SONIFICATION_RESPECTFUL) || (stream == AUDIO_STREAM_SYSTEM) || ((stream_strategy == STRATEGY_ENFORCED_AUDIBLE) && (mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) == AUDIO_POLICY_FORCE_NONE))) && mStreams.canBeMuted(stream)) { volumeDb += SONIFICATION_HEADSET_VOLUME_FACTOR_DB; // when the phone is ringing we must consider that music could have been paused just before // by the music application and behave as if music was active if the last music track was // just stopped if (isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY) || mLimitRingtoneVolume) { audio_devices_t musicDevice = getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/); float musicVolDB = computeVolume(AUDIO_STREAM_MUSIC, mStreams.valueFor(AUDIO_STREAM_MUSIC).getVolumeIndex(musicDevice), musicDevice); float minVolDB = (musicVolDB > SONIFICATION_HEADSET_VOLUME_MIN_DB) ? musicVolDB : SONIFICATION_HEADSET_VOLUME_MIN_DB; if (volumeDb > minVolDB) { volumeDb = minVolDB; ALOGV("computeVolume limiting volume to %f musicVol %f", minVolDB, musicVolDB); } } } return volumeDb; } status_t AudioPolicyManager::checkAndSetVolume(audio_stream_type_t stream, int index, const sp& outputDesc, audio_devices_t device, int delayMs, bool force) { // do not change actual stream volume if the stream is muted if (outputDesc->mMuteCount[stream] != 0) { ALOGVV("checkAndSetVolume() stream %d muted count %d", stream, outputDesc->mMuteCount[stream]); return NO_ERROR; } audio_policy_forced_cfg_t forceUseForComm = mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_COMMUNICATION); // do not change in call volume if bluetooth is connected and vice versa if ((stream == AUDIO_STREAM_VOICE_CALL && forceUseForComm == AUDIO_POLICY_FORCE_BT_SCO) || (stream == AUDIO_STREAM_BLUETOOTH_SCO && forceUseForComm != AUDIO_POLICY_FORCE_BT_SCO)) { ALOGV("checkAndSetVolume() cannot set stream %d volume with force use = %d for comm", stream, forceUseForComm); return INVALID_OPERATION; } if (device == AUDIO_DEVICE_NONE) { device = outputDesc->device(); } float volumeDb = computeVolume(stream, index, device); if (outputDesc->isFixedVolume(device)) { volumeDb = 0.0f; } outputDesc->setVolume(volumeDb, stream, device, delayMs, force); if (stream == AUDIO_STREAM_VOICE_CALL || stream == AUDIO_STREAM_BLUETOOTH_SCO) { float voiceVolume; // Force voice volume to max for bluetooth SCO as volume is managed by the headset if (stream == AUDIO_STREAM_VOICE_CALL) { voiceVolume = (float)index/(float)mStreams.valueFor(stream).getVolumeIndexMax(); } else { voiceVolume = 1.0; } if (voiceVolume != mLastVoiceVolume && outputDesc == mPrimaryOutput) { mpClientInterface->setVoiceVolume(voiceVolume, delayMs); mLastVoiceVolume = voiceVolume; } } return NO_ERROR; } void AudioPolicyManager::applyStreamVolumes(const sp& outputDesc, audio_devices_t device, int delayMs, bool force) { ALOGVV("applyStreamVolumes() for device %08x", device); for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) { if (stream == AUDIO_STREAM_PATCH) { continue; } checkAndSetVolume((audio_stream_type_t)stream, mStreams.valueFor((audio_stream_type_t)stream).getVolumeIndex(device), outputDesc, device, delayMs, force); } } void AudioPolicyManager::setStrategyMute(routing_strategy strategy, bool on, const sp& outputDesc, int delayMs, audio_devices_t device) { ALOGVV("setStrategyMute() strategy %d, mute %d, output %d", strategy, on, output); for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) { if (stream == AUDIO_STREAM_PATCH) { continue; } if (getStrategy((audio_stream_type_t)stream) == strategy) { setStreamMute((audio_stream_type_t)stream, on, outputDesc, delayMs, device); } } } void AudioPolicyManager::setStreamMute(audio_stream_type_t stream, bool on, const sp& outputDesc, int delayMs, audio_devices_t device) { const StreamDescriptor& streamDesc = mStreams.valueFor(stream); if (device == AUDIO_DEVICE_NONE) { device = outputDesc->device(); } ALOGVV("setStreamMute() stream %d, mute %d, mMuteCount %d device %04x", stream, on, outputDesc->mMuteCount[stream], device); if (on) { if (outputDesc->mMuteCount[stream] == 0) { if (streamDesc.canBeMuted() && ((stream != AUDIO_STREAM_ENFORCED_AUDIBLE) || (mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) == AUDIO_POLICY_FORCE_NONE))) { checkAndSetVolume(stream, 0, outputDesc, 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), outputDesc, device, delayMs); } } } void AudioPolicyManager::handleIncallSonification(audio_stream_type_t stream, bool starting, bool stateChange) { // if the stream pertains to sonification strategy and we are in call we must // mute the stream if it is low visibility. If it is high visibility, we must play a tone // in the device used for phone strategy and play the tone if the selected device does not // interfere with the device used for phone strategy // if stateChange is true, we are called from setPhoneState() and we must mute or unmute as // many times as there are active tracks on the output const routing_strategy stream_strategy = getStrategy(stream); if ((stream_strategy == STRATEGY_SONIFICATION) || ((stream_strategy == STRATEGY_SONIFICATION_RESPECTFUL))) { sp outputDesc = mPrimaryOutput; ALOGV("handleIncallSonification() stream %d starting %d device %x stateChange %d", stream, starting, outputDesc->mDevice, stateChange); if (outputDesc->mRefCount[stream]) { int muteCount = 1; if (stateChange) { muteCount = outputDesc->mRefCount[stream]; } if (audio_is_low_visibility(stream)) { ALOGV("handleIncallSonification() low visibility, muteCount %d", muteCount); for (int i = 0; i < muteCount; i++) { setStreamMute(stream, starting, mPrimaryOutput); } } else { ALOGV("handleIncallSonification() high visibility"); if (outputDesc->device() & getDeviceForStrategy(STRATEGY_PHONE, true /*fromCache*/)) { ALOGV("handleIncallSonification() high visibility muted, muteCount %d", muteCount); for (int i = 0; i < muteCount; i++) { setStreamMute(stream, starting, mPrimaryOutput); } } if (starting) { mpClientInterface->startTone(AUDIO_POLICY_TONE_IN_CALL_NOTIFICATION, AUDIO_STREAM_VOICE_CALL); } else { mpClientInterface->stopTone(); } } } } } // --- SessionRoute class implementation void AudioPolicyManager::SessionRoute::log(const char* prefix) { ALOGI("%s[SessionRoute strm:0x%X, src:%d, sess:0x%X, dev:0x%X refs:%d act:%d", prefix, mStreamType, mSource, mSession, mDeviceDescriptor != 0 ? mDeviceDescriptor->type() : AUDIO_DEVICE_NONE, mRefCount, mActivityCount); } // --- SessionRouteMap class implementation bool AudioPolicyManager::SessionRouteMap::hasRoute(audio_session_t session) { return indexOfKey(session) >= 0 && valueFor(session)->mDeviceDescriptor != 0; } bool AudioPolicyManager::SessionRouteMap::hasRouteChanged(audio_session_t session) { if (indexOfKey(session) >= 0) { if (valueFor(session)->mChanged) { valueFor(session)->mChanged = false; return true; } } return false; } void AudioPolicyManager::SessionRouteMap::removeRoute(audio_session_t session) { sp route = indexOfKey(session) >= 0 ? valueFor(session) : 0; if (route != 0) { ALOG_ASSERT(route->mRefCount > 0); --route->mRefCount; if (route->mRefCount <= 0) { removeItem(session); } } } int AudioPolicyManager::SessionRouteMap::incRouteActivity(audio_session_t session) { sp route = indexOfKey(session) >= 0 ? valueFor(session) : 0; return route != 0 ? ++(route->mActivityCount) : -1; } int AudioPolicyManager::SessionRouteMap::decRouteActivity(audio_session_t session) { sp route = indexOfKey(session) >= 0 ? valueFor(session) : 0; if (route != 0 && route->mActivityCount > 0) { return --(route->mActivityCount); } else { return -1; } } void AudioPolicyManager::SessionRouteMap::log(const char* caption) { ALOGI("%s ----", caption); for(size_t index = 0; index < size(); index++) { valueAt(index)->log(" "); } } void AudioPolicyManager::SessionRouteMap::addRoute(audio_session_t session, audio_stream_type_t streamType, audio_source_t source, sp descriptor) { if (mMapType == MAPTYPE_INPUT && streamType != SessionRoute::STREAM_TYPE_NA) { ALOGE("Adding Output Route to InputRouteMap"); return; } else if (mMapType == MAPTYPE_OUTPUT && source != SessionRoute::SOURCE_TYPE_NA) { ALOGE("Adding Input Route to OutputRouteMap"); return; } sp route = indexOfKey(session) >= 0 ? valueFor(session) : 0; if (route != 0) { if ((route->mDeviceDescriptor == 0 && descriptor != 0) || (!route->mDeviceDescriptor->equals(descriptor))) { route->mChanged = true; } route->mRefCount++; route->mDeviceDescriptor = descriptor; } else { route = new AudioPolicyManager::SessionRoute(session, streamType, source, descriptor); route->mRefCount++; add(session, route); if (descriptor != 0) { route->mChanged = true; } } } void AudioPolicyManager::defaultAudioPolicyConfig(void) { sp module; sp profile; sp defaultInputDevice = new