| /* |
| * Copyright (c) 2013-2019 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. |
| */ |
| |
| #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) |
| #define SAMPLE_RATE_8000 8000 |
| #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> |
| |
| namespace android { |
| /*audio policy: workaround for truncated touch sounds*/ |
| //FIXME: workaround for truncated touch sounds |
| // to be removed when the problem is handled by system UI |
| #define TOUCH_SOUND_FIXED_DELAY_MS 100 |
| |
| sp<APMConfigHelper> AudioPolicyManagerCustom::mApmConfigs = new APMConfigHelper(); |
| |
| audio_output_flags_t AudioPolicyManagerCustom::getFallBackPath() |
| { |
| audio_output_flags_t flag = AUDIO_OUTPUT_FLAG_FAST; |
| const char *fallback_path = mApmConfigs->getVoiceConcFallbackPath().c_str(); |
| |
| if (strlen(fallback_path) > 0) { |
| if (!strncmp(fallback_path, "deep-buffer", 11)) { |
| flag = AUDIO_OUTPUT_FLAG_DEEP_BUFFER; |
| } |
| else if (!strncmp(fallback_path, "fast", 4)) { |
| flag = AUDIO_OUTPUT_FLAG_FAST; |
| } |
| else { |
| ALOGD("voice_conc:not a recognised path(%s) in prop vendor.voice.conc.fallbackpath", |
| fallback_path); |
| } |
| } |
| else { |
| ALOGD("voice_conc:prop vendor.voice.conc.fallbackpath not set"); |
| } |
| |
| ALOGD("voice_conc:picked up flag(0x%x) from prop vendor.voice.conc.fallbackpath", |
| flag); |
| |
| return flag; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // 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 deviceType, |
| audio_policy_dev_state_t state, |
| const char *device_address, |
| const char *device_name, |
| audio_format_t encodedFormat) |
| { |
| ALOGD("setDeviceConnectionStateInt() device: 0x%X, state %d, address %s name %s format 0x%X", |
| deviceType, state, device_address, device_name, encodedFormat); |
| |
| // connect/disconnect only 1 device at a time |
| if (!audio_is_output_device(deviceType) && !audio_is_input_device(deviceType)) return BAD_VALUE; |
| |
| sp<DeviceDescriptor> device = |
| mHwModules.getDeviceDescriptor(deviceType, device_address, device_name, encodedFormat, |
| state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE); |
| if (device == 0) { |
| return INVALID_OPERATION; |
| } |
| |
| // handle output devices |
| if (audio_is_output_device(deviceType)) { |
| SortedVector <audio_io_handle_t> outputs; |
| |
| ssize_t index = mAvailableOutputDevices.indexOf(device); |
| |
| // 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) { |
| if (mApmConfigs->isHDMISpkEnabled() && |
| (popcount(deviceType) == 1) && (deviceType & AUDIO_DEVICE_OUT_AUX_DIGITAL)) { |
| if (!strncmp(device_address, "hdmi_spkr", 9)) { |
| mHdmiAudioDisabled = false; |
| } else { |
| mHdmiAudioEvent = true; |
| } |
| } |
| ALOGW("setDeviceConnectionState() device already connected: %x", deviceType); |
| return INVALID_OPERATION; |
| } |
| ALOGV("%s() connecting device %s format %x", |
| __func__, device->toString().c_str(), encodedFormat); |
| |
| // register new device as available |
| index = mAvailableOutputDevices.add(device); |
| if (mApmConfigs->isHDMISpkEnabled() && |
| (popcount(deviceType) == 1) && (deviceType & AUDIO_DEVICE_OUT_AUX_DIGITAL)) { |
| if (!strncmp(device_address, "hdmi_spkr", 9)) { |
| mHdmiAudioDisabled = false; |
| } else { |
| mHdmiAudioEvent = true; |
| } |
| if (mHdmiAudioDisabled || !mHdmiAudioEvent) { |
| mAvailableOutputDevices.remove(device); |
| ALOGW("HDMI sink not connected, do not route audio to HDMI out"); |
| return INVALID_OPERATION; |
| } |
| } |
| if (index >= 0) { |
