/* * Copyright (c) 2013-2016, The Linux Foundation. All rights reserved. * Not a contribution. * * 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. * * This file was modified by Dolby Laboratories, Inc. The portions of the * code that are surrounded by "DOLBY..." are copyrighted and * licensed separately, as follows: * * (C) 2015 Dolby Laboratories, Inc. * * 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 "AudioPolicyManagerCustom" //#define LOG_NDEBUG 0 //#define VERY_VERBOSE_LOGGING #ifdef VERY_VERBOSE_LOGGING #define ALOGVV ALOGV #else #define ALOGVV(a...) do { } while(0) #endif // A device mask for all audio output devices that are considered "remote" when evaluating // active output devices in isStreamActiveRemotely() #define APM_AUDIO_OUT_DEVICE_REMOTE_ALL AUDIO_DEVICE_OUT_REMOTE_SUBMIX // A device mask for all audio input and output devices where matching inputs/outputs on device // type alone is not enough: the address must match too #define APM_AUDIO_DEVICE_MATCH_ADDRESS_ALL (AUDIO_DEVICE_IN_REMOTE_SUBMIX | \ AUDIO_DEVICE_OUT_REMOTE_SUBMIX) #include #include #include #include #include #include #include #include #include "AudioPolicyManager.h" #include #ifdef DOLBY_ENABLE #include "DolbyAudioPolicy_impl.h" #endif // DOLBY_END namespace android { #ifdef VOICE_CONCURRENCY audio_output_flags_t AudioPolicyManagerCustom::getFallBackPath() { audio_output_flags_t flag = AUDIO_OUTPUT_FLAG_FAST; char propValue[PROPERTY_VALUE_MAX]; if (property_get("voice.conc.fallbackpath", propValue, NULL)) { if (!strncmp(propValue, "deep-buffer", 11)) { flag = AUDIO_OUTPUT_FLAG_DEEP_BUFFER; } else if (!strncmp(propValue, "fast", 4)) { flag = AUDIO_OUTPUT_FLAG_FAST; } else { ALOGD("voice_conc:not a recognised path(%s) in prop voice.conc.fallbackpath", propValue); } } else { ALOGD("voice_conc:prop voice.conc.fallbackpath not set"); } ALOGD("voice_conc:picked up flag(0x%x) from prop voice.conc.fallbackpath", flag); return flag; } #endif /*VOICE_CONCURRENCY*/ // ---------------------------------------------------------------------------- // AudioPolicyInterface implementation // ---------------------------------------------------------------------------- extern "C" AudioPolicyInterface* createAudioPolicyManager( AudioPolicyClientInterface *clientInterface) { return new AudioPolicyManagerCustom(clientInterface); } extern "C" void destroyAudioPolicyManager(AudioPolicyInterface *interface) { delete interface; } status_t AudioPolicyManagerCustom::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) { #ifdef AUDIO_EXTN_HDMI_SPK_ENABLED if ((popcount(device) == 1) && (device & AUDIO_DEVICE_OUT_AUX_DIGITAL)) { if (!strncmp(device_address, "hdmi_spkr", 9)) { mHdmiAudioDisabled = false; } else { mHdmiAudioEvent = true; } } #endif 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); #ifdef AUDIO_EXTN_HDMI_SPK_ENABLED if ((popcount(device) == 1) && (device & AUDIO_DEVICE_OUT_AUX_DIGITAL)) { if (!strncmp(device_address, "hdmi_spkr", 9)) { mHdmiAudioDisabled = false; } else { mHdmiAudioEvent = true; } if (mHdmiAudioDisabled || !mHdmiAudioEvent) { mAvailableOutputDevices.remove(devDesc); ALOGW("HDMI sink not connected, do not route audio to HDMI out"); return INVALID_OPERATION; } } #endif 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) { #ifdef AUDIO_EXTN_HDMI_SPK_ENABLED if ((popcount(device) == 1) && (device & AUDIO_DEVICE_OUT_AUX_DIGITAL)) { if (!strncmp(device_address, "hdmi_spkr", 9)) { mHdmiAudioDisabled = true; } else { mHdmiAudioEvent = false; } } #endif 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); #ifdef AUDIO_EXTN_HDMI_SPK_ENABLED if ((popcount(device) == 1) && (device & AUDIO_DEVICE_OUT_AUX_DIGITAL)) { if (!strncmp(device_address, "hdmi_spkr", 9)) { mHdmiAudioDisabled = true; } else { mHdmiAudioEvent = false; } } #endif 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(); #ifdef DOLBY_ENABLE // Before closing the opened outputs, update endpoint property with device capabilities audio_devices_t audioOutputDevice = getDeviceForStrategy(getStrategy(AUDIO_STREAM_MUSIC), true); mDolbyAudioPolicy.setEndpointSystemProperty(audioOutputDevice, mHwModules); #endif // DOLBY_END if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL && hasPrimaryOutput()) { audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/); updateCallRouting(newDevice); } #ifdef FM_POWER_OPT // handle FM device connection state to trigger FM AFE loopback if (device == AUDIO_DEVICE_OUT_FM && hasPrimaryOutput()) { audio_devices_t newDevice = AUDIO_DEVICE_NONE; if (state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) { mPrimaryOutput->changeRefCount(AUDIO_STREAM_MUSIC, 1); newDevice = (audio_devices_t)(getNewOutputDevice(mPrimaryOutput, false)|AUDIO_DEVICE_OUT_FM); mFMIsActive = true; } else { newDevice = (audio_devices_t)(getNewOutputDevice(mPrimaryOutput, false)); mFMIsActive = false; mPrimaryOutput->changeRefCount(AUDIO_STREAM_MUSIC, -1); } AudioParameter param = AudioParameter(); param.addInt(String8("handle_fm"), (int)newDevice); mpClientInterface->setParameters(mPrimaryOutput->mIoHandle, param.toString()); } #endif /* FM_POWER_OPT end */ 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(devDesc, 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(devDesc, 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 && hasPrimaryOutput()) { 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; } // 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 AudioPolicyManagerCustom::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); #ifdef VOICE_CONCURRENCY char concpropValue[PROPERTY_VALUE_MAX]; if (property_get("voice.playback.conc.disabled", concpropValue, NULL)) { bool propenabled = atoi(concpropValue) || !strncmp("true", concpropValue, 4); if (propenabled) { if (isInCall()) { ALOGD("\n copl: blocking compress offload on call mode\n"); return false; } } } #endif #ifdef RECORD_PLAY_CONCURRENCY char recConcPropValue[PROPERTY_VALUE_MAX]; bool prop_rec_play_enabled = false; if (property_get("rec.playback.conc.disabled", recConcPropValue, NULL)) { prop_rec_play_enabled = atoi(recConcPropValue) || !strncmp("true", recConcPropValue, 4); } if ((prop_rec_play_enabled) && ((true == mIsInputRequestOnProgress) || (mInputs.activeInputsCount() > 0))) { ALOGD("copl: blocking compress offload for record concurrency"); return false; } #endif // 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; } // Check if offload has been disabled bool offloadDisabled = property_get_bool("audio.offload.disable", false); if (offloadDisabled) { ALOGI("offload disabled by audio.offload.disable=%d", offloadDisabled); return false; } char propValue[PROPERTY_VALUE_MAX]; bool pcmOffload = false; #ifdef PCM_OFFLOAD_ENABLED if ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM_OFFLOAD) { bool prop_enabled = false; if ((AUDIO_FORMAT_PCM_16_BIT_OFFLOAD == offloadInfo.format) && property_get("audio.offload.pcm.16bit.enable", propValue, NULL)) { prop_enabled = atoi(propValue) || !strncmp("true", propValue, 4); } #ifdef PCM_OFFLOAD_ENABLED_24 if ((AUDIO_FORMAT_PCM_24_BIT_OFFLOAD == offloadInfo.format) && property_get("audio.offload.pcm.24bit.enable", propValue, NULL)) { prop_enabled = atoi(propValue) || !strncmp("true", propValue, 4); } #endif if (prop_enabled) { ALOGI("PCM offload property is enabled"); pcmOffload = true; } if (!pcmOffload) { ALOGD("system property not enabled for PCM offload format[%x]",offloadInfo.format); return false; } } #endif if (!pcmOffload) { bool compressedOffloadDisabled = property_get_bool("audio.offload.compress.disable", false); if (compressedOffloadDisabled) { ALOGI("compressed offload disabled by audio.offload.compress.disable=%d", compressedOffloadDisabled); return false; } //check if it's multi-channel AAC (includes sub formats) and FLAC format if ((popcount(offloadInfo.channel_mask) > 2) && (((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC) || ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_VORBIS))) { ALOGD("offload disabled for multi-channel AAC,FLAC and VORBIS format"); return false; } #ifdef AUDIO_EXTN_FORMATS_ENABLED //check if it's multi-channel FLAC/ALAC/WMA format with sample rate > 48k if ((popcount(offloadInfo.channel_mask) > 2) && (((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_FLAC) || (((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_ALAC) && (offloadInfo.sample_rate > 48000)) || (((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA) && (offloadInfo.sample_rate > 48000)) || (((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA_PRO) && (offloadInfo.sample_rate > 48000)) || ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC_ADTS))) { ALOGD("offload disabled for multi-channel FLAC/ALAC/WMA/AAC_ADTS clips with sample rate > 48kHz"); return false; } #endif //TODO: enable audio offloading with video when ready const bool allowOffloadWithVideo = property_get_bool("audio.offload.video", false /* default_value */); if (offloadInfo.has_video && !allowOffloadWithVideo) { ALOGV("isOffloadSupported: has_video == true, returning false"); return false; } const bool allowOffloadStreamingWithVideo = property_get_bool("av.streaming.offload.enable", false /*default value*/); if(offloadInfo.has_video && offloadInfo.is_streaming && !allowOffloadStreamingWithVideo) { ALOGW("offload disabled by av.streaming.offload.enable = %s ", propValue ); 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); //duration checks only valid for MP3/AAC/ formats, //do not check duration for other audio formats, e.g. dolby AAC/AC3 and amrwb+ formats if ((offloadInfo.format == AUDIO_FORMAT_MP3) || ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC) || ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_VORBIS) || #ifdef AUDIO_EXTN_FORMATS_ENABLED ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_FLAC) || ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA) || ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA_PRO) || ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_ALAC) || ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_APE) || ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC_ADTS) || #endif pcmOffload) 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; } // Check for soundcard status String8 valueStr = mpClientInterface->getParameters((audio_io_handle_t)0, String8("SND_CARD_STATUS")); AudioParameter result = AudioParameter(valueStr); int isonline = 0; if ((result.getInt(String8("SND_CARD_STATUS"), isonline) == NO_ERROR) && !isonline) { ALOGD("copl: soundcard is offline rejecting offload request"); 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); } audio_devices_t AudioPolicyManagerCustom::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)|| isStrategyActive(mPrimaryOutput, 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)|| (isStrategyActive(mPrimaryOutput,STRATEGY_SONIFICATION) && (!isStrategyActive(mPrimaryOutput,STRATEGY_MEDIA)))) { device = getDeviceForStrategy(STRATEGY_SONIFICATION, fromCache); } else if (isStrategyActive(outputDesc, STRATEGY_SONIFICATION_RESPECTFUL) || isStrategyActive(mPrimaryOutput,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; } void AudioPolicyManagerCustom::setPhoneState(audio_mode_t state) { ALOGV("setPhoneState() state %d", state); // store previous phone state for management of sonification strategy below audio_devices_t newDevice = AUDIO_DEVICE_NONE; 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 (isStateInCall(oldState)) { ALOGV("setPhoneState() in call state management: new state is %d", state); for (size_t j = 0; j < mOutputs.size(); j++) { audio_io_handle_t curOutput = mOutputs.keyAt(j); for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) { if (stream == AUDIO_STREAM_PATCH) { continue; } handleIncallSonification((audio_stream_type_t)stream, false, true, curOutput); } } // force reevaluating accessibility routing when call stops 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; #ifdef VOICE_CONCURRENCY char propValue[PROPERTY_VALUE_MAX]; bool prop_playback_enabled = false, prop_rec_enabled=false, prop_voip_enabled = false; if(property_get("voice.playback.conc.disabled", propValue, NULL)) { prop_playback_enabled = atoi(propValue) || !strncmp("true", propValue, 4); } if(property_get("voice.record.conc.disabled", propValue, NULL)) { prop_rec_enabled = atoi(propValue) || !strncmp("true", propValue, 4); } if(property_get("voice.voip.conc.disabled", propValue, NULL)) { prop_voip_enabled = atoi(propValue) || !strncmp("true", propValue, 4); } if ((AUDIO_MODE_IN_CALL != oldState) && (AUDIO_MODE_IN_CALL == state)) { ALOGD("voice_conc:Entering to call mode oldState :: %d state::%d ", oldState, state); mvoice_call_state = state; if (prop_rec_enabled) { //Close all active inputs audio_io_handle_t activeInput = mInputs.getActiveInput(); if (activeInput != 0) { sp activeDesc = mInputs.valueFor(activeInput); switch(activeDesc->mInputSource) { case AUDIO_SOURCE_VOICE_UPLINK: case AUDIO_SOURCE_VOICE_DOWNLINK: case AUDIO_SOURCE_VOICE_CALL: ALOGD("voice_conc:FOUND active input during call active: %d",activeDesc->mInputSource); break; case AUDIO_SOURCE_VOICE_COMMUNICATION: if(prop_voip_enabled) { ALOGD("voice_conc:CLOSING VoIP input source on call setup :%d ",activeDesc->mInputSource); stopInput(activeInput, activeDesc->mSessions.itemAt(0)); releaseInput(activeInput, activeDesc->mSessions.itemAt(0)); } break; default: ALOGD("voice_conc:CLOSING input on call setup for inputSource: %d",activeDesc->mInputSource); stopInput(activeInput, activeDesc->mSessions.itemAt(0)); releaseInput(activeInput, activeDesc->mSessions.itemAt(0)); break; } } } else if (prop_voip_enabled) { audio_io_handle_t activeInput = mInputs.getActiveInput(); if (activeInput != 0) { sp activeDesc = mInputs.valueFor(activeInput); if (AUDIO_SOURCE_VOICE_COMMUNICATION == activeDesc->mInputSource) { ALOGD("voice_conc:CLOSING VoIP on call setup : %d",activeDesc->mInputSource); stopInput(activeInput, activeDesc->mSessions.itemAt(0)); releaseInput(activeInput, activeDesc->mSessions.itemAt(0)); } } } if (prop_playback_enabled) { // Move tracks associated to this strategy from previous output to new output for (int i = AUDIO_STREAM_SYSTEM; i < (int)AUDIO_STREAM_CNT; i++) { ALOGV("voice_conc:Invalidate on call mode for stream :: %d ", i); if (i == AUDIO_STREAM_PATCH) { ALOGV("voice_conc:not calling invalidate for AUDIO_STREAM_PATCH"); continue; } if (AUDIO_OUTPUT_FLAG_DEEP_BUFFER == mFallBackflag) { if ((AUDIO_STREAM_MUSIC == i) || (AUDIO_STREAM_VOICE_CALL == i) ) { ALOGD("voice_conc:Invalidate stream type %d", i); mpClientInterface->invalidateStream((audio_stream_type_t)i); } } else if (AUDIO_OUTPUT_FLAG_FAST == mFallBackflag) { ALOGD("voice_conc:Invalidate stream type %d", i); mpClientInterface->invalidateStream((audio_stream_type_t)i); } } } for (size_t i = 0; i < mOutputs.size(); i++) { sp outputDesc = mOutputs.valueAt(i); if ( (outputDesc == NULL) || (outputDesc->mProfile == NULL)) { ALOGD("voice_conc:ouput desc / profile is NULL"); continue; } if (AUDIO_OUTPUT_FLAG_FAST == mFallBackflag) { if (((!outputDesc->isDuplicated() &&outputDesc->mProfile->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY)) && prop_playback_enabled) { ALOGD("voice_conc:calling suspendOutput on call mode for primary output"); mpClientInterface->suspendOutput(mOutputs.keyAt(i)); } //Close compress all sessions else if ((outputDesc->mProfile->mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) && prop_playback_enabled) { ALOGD("voice_conc:calling closeOutput on call mode for COMPRESS output"); closeOutput(mOutputs.keyAt(i)); } else if ((outputDesc->mProfile->mFlags & AUDIO_OUTPUT_FLAG_VOIP_RX) && prop_voip_enabled) { ALOGD("voice_conc:calling closeOutput on call mode for DIRECT output"); closeOutput(mOutputs.keyAt(i)); } } else if (AUDIO_OUTPUT_FLAG_DEEP_BUFFER == mFallBackflag) { if ((outputDesc->mProfile->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) && prop_playback_enabled) { ALOGD("voice_conc:calling closeOutput on call mode for COMPRESS output"); closeOutput(mOutputs.keyAt(i)); } } } } if ((AUDIO_MODE_IN_CALL == oldState || AUDIO_MODE_IN_COMMUNICATION == oldState) && (AUDIO_MODE_NORMAL == state) && prop_playback_enabled && mvoice_call_state) { ALOGD("voice_conc:EXITING from call mode oldState :: %d state::%d \n",oldState, state); mvoice_call_state = 0; if (AUDIO_OUTPUT_FLAG_FAST == mFallBackflag) { //restore PCM (deep-buffer) output after call termination for (size_t i = 0; i < mOutputs.size(); i++) { sp outputDesc = mOutputs.valueAt(i); if ( (outputDesc == NULL) || (outputDesc->mProfile == NULL)) { ALOGD("voice_conc:ouput desc / profile is NULL"); continue; } if (!outputDesc->isDuplicated() && outputDesc->mProfile->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY) { ALOGD("voice_conc:calling restoreOutput after call mode for primary output"); mpClientInterface->restoreOutput(mOutputs.keyAt(i)); } } } //call invalidate tracks so that any open streams can fall back to deep buffer/compress path from ULL for (int i = AUDIO_STREAM_SYSTEM; i < (int)AUDIO_STREAM_CNT; i++) { ALOGV("voice_conc:Invalidate on call mode for stream :: %d ", i); if (i == AUDIO_STREAM_PATCH) { ALOGV("voice_conc:not calling invalidate for AUDIO_STREAM_PATCH"); continue; } if (AUDIO_OUTPUT_FLAG_DEEP_BUFFER == mFallBackflag) { if ((AUDIO_STREAM_MUSIC == i) || (AUDIO_STREAM_VOICE_CALL == i) ) { mpClientInterface->invalidateStream((audio_stream_type_t)i); } } else if (AUDIO_OUTPUT_FLAG_FAST == mFallBackflag) { mpClientInterface->invalidateStream((audio_stream_type_t)i); } } } #endif #ifdef RECORD_PLAY_CONCURRENCY char recConcPropValue[PROPERTY_VALUE_MAX]; bool prop_rec_play_enabled = false; if (property_get("rec.playback.conc.disabled", recConcPropValue, NULL)) { prop_rec_play_enabled = atoi(recConcPropValue) || !strncmp("true", recConcPropValue, 4); } if (prop_rec_play_enabled) { if (AUDIO_MODE_IN_COMMUNICATION == mEngine->getPhoneState()) { ALOGD("phone state changed to MODE_IN_COMM invlaidating music and voice streams"); // call invalidate for voice streams, so that it can use deepbuffer with VoIP out device from HAL mpClientInterface->invalidateStream(AUDIO_STREAM_VOICE_CALL); // call invalidate for music, so that compress will fallback to deep-buffer with VoIP out device mpClientInterface->invalidateStream(AUDIO_STREAM_MUSIC); // close compress output to make sure session will be closed before timeout(60sec) for (size_t i = 0; i < mOutputs.size(); i++) { sp outputDesc = mOutputs.valueAt(i); if ((outputDesc == NULL) || (outputDesc->mProfile == NULL)) { ALOGD("ouput desc / profile is NULL"); continue; } if (outputDesc->mProfile->mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { ALOGD("calling closeOutput on call mode for COMPRESS output"); closeOutput(mOutputs.keyAt(i)); } } } else if ((oldState == AUDIO_MODE_IN_COMMUNICATION) && (mEngine->getPhoneState() == AUDIO_MODE_NORMAL)) { // call invalidate for music so that music can fallback to compress mpClientInterface->invalidateStream(AUDIO_STREAM_MUSIC); } } #endif mPrevPhoneState = oldState; 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*/)); } } if (hasPrimaryOutput()) { // 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 = mPrimaryOutput->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); } } //update device for all non-primary outputs for (size_t i = 0; i < mOutputs.size(); i++) { audio_io_handle_t output = mOutputs.keyAt(i); if (output != mPrimaryOutput->mIoHandle) { newDevice = getNewOutputDevice(mOutputs.valueFor(output), false /*fromCache*/); setOutputDevice(mOutputs.valueFor(output), newDevice, (newDevice != AUDIO_DEVICE_NONE)); } } // 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 (size_t j = 0; j < mOutputs.size(); j++) { audio_io_handle_t curOutput = mOutputs.keyAt(j); for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) { if (stream == AUDIO_STREAM_PATCH) { continue; } handleIncallSonification((audio_stream_type_t)stream, true, true, curOutput); } } // force reevaluating accessibility routing when call starts mpClientInterface->invalidateStream(AUDIO_STREAM_ACCESSIBILITY); } // 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; } } void AudioPolicyManagerCustom::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 && hasPrimaryOutput()) { audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, true /*fromCache*/); updateCallRouting(newDevice); } // Use reverse loop to make sure any low latency usecases (generally tones) // are not routed before non LL usecases (generally music). // We can safely assume that LL output would always have lower index, // and use this work-around to avoid routing of output with music stream // from the context of short lived LL output. // Note: in case output's share backend(HAL sharing is implicit) all outputs // gets routing update while processing first output itself. for (size_t i = mOutputs.size(); i > 0; i--) { sp outputDesc = mOutputs.valueAt(i-1); 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)); } } status_t AudioPolicyManagerCustom::stopSource(sp outputDesc, audio_stream_type_t stream, bool forceDeviceUpdate) { if (stream < 0 || stream >= AUDIO_STREAM_CNT) { ALOGW("stopSource() invalid stream %d", stream); return INVALID_OPERATION; } // 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()) { if (outputDesc->isDuplicated()) { handleIncallSonification(stream, false, false, outputDesc->subOutput1()->mIoHandle); handleIncallSonification(stream, false, false, outputDesc->subOutput2()->mIoHandle); } handleIncallSonification(stream, false, false, outputDesc->mIoHandle); } 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 || forceDeviceUpdate) { outputDesc->mStopTime[stream] = systemTime(); audio_devices_t prevDevice = outputDesc->device(); 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())) { audio_devices_t dev = getNewOutputDevice(mOutputs.valueFor(curOutput), false /*fromCache*/); uint32_t delayMs; if (dev == prevDevice) { delayMs = 0; } else { delayMs = outputDesc->latency()*2; } setOutputDevice(desc, dev, true, delayMs); } } // 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; } } status_t AudioPolicyManagerCustom::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; if (stream < 0 || stream >= AUDIO_STREAM_CNT) { ALOGW("startSource() invalid stream %d", stream); return INVALID_OPERATION; } *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, outputDesc->mIoHandle); } // 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); } } else { // handle special case for sonification while in call if (isInCall()) { handleIncallSonification(stream, true, false, outputDesc->mIoHandle); } } return NO_ERROR; } void AudioPolicyManagerCustom::handleIncallSonification(audio_stream_type_t stream, bool starting, bool stateChange, audio_io_handle_t output) { if(!hasPrimaryOutput()) { return; } // no action needed for AUDIO_STREAM_PATCH stream type, it's for internal flinger tracks if (stream == AUDIO_STREAM_PATCH) { return; } // 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 = mOutputs.valueFor(output); 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, outputDesc); } } 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, outputDesc); } } if (starting) { mpClientInterface->startTone(AUDIO_POLICY_TONE_IN_CALL_NOTIFICATION, AUDIO_STREAM_VOICE_CALL); } else { mpClientInterface->stopTone(); } } } } } void AudioPolicyManagerCustom::handleNotificationRoutingForStream(audio_stream_type_t stream) { switch(stream) { case AUDIO_STREAM_MUSIC: checkOutputForStrategy(STRATEGY_SONIFICATION_RESPECTFUL); updateDevicesAndOutputs(); break; default: break; } } status_t AudioPolicyManagerCustom::checkAndSetVolume(audio_stream_type_t stream, int index, const sp& outputDesc, audio_devices_t device, int delayMs, bool force) { if (stream < 0 || stream >= AUDIO_STREAM_CNT) { ALOGW("checkAndSetVolume() invalid stream %d", stream); return INVALID_OPERATION; } // 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) || isDirectOutput(outputDesc->mIoHandle) || device & AUDIO_DEVICE_OUT_ALL_USB)) { mpClientInterface->setVoiceVolume(voiceVolume, delayMs); mLastVoiceVolume = voiceVolume; } #ifdef FM_POWER_OPT } else if (stream == AUDIO_STREAM_MUSIC && hasPrimaryOutput() && outputDesc == mPrimaryOutput && mFMIsActive) { /* Avoid unnecessary set_parameter calls as it puts the primary outputs FastMixer in HOT_IDLE leading to breaks in audio */ if (volumeDb != mPrevFMVolumeDb) { mPrevFMVolumeDb = volumeDb; AudioParameter param = AudioParameter(); param.addFloat(String8("fm_volume"), Volume::DbToAmpl(volumeDb)); //Double delayMs to avoid sound burst while device switch. mpClientInterface->setParameters(mPrimaryOutput->mIoHandle, param.toString(), delayMs*2); } #endif /* FM_POWER_OPT end */ } return NO_ERROR; } bool AudioPolicyManagerCustom::isDirectOutput(audio_io_handle_t output) { for (size_t i = 0; i < mOutputs.size(); i++) { audio_io_handle_t curOutput = mOutputs.keyAt(i); sp desc = mOutputs.valueAt(i); if ((curOutput == output) && (desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { return true; } } return false; } status_t AudioPolicyManagerCustom::getOutputForAttr(const audio_attributes_t *attr, audio_io_handle_t *output, audio_session_t session, audio_stream_type_t *stream, uid_t uid, 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_offload_info_t tOffloadInfo = AUDIO_INFO_INITIALIZER; bool offloadDisabled = property_get_bool("audio.offload.disable", false); bool pcmOffloadEnabled = false; if (offloadDisabled) { ALOGI("offload disabled by audio.offload.disable=%d", offloadDisabled); } //read track offload property only if the global offload switch is off. if (!offloadDisabled) { pcmOffloadEnabled = property_get_bool("audio.offload.track.enable", false); } if (offloadInfo == NULL && pcmOffloadEnabled) { tOffloadInfo.sample_rate = samplingRate; tOffloadInfo.channel_mask = channelMask; tOffloadInfo.format = format; tOffloadInfo.stream_type = *stream; tOffloadInfo.bit_width = 16; //hard coded for PCM_16 if (attr != NULL) { ALOGV("found attribute .. setting usage %d ", attr->usage); tOffloadInfo.usage = attr->usage; } else { ALOGI("%s:: attribute is NULL .. no usage set", __func__); } offloadInfo = &tOffloadInfo; } return AudioPolicyManager::getOutputForAttr(attr, output, session, stream, (uid_t)uid, (uint32_t)samplingRate, format, (audio_channel_mask_t)channelMask, flags, (audio_port_handle_t)selectedDeviceId, offloadInfo); } audio_io_handle_t AudioPolicyManagerCustom::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 if (((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) && (stream != AUDIO_STREAM_MUSIC)) { // compress should not be used for non-music streams ALOGE("Offloading only allowed with music stream"); return 0; } if ((stream == AUDIO_STREAM_VOICE_CALL) && (channelMask == 1) && (samplingRate == 8000 || samplingRate == 16000 || samplingRate == 32000 || samplingRate == 48000)) { // Allow Voip direct output only if: // audio mode is MODE_IN_COMMUNCATION; AND // voip output is not opened already; AND // requested sample rate matches with that of voip input stream (if opened already) int value = 0; uint32_t mode = 0, voipOutCount = 1, voipSampleRate = 1; String8 valueStr = mpClientInterface->getParameters((audio_io_handle_t)0, String8("audio_mode")); AudioParameter result = AudioParameter(valueStr); if (result.getInt(String8("audio_mode"), value) == NO_ERROR) { mode = value; } valueStr = mpClientInterface->getParameters((audio_io_handle_t)0, String8("voip_out_stream_count")); result = AudioParameter(valueStr); if (result.getInt(String8("voip_out_stream_count"), value) == NO_ERROR) { voipOutCount = value; } valueStr = mpClientInterface->getParameters((audio_io_handle_t)0, String8("voip_sample_rate")); result = AudioParameter(valueStr); if (result.getInt(String8("voip_sample_rate"), value) == NO_ERROR) { voipSampleRate = value; } if ((mode == AUDIO_MODE_IN_COMMUNICATION) && (voipOutCount == 0) && ((voipSampleRate == 