DeviceDescriptor(String8("builtin-mic"), AUDIO_DEVICE_IN_BUILTIN_MIC); mAvailableOutputDevices.add(mDefaultOutputDevice); mAvailableInputDevices.add(defaultInputDevice); module = new HwModule("primary"); profile = new IOProfile(String8("primary"), AUDIO_PORT_ROLE_SOURCE); profile->attach(module); profile->mSamplingRates.add(44100); profile->mFormats.add(AUDIO_FORMAT_PCM_16_BIT); profile->mChannelMasks.add(AUDIO_CHANNEL_OUT_STEREO); profile->mSupportedDevices.add(mDefaultOutputDevice); profile->mFlags = AUDIO_OUTPUT_FLAG_PRIMARY; module->mOutputProfiles.add(profile); profile = new IOProfile(String8("primary"), AUDIO_PORT_ROLE_SINK); profile->attach(module); profile->mSamplingRates.add(8000); profile->mFormats.add(AUDIO_FORMAT_PCM_16_BIT); profile->mChannelMasks.add(AUDIO_CHANNEL_IN_MONO); profile->mSupportedDevices.add(defaultInputDevice); module->mInputProfiles.add(profile); mHwModules.add(module); } audio_stream_type_t AudioPolicyManager::streamTypefromAttributesInt(const audio_attributes_t *attr) { // flags to stream type mapping if ((attr->flags & AUDIO_FLAG_AUDIBILITY_ENFORCED) == AUDIO_FLAG_AUDIBILITY_ENFORCED) { return AUDIO_STREAM_ENFORCED_AUDIBLE; } if ((attr->flags & AUDIO_FLAG_SCO) == AUDIO_FLAG_SCO) { return AUDIO_STREAM_BLUETOOTH_SCO; } if ((attr->flags & AUDIO_FLAG_BEACON) == AUDIO_FLAG_BEACON) { return AUDIO_STREAM_TTS; } // usage to stream type mapping switch (attr->usage) { case AUDIO_USAGE_MEDIA: case AUDIO_USAGE_GAME: case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE: return AUDIO_STREAM_MUSIC; case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY: if (isStreamActive(AUDIO_STREAM_ALARM)) { return AUDIO_STREAM_ALARM; } if (isStreamActive(AUDIO_STREAM_RING)) { return AUDIO_STREAM_RING; } if (isInCall()) { return AUDIO_STREAM_VOICE_CALL; } return AUDIO_STREAM_ACCESSIBILITY; case AUDIO_USAGE_ASSISTANCE_SONIFICATION: return AUDIO_STREAM_SYSTEM; case AUDIO_USAGE_VOICE_COMMUNICATION: return AUDIO_STREAM_VOICE_CALL; case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING: return AUDIO_STREAM_DTMF; case AUDIO_USAGE_ALARM: return AUDIO_STREAM_ALARM; case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE: return AUDIO_STREAM_RING; case AUDIO_USAGE_NOTIFICATION: case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST: case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT: case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED: case AUDIO_USAGE_NOTIFICATION_EVENT: return AUDIO_STREAM_NOTIFICATION; case AUDIO_USAGE_UNKNOWN: default: return AUDIO_STREAM_MUSIC; } } bool AudioPolicyManager::isValidAttributes(const audio_attributes_t *paa) { // has flags that map to a strategy? if ((paa->flags & (AUDIO_FLAG_AUDIBILITY_ENFORCED | AUDIO_FLAG_SCO | AUDIO_FLAG_BEACON)) != 0) { return true; } // has known usage? switch (paa->usage) { case AUDIO_USAGE_UNKNOWN: case AUDIO_USAGE_MEDIA: case AUDIO_USAGE_VOICE_COMMUNICATION: case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING: case AUDIO_USAGE_ALARM: case AUDIO_USAGE_NOTIFICATION: case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE: case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST: case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT: case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED: case AUDIO_USAGE_NOTIFICATION_EVENT: case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY: case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE: case AUDIO_USAGE_ASSISTANCE_SONIFICATION: case AUDIO_USAGE_GAME: case AUDIO_USAGE_VIRTUAL_SOURCE: break; default: return false; } return true; } bool AudioPolicyManager::isStrategyActive(const sp outputDesc, routing_strategy strategy, uint32_t inPastMs, nsecs_t sysTime) const { if ((sysTime == 0) && (inPastMs != 0)) { sysTime = systemTime(); } for (int i = 0; i < (int)AUDIO_STREAM_CNT; i++) { if (i == AUDIO_STREAM_PATCH) { continue; } if (((getStrategy((audio_stream_type_t)i) == strategy) || (NUM_STRATEGIES == strategy)) && outputDesc->isStreamActive((audio_stream_type_t)i, inPastMs, sysTime)) { return true; } } return false; } audio_policy_forced_cfg_t AudioPolicyManager::getForceUse(audio_policy_force_use_t usage) { return mEngine->getForceUse(usage); } bool AudioPolicyManager::isInCall() { return isStateInCall(mEngine->getPhoneState()); } bool AudioPolicyManager::isStateInCall(int state) { return is_state_in_call(state); } }; // namespace android