| sp<HwModule> module = mHwModules.getModuleForDevice(device, encodedFormat); |
| if (module == 0) { |
| ALOGD("setDeviceConnectionState() could not find HW module for device %s", |
| device->toString().c_str()); |
| mAvailableOutputDevices.remove(device); |
| return INVALID_OPERATION; |
| } |
| ALOGV("setDeviceConnectionState() module name=%s", module->getName()); |
| mAvailableOutputDevices[index]->attach(module); |
| } else { |
| return NO_MEMORY; |
| } |
| |
| // Before checking outputs, broadcast connect event to allow HAL to retrieve dynamic |
| // parameters on newly connected devices (instead of opening the outputs...) |
| broadcastDeviceConnectionState(device, state); |
| |
| if (checkOutputsForDevice(device, state, outputs) != NO_ERROR) { |
| mAvailableOutputDevices.remove(device); |
| |
| mHwModules.cleanUpForDevice(device); |
| |
| broadcastDeviceConnectionState(device, AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE); |
| return INVALID_OPERATION; |
| } |
| // Propagate device availability to Engine |
| mEngine->setDeviceConnectionState(device, state); |
| //TODO CP Begin |
| #if 0 |
| if (deviceType == AUDIO_DEVICE_OUT_AUX_DIGITAL) { |
| chkDpConnAndAllowedForVoice(); |
| } |
| #endif |
| //TODO CP End |
| |
| // outputs should never be empty here |
| ALOG_ASSERT(outputs.size() != 0, "setDeviceConnectionState():" |
| "checkOutputsForDevice() returned no outputs but status OK"); |
| ALOGV("%s() checkOutputsForDevice() returned %zu outputs", __func__, outputs.size()); |
| |
| } break; |
| // handle output device disconnection |
| case AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE: { |
| if (index < 0) { |
| if (mApmConfigs->isHDMISpkEnabled() && |
| (popcount(deviceType) == 1) && (deviceType & AUDIO_DEVICE_OUT_AUX_DIGITAL)) { |
| if (!strncmp(device_address, "hdmi_spkr", 9)) { |
| mHdmiAudioDisabled = true; |
| } else { |
| mHdmiAudioEvent = false; |
| } |
| } |
| ALOGW("setDeviceConnectionState() device not connected: %x", deviceType); |
| return INVALID_OPERATION; |
| } |
| |
| ALOGV("%s() disconnecting output device %s", __func__, device->toString().c_str()); |
| |
| // Send Disconnect to HALs |
| broadcastDeviceConnectionState(device, state); |
| |
| // remove device from available output devices |
| mAvailableOutputDevices.remove(device); |
| |
| mOutputs.clearSessionRoutesForDevice(device); |
| if (mApmConfigs->isHDMISpkEnabled() && |
| (popcount(deviceType) == 1) && (deviceType & AUDIO_DEVICE_OUT_AUX_DIGITAL)) { |
| if (!strncmp(device_address, "hdmi_spkr", 9)) { |
| mHdmiAudioDisabled = true; |
| } else { |
| mHdmiAudioEvent = false; |
| } |
| } |
| checkOutputsForDevice(device, state, outputs); |
| |
| // Reset active device codec |
| device->setEncodedFormat(AUDIO_FORMAT_DEFAULT); |
| |
| // Propagate device availability to Engine |
| mEngine->setDeviceConnectionState(device, state); |
| //TODO CP Begin |
| #if 0 |
| if (deviceType == AUDIO_DEVICE_OUT_AUX_DIGITAL) { |
| mEngine->setDpConnAndAllowedForVoice(false); |
| } |
| #endif |
| //TODO CP End |
| } break; |
| |
| default: |
| ALOGE("%s() invalid state: %x", __func__, state); |
| return BAD_VALUE; |
| } |
| |
| if (!outputs.isEmpty()) { |
| for (size_t i = 0; i < outputs.size(); i++) { |
| sp<SwAudioOutputDescriptor> desc = mOutputs.valueFor(outputs[i]); |
| // close voip output before track invalidation to allow creation of |
| // new voip stream from restoreTrack |
| if ((desc->mFlags == (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_VOIP_RX)) != 0) { |
| closeOutput(outputs[i]); |
| outputs.remove(outputs[i]); |
| } |
| } |
| } