| /* |
| * Copyright (c) 2013-2015, 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 |
| |
| #define MIN(a, b) ((a) < (b) ? (a) : (b)) |
| |
| // 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) |
| // Following delay should be used if the calculated routing delay from all active |
| // input streams is higher than this value |
| #define MAX_VOICE_CALL_START_DELAY_MS 100 |
| |
| #include <inttypes.h> |
| #include <math.h> |
| |
| #include <cutils/properties.h> |
| #include <utils/Log.h> |
| #include <hardware/audio.h> |
| #include <hardware/audio_effect.h> |
| #include <media/AudioParameter.h> |
| #include <soundtrigger/SoundTrigger.h> |
| #include "AudioPolicyManager.h" |
| #include <policy.h> |
| #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<DeviceDescriptor> devDesc = |
| mHwModules.getDeviceDescriptor(device, device_address, device_name); |
| |
| // handle output devices |
| if (audio_is_output_device(device)) { |
| SortedVector <audio_io_handle_t> 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<HwModule> 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<SwAudioOutputDescriptor> 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 = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/); |
| if (state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) { |
| mPrimaryOutput->changeRefCount(AUDIO_STREAM_MUSIC, 1); |
| newDevice = newDevice | AUDIO_DEVICE_OUT_FM; |
| } else { |
| 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<SwAudioOutputDescriptor> 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 <audio_io_handle_t> 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<HwModule> 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 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; |
| } |
| //check if it's multi-channel AAC (includes sub formats) and FLAC and VORBIS format |
| if ((popcount(offloadInfo.channel_mask) > 2) && |
| (((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC) || |
| ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_FLAC) || |
| ((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))) { |
| ALOGD("offload disabled for multi-channel FLAC/ALAC/WMA 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; |
| } |
| |
| //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/VORBIS/WMA/ALAC/APE 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) |
| #endif |
| ) |
| 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<IOProfile> profile = getProfileForDirectOutput(AUDIO_DEVICE_NONE /*ignore device */, |
| offloadInfo.sample_rate, |
| offloadInfo.format, |
| offloadInfo.channel_mask, |
| AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD); |
| ALOGV("isOffloadSupported() profile %sfound", profile != 0 ? "" : "NOT "); |
| return (profile != 0); |
| } |
| audio_devices_t AudioPolicyManagerCustom::getNewOutputDevice(const sp<AudioOutputDescriptor>& outputDesc, |
| bool fromCache) |
| { |
| audio_devices_t device = AUDIO_DEVICE_NONE; |
| |
| ssize_t index = mAudioPatches.indexOfKey(outputDesc->mPatchHandle); |
| if (index >= 0) { |
| sp<AudioPatch> 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) |
| && (!isStrategyActive(mPrimaryOutput, STRATEGY_MEDIA)))) { |
| 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 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<SwAudioOutputDescriptor> hwOutputDesc = mPrimaryOutput; |
| #ifdef VOICE_CONCURRENCY |
| int voice_call_state = 0; |
| 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); |
| } |
| |
| bool mode_in_call = (AUDIO_MODE_IN_CALL != oldState) && (AUDIO_MODE_IN_CALL == state); |
| //query if it is a actual voice call initiated by telephony |
| if (mode_in_call) { |
| String8 valueStr = mpClientInterface->getParameters((audio_io_handle_t)0, String8("in_call")); |
| AudioParameter result = AudioParameter(valueStr); |
| if (result.getInt(String8("in_call"), voice_call_state) == NO_ERROR) |
| ALOGD("voice_conc:SetPhoneState: Voice call state = %d", voice_call_state); |
| } |
| |
| if (mode_in_call && voice_call_state && !mvoice_call_state) { |
| ALOGD("voice_conc:Entering to call mode oldState :: %d state::%d ", |
| oldState, state); |
| mvoice_call_state = voice_call_state; |
| if (prop_rec_enabled) { |
| //Close all active inputs |
| audio_io_handle_t activeInput = mInputs.getActiveInput(); |
| if (activeInput != 0) { |
| sp<AudioInputDescriptor> 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<AudioInputDescriptor> 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<SwAudioOutputDescriptor> 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<SwAudioOutputDescriptor> 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<SwAudioOutputDescriptor> 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<SwAudioOutputDescriptor> 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*/)); |
| } |
| ALOGV("Setting the delay from %dms to %dms", delayMs, |
| MIN(delayMs, MAX_VOICE_CALL_START_DELAY_MS)); |
| delayMs = MIN(delayMs, MAX_VOICE_CALL_START_DELAY_MS); |
| } |
| |
| 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); |
| } |
| } |
| } |
| |
| // 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<SwAudioOutputDescriptor> 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<AudioOutputDescriptor> outputDesc1, |
| 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 |
| #ifdef NON_WEARABLE_TARGET |
| sp<AudioOutputDescriptor> outputDesc = outputDesc1; |
| #else |
| sp<SwAudioOutputDescriptor> outputDesc = (sp<SwAudioOutputDescriptor>) outputDesc1; |
| #endif |
| handleEventForBeacon(stream == AUDIO_STREAM_TTS ? STOPPING_BEACON : STOPPING_OUTPUT); |
| |
| // handle special case for sonification while in call |
| if (isInCall() && (outputDesc->mRefCount[stream] == 1)) { |
| if (outputDesc->isDuplicated()) { |
| #ifdef NON_WEARABLE_TARGET |
| handleIncallSonification(stream, false, false, outputDesc->subOutput1()->mIoHandle); |
| handleIncallSonification(stream, false, false, outputDesc->subOutput2()->mIoHandle); |
| #else |
| handleIncallSonification(stream, false, false, outputDesc->mOutput1->mIoHandle); |
| handleIncallSonification(stream, false, false, outputDesc->mOutput2->mIoHandle); |
| #endif |
| } |
| 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 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<AudioOutputDescriptor> 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*/); |
| setOutputDevice(desc, |
| dev, |
| 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; |
| } |
| } |
| status_t AudioPolicyManagerCustom::startSource(sp<AudioOutputDescriptor> outputDesc1, |
| 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; |
| } |
| |
| #ifdef NON_WEARABLE_TARGET |
| sp<AudioOutputDescriptor> outputDesc = outputDesc1; |
| #else |
| sp<SwAudioOutputDescriptor> outputDesc = (sp<SwAudioOutputDescriptor>) outputDesc1; |
| #endif |
| |
| *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<AudioOutputDescriptor> desc = mOutputs.valueAt(i); |
| if (desc != outputDesc) { |
| // force a device change if any other output is managed by the same hw |
| // module and has a current device selection that differs from selected device. |
| // In this case, the audio HAL must receive the new device selection so that it can |
| // change the device currently selected by the other active output. |
| if (outputDesc->sharesHwModuleWith(desc) && |
| desc->device() != 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<SwAudioOutputDescriptor> 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; |
| } |
| } |
| #ifdef NON_WEARABLE_TARGET |
| status_t AudioPolicyManagerCustom::checkAndSetVolume(audio_stream_type_t stream, |
| int index, |
| const sp<AudioOutputDescriptor>& outputDesc, |
| audio_devices_t device, |
| int delayMs, bool force) |
| #else |
| status_t AudioPolicyManagerCustom::checkAndSetVolume(audio_stream_type_t stream, |
| int index, |
| const sp<SwAudioOutputDescriptor>& outputDesc, |
| audio_devices_t device, |
| int delayMs, bool force) |
| |
| #endif |
| { |
| 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) { |
| AudioParameter param = AudioParameter(); |
| param.addFloat(String8("fm_volume"), Volume::DbToAmpl(volumeDb)); |
| mpClientInterface->setParameters(mPrimaryOutput->mIoHandle, param.toString(), delayMs); |
| #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<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i); |
| if ((curOutput == output) && (desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { |
| return true; |
| } |
| } |
| return false; |
| } |
| 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<AudioOutputDescriptor> 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; |
| } |
| |
| #ifdef COMPRESS_VOIP_ENABLED |
| if ((stream == AUDIO_STREAM_VOICE_CALL) && |
| (channelMask == 1) && |
| (samplingRate == 8000 || samplingRate == 16000)) { |
| // 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)((flags &~AUDIO_OUTPUT_FLAG_FAST) | |
| AUDIO_OUTPUT_FLAG_VOIP_RX | AUDIO_OUTPUT_FLAG_DIRECT); |
| ALOGD("Set VoIP and Direct output flags for PCM format"); |
| } |
| } |
| } |
| } |
| #endif |
| |
| #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); |
| } |
| // only allow deep buffering for music stream type |
| if (stream != AUDIO_STREAM_MUSIC) { |
| flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER); |
| } |
| if (stream == AUDIO_STREAM_TTS) { |
| flags = AUDIO_OUTPUT_FLAG_TTS; |
| } |
| |
| // 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); |
| } |
| if (stream == AUDIO_STREAM_TTS) { |
| flags = AUDIO_OUTPUT_FLAG_TTS; |
| } |
| |
| sp<IOProfile> 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<SwAudioOutputDescriptor> outputDesc = NULL; |
| |
| for (size_t i = 0; i < mOutputs.size(); i++) { |
| sp<SwAudioOutputDescriptor> 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<EffectDescriptor> 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)) { |
| // get which output is suitable for the specified stream. The actual |
| // routing change will happen when startOutput() will be called |
| SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(device, mOutputs); |
| |
| // 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<AudioInputDescriptor> 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<AudioInputDescriptor> 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; |
| } |
| } |
| } |
| |
| // 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<SwAudioOutputDescriptor> 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; |
| } |
| |
| AudioPolicyManagerCustom::AudioPolicyManagerCustom(AudioPolicyClientInterface *clientInterface) |
| : AudioPolicyManager(clientInterface) |
| { |
| #ifdef RECORD_PLAY_CONCURRENCY |
| mIsInputRequestOnProgress = false; |
| #endif |
| |
| |
| #ifdef VOICE_CONCURRENCY |
| mFallBackflag = getFallBackPath(); |
| #endif |
| } |
| } |