0) || (voipSampleRate == samplingRate))) { if (audio_is_linear_pcm(format)) { char propValue[PROPERTY_VALUE_MAX] = {0}; property_get("use.voice.path.for.pcm.voip", propValue, "0"); bool voipPcmSysPropEnabled = !strncmp("true", propValue, sizeof("true")); if (voipPcmSysPropEnabled && (format == AUDIO_FORMAT_PCM_16_BIT)) { flags = (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_VOIP_RX | AUDIO_OUTPUT_FLAG_DIRECT); ALOGD("Set VoIP and Direct output flags for PCM format"); } } } } #ifdef VOICE_CONCURRENCY char propValue[PROPERTY_VALUE_MAX]; bool prop_play_enabled=false, prop_voip_enabled = false; if(property_get("voice.playback.conc.disabled", propValue, NULL)) { prop_play_enabled = atoi(propValue) || !strncmp("true", propValue, 4); } if(property_get("voice.voip.conc.disabled", propValue, NULL)) { prop_voip_enabled = atoi(propValue) || !strncmp("true", propValue, 4); } if (prop_play_enabled && mvoice_call_state) { //check if voice call is active / running in background if((AUDIO_MODE_IN_CALL == mEngine->getPhoneState()) || ((AUDIO_MODE_IN_CALL == mPrevPhoneState) && (AUDIO_MODE_IN_COMMUNICATION == mEngine->getPhoneState()))) { if(AUDIO_OUTPUT_FLAG_VOIP_RX & flags) { if(prop_voip_enabled) { ALOGD("voice_conc:getoutput:IN call mode return no o/p for VoIP %x", flags ); return 0; } } else { if (AUDIO_OUTPUT_FLAG_FAST == mFallBackflag) { ALOGD("voice_conc:IN call mode adding ULL flags .. flags: %x ", flags ); flags = AUDIO_OUTPUT_FLAG_FAST; } else if (AUDIO_OUTPUT_FLAG_DEEP_BUFFER == mFallBackflag) { if (AUDIO_STREAM_MUSIC == stream) { flags = AUDIO_OUTPUT_FLAG_DEEP_BUFFER; ALOGD("voice_conc:IN call mode adding deep-buffer flags %x ", flags ); } else { flags = AUDIO_OUTPUT_FLAG_FAST; ALOGD("voice_conc:IN call mode adding fast flags %x ", flags ); } } } } } else if (prop_voip_enabled && mvoice_call_state) { //check if voice call is active / running in background //some of VoIP apps(like SIP2SIP call) supports resume of VoIP call when call in progress //return only ULL ouput if((AUDIO_MODE_IN_CALL == mEngine->getPhoneState()) || ((AUDIO_MODE_IN_CALL == mPrevPhoneState) && (AUDIO_MODE_IN_COMMUNICATION == mEngine->getPhoneState()))) { if(AUDIO_OUTPUT_FLAG_VOIP_RX & flags) { ALOGD("voice_conc:getoutput:IN call mode return no o/p for VoIP %x", flags ); return 0; } } } #endif #ifdef RECORD_PLAY_CONCURRENCY char recConcPropValue[PROPERTY_VALUE_MAX]; bool prop_rec_play_enabled = false; if (property_get("rec.playback.conc.disabled", recConcPropValue, NULL)) { prop_rec_play_enabled = atoi(recConcPropValue) || !strncmp("true", recConcPropValue, 4); } if ((prop_rec_play_enabled) && ((true == mIsInputRequestOnProgress) || (mInputs.activeInputsCount() > 0))) { if (AUDIO_MODE_IN_COMMUNICATION == mEngine->getPhoneState()) { if (AUDIO_OUTPUT_FLAG_VOIP_RX & flags) { // allow VoIP using voice path // Do nothing } else if((flags & AUDIO_OUTPUT_FLAG_FAST) == 0) { ALOGD("voice_conc:MODE_IN_COMM is setforcing deep buffer output for non ULL... flags: %x", flags); // use deep buffer path for all non ULL outputs flags = AUDIO_OUTPUT_FLAG_DEEP_BUFFER; } } else if ((flags & AUDIO_OUTPUT_FLAG_FAST) == 0) { ALOGD("voice_conc:Record mode is on forcing deep buffer output for non ULL... flags: %x ", flags); // use deep buffer path for all non ULL outputs flags = AUDIO_OUTPUT_FLAG_DEEP_BUFFER; } } if (prop_rec_play_enabled && (stream == AUDIO_STREAM_ENFORCED_AUDIBLE)) { ALOGD("Record conc is on forcing ULL output for ENFORCED_AUDIBLE"); flags = AUDIO_OUTPUT_FLAG_FAST; } #endif #ifdef AUDIO_EXTN_AFE_PROXY_ENABLED /* * WFD audio routes back to target speaker when starting a ringtone playback. * This is because primary output is reused for ringtone, so output device is * updated based on SONIFICATION strategy for both ringtone and music playback. * The same issue is not seen on remoted_submix HAL based WFD audio because * primary output is not reused and a new output is created for ringtone playback. * Issue is fixed by updating output flag to AUDIO_OUTPUT_FLAG_FAST when there is * a non-music stream playback on WFD, so primary output is not reused for ringtone. */ audio_devices_t availableOutputDeviceTypes = mAvailableOutputDevices.types(); if ((availableOutputDeviceTypes & AUDIO_DEVICE_OUT_PROXY) && (stream != AUDIO_STREAM_MUSIC)) { ALOGD("WFD audio: use OUTPUT_FLAG_FAST for non music stream. flags:%x", flags ); //For voip paths if(flags & AUDIO_OUTPUT_FLAG_DIRECT) flags = AUDIO_OUTPUT_FLAG_DIRECT; else //route every thing else to ULL path flags = AUDIO_OUTPUT_FLAG_FAST; } #endif // 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); } bool forced_deep = false; // only allow deep buffering for music stream type if (stream != AUDIO_STREAM_MUSIC) { flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER); } else if (/* stream == AUDIO_STREAM_MUSIC && */ flags == AUDIO_OUTPUT_FLAG_NONE && property_get_bool("audio.deep_buffer.media", false /* default_value */)) { forced_deep = true; } if (stream == AUDIO_STREAM_TTS) { flags = AUDIO_OUTPUT_FLAG_TTS; } // check if direct output for track offload already exits bool is_track_offload_active = false; for (size_t i = 0; i < mOutputs.size(); i++) { sp desc = mOutputs.valueAt(i); if (desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT_PCM) { is_track_offload_active = true; ALOGD("Track offload already active"); break; } } // Do offload magic here if ((flags == AUDIO_OUTPUT_FLAG_NONE) && (stream == AUDIO_STREAM_MUSIC) && (offloadInfo != NULL) && !is_track_offload_active && ((offloadInfo->usage == AUDIO_USAGE_MEDIA) || (offloadInfo->usage == AUDIO_USAGE_GAME))) { flags = (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_DIRECT_PCM); ALOGD("AudioCustomHAL --> Force Direct Flag .. flag (0x%x)", flags); } 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|AUDIO_OUTPUT_FLAG_DIRECT_PCM)) == 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. // // Supplementary annotation: // For sake of track offload introduced, we need a rollback for both compress offload // and track offload use cases. if ((flags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_DIRECT_PCM)) && mEffects.isNonOffloadableEffectEnabled()) { ALOGD("non offloadable effect is enabled, try with non direct output"); goto non_direct_output; } profile = getProfileForDirectOutput(device, samplingRate, format, channelMask, (audio_output_flags_t)flags); if (profile != 0) { if (!(flags & AUDIO_OUTPUT_FLAG_DIRECT_PCM) && (profile->mFlags & AUDIO_OUTPUT_FLAG_DIRECT_PCM)) { ALOGI("got Direct_PCM without requesting ... reject "); profile = NULL; goto non_direct_output; } sp outputDesc = NULL; // if multiple concurrent offload decode is supported // do no check for reuse and also don't close previous output if its offload // previous output will be closed during track destruction if (!(property_get_bool("audio.offload.multiple.enabled", false) && ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0))) { 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); } } // if the selected profile is offloaded and no offload info was specified, // create a default one audio_offload_info_t defaultOffloadInfo = AUDIO_INFO_INITIALIZER; if ((profile->mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) && !offloadInfo) { flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD); defaultOffloadInfo.sample_rate = samplingRate; defaultOffloadInfo.channel_mask = channelMask; defaultOffloadInfo.format = format; defaultOffloadInfo.stream_type = stream; defaultOffloadInfo.bit_rate = 0; defaultOffloadInfo.duration_us = -1; defaultOffloadInfo.has_video = true; // conservative defaultOffloadInfo.is_streaming = true; // likely offloadInfo = &defaultOffloadInfo; } 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; } 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) { #ifdef DOLBY_ENABLE status_t status = mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, srcOutput, dstOutput); if (status == NO_ERROR) { for (size_t i = 0; i < mEffects.size(); i++) { sp desc = mEffects.valueAt(i); if (desc->mSession == AUDIO_SESSION_OUTPUT_MIX) { // update the mIo member of EffectDescriptor for the global effect ALOGV("%s updating mIo", __FUNCTION__); desc->mIo = dstOutput; } } } else { ALOGW("%s moveEffects from %d to %d failed", __FUNCTION__, srcOutput, dstOutput); } #else // DOLBY_END mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, srcOutput, dstOutput); #endif // LINE_ADDED_BY_DOLBY } 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)) { if (forced_deep) { flags = (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_DEEP_BUFFER); ALOGI("setting force DEEP buffer now "); } // 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; } status_t AudioPolicyManagerCustom::getInputForAttr(const audio_attributes_t *attr, audio_io_handle_t *input, audio_session_t session, uid_t uid, 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) { audio_source_t inputSource = attr->source; #ifdef VOICE_CONCURRENCY char propValue[PROPERTY_VALUE_MAX]; bool prop_rec_enabled=false, prop_voip_enabled = false; if(property_get("voice.record.conc.disabled", propValue, NULL)) { prop_rec_enabled = atoi(propValue) || !strncmp("true", propValue, 4); } if(property_get("voice.voip.conc.disabled", propValue, NULL)) { prop_voip_enabled = atoi(propValue) || !strncmp("true", propValue, 4); } if (prop_rec_enabled && mvoice_call_state) { //check if voice call is active / running in background //some of VoIP apps(like SIP2SIP call) supports resume of VoIP call when call in progress //Need to block input request if((AUDIO_MODE_IN_CALL == mEngine->getPhoneState()) || ((AUDIO_MODE_IN_CALL == mPrevPhoneState) && (AUDIO_MODE_IN_COMMUNICATION == mEngine->getPhoneState()))) { switch(inputSource) { case AUDIO_SOURCE_VOICE_UPLINK: case AUDIO_SOURCE_VOICE_DOWNLINK: case AUDIO_SOURCE_VOICE_CALL: ALOGD("voice_conc:Creating input during incall mode for inputSource: %d", inputSource); break; case AUDIO_SOURCE_VOICE_COMMUNICATION: if(prop_voip_enabled) { ALOGD("voice_conc:BLOCK VoIP requst incall mode for inputSource: %d", inputSource); return NO_INIT; } break; default: ALOGD("voice_conc:BLOCK VoIP requst incall mode for inputSource: %d", inputSource); return NO_INIT; } } }//check for VoIP flag else if(prop_voip_enabled && mvoice_call_state) { //check if voice call is active / running in background //some of VoIP apps(like SIP2SIP call) supports resume of VoIP call when call in progress //Need to block input request if((AUDIO_MODE_IN_CALL == mEngine->getPhoneState()) || ((AUDIO_MODE_IN_CALL == mPrevPhoneState) && (AUDIO_MODE_IN_COMMUNICATION == mEngine->getPhoneState()))) { if(inputSource == AUDIO_SOURCE_VOICE_COMMUNICATION) { ALOGD("BLOCKING VoIP request during incall mode for inputSource: %d ",inputSource); return NO_INIT; } } } #endif return AudioPolicyManager::getInputForAttr(attr, input, session, uid, samplingRate, format, channelMask, flags, selectedDeviceId, inputType); } status_t AudioPolicyManagerCustom::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) && !activeDesc->hasPreemptedSession(session)) { ALOGW("startInput(%d) preempting low-priority input %d", input, activeInput); audio_session_t activeSession = activeDesc->mSessions.itemAt(0); SortedVector sessions = activeDesc->getPreemptedSessions(); sessions.add(activeSession); inputDesc->setPreemptedSessions(sessions); stopInput(activeInput, activeSession); releaseInput(activeInput, activeSession); } else { ALOGE("startInput(%d) failed: other input %d already started", input, activeInput); return INVALID_OPERATION; } } } // Routing? mInputRoutes.incRouteActivity(session); #ifdef RECORD_PLAY_CONCURRENCY mIsInputRequestOnProgress = true; char getPropValue[PROPERTY_VALUE_MAX]; bool prop_rec_play_enabled = false; if (property_get("rec.playback.conc.disabled", getPropValue, NULL)) { prop_rec_play_enabled = atoi(getPropValue) || !strncmp("true", getPropValue, 4); } if ((prop_rec_play_enabled) &&(mInputs.activeInputsCount() == 0)){ // send update to HAL on record playback concurrency AudioParameter param = AudioParameter(); param.add(String8("rec_play_conc_on"), String8("true")); ALOGD("startInput() setParameters rec_play_conc is setting to ON "); mpClientInterface->setParameters(0, param.toString()); // Call invalidate to reset all opened non ULL audio tracks // Move tracks associated to this strategy from previous output to new output for (int i = AUDIO_STREAM_SYSTEM; i < (int)AUDIO_STREAM_CNT; i++) { // Do not call invalidate for ENFORCED_AUDIBLE (otherwise pops are seen for camcorder) if ((i != AUDIO_STREAM_ENFORCED_AUDIBLE && (i != AUDIO_STREAM_PATCH))) { ALOGD("Invalidate on releaseInput for stream :: %d ", i); //FIXME see fixme on name change mpClientInterface->invalidateStream((audio_stream_type_t)i); } } // close compress tracks for (size_t i = 0; i < mOutputs.size(); i++) { sp outputDesc = mOutputs.valueAt(i); if ((outputDesc == NULL) || (outputDesc->mProfile == NULL)) { ALOGD("ouput desc / profile is NULL"); continue; } if (outputDesc->mProfile->mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { // close compress sessions ALOGD("calling closeOutput on record conc for COMPRESS output"); closeOutput(mOutputs.keyAt(i)); } } } #endif 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->mCbFlags & AudioMix::kCbFlagNotifyActivity) != 0)) { mpClientInterface->onDynamicPolicyMixStateUpdate(inputDesc->mPolicyMix->mRegistrationId, MIX_STATE_MIXING); } 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++; #ifdef RECORD_PLAY_CONCURRENCY mIsInputRequestOnProgress = false; #endif return NO_ERROR; } status_t AudioPolicyManagerCustom::stopInput(audio_io_handle_t input, audio_session_t session) { status_t status; status = AudioPolicyManager::stopInput(input, session); #ifdef RECORD_PLAY_CONCURRENCY char propValue[PROPERTY_VALUE_MAX]; bool prop_rec_play_enabled = false; if (property_get("rec.playback.conc.disabled", propValue, NULL)) { prop_rec_play_enabled = atoi(propValue) || !strncmp("true", propValue, 4); } if ((prop_rec_play_enabled) && (mInputs.activeInputsCount() == 0)) { //send update to HAL on record playback concurrency AudioParameter param = AudioParameter(); param.add(String8("rec_play_conc_on"), String8("false")); ALOGD("stopInput() setParameters rec_play_conc is setting to OFF "); mpClientInterface->setParameters(0, param.toString()); //call invalidate tracks so that any open streams can fall back to deep buffer/compress path from ULL for (int i = AUDIO_STREAM_SYSTEM; i < (int)AUDIO_STREAM_CNT; i++) { //Do not call invalidate for ENFORCED_AUDIBLE (otherwise pops are seen for camcorder stop tone) if ((i != AUDIO_STREAM_ENFORCED_AUDIBLE) && (i != AUDIO_STREAM_PATCH)) { ALOGD(" Invalidate on stopInput for stream :: %d ", i); //FIXME see fixme on name change mpClientInterface->invalidateStream((audio_stream_type_t)i); } } } #endif return status; } void AudioPolicyManagerCustom::closeAllInputs() { bool patchRemoved = false; for(size_t input_index = mInputs.size(); input_index > 0; input_index--) { sp inputDesc = mInputs.valueAt(input_index-1); 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; } if ((inputDesc->mIsSoundTrigger) && (mInputs.size() == 1)) { ALOGD("Do not close sound trigger input handle"); } else { mpClientInterface->closeInput(mInputs.keyAt(input_index-1)); mInputs.removeItem(mInputs.keyAt(input_index-1)); } } nextAudioPortGeneration(); if (patchRemoved) { mpClientInterface->onAudioPatchListUpdate(); } } AudioPolicyManagerCustom::AudioPolicyManagerCustom(AudioPolicyClientInterface *clientInterface) : AudioPolicyManager(clientInterface), mHdmiAudioDisabled(false), mHdmiAudioEvent(false), mPrevPhoneState(0), mPrevFMVolumeDb(0.0f), mFMIsActive(false) { char ssr_enabled[PROPERTY_VALUE_MAX] = {0}; bool prop_ssr_enabled = false; if (property_get("ro.qc.sdk.audio.ssr", ssr_enabled, NULL)) { prop_ssr_enabled = atoi(ssr_enabled) || !strncmp("true", ssr_enabled, 4); } for (size_t i = 0; i < mHwModules.size(); i++) { ALOGV("Hw module %d", i); for (size_t j = 0; j < mHwModules[i]->mInputProfiles.size(); j++) { const sp inProfile = mHwModules[i]->mInputProfiles[j]; ALOGV("Input profile ", j); for (size_t k = 0; k < inProfile->mChannelMasks.size(); k++) { audio_channel_mask_t channelMask = inProfile->mChannelMasks.itemAt(k); ALOGV("Channel Mask %x size %d", channelMask, inProfile->mChannelMasks.size()); if (AUDIO_CHANNEL_IN_5POINT1 == channelMask) { if (!prop_ssr_enabled) { ALOGI("removing AUDIO_CHANNEL_IN_5POINT1 from" " input profile as SSR(surround sound record)" " is not supported on this chipset variant"); inProfile->mChannelMasks.removeItemsAt(k, 1); ALOGV("Channel Mask size now %d", inProfile->mChannelMasks.size()); } } } } } #ifdef RECORD_PLAY_CONCURRENCY mIsInputRequestOnProgress = false; #endif #ifdef VOICE_CONCURRENCY mFallBackflag = getFallBackPath(); #endif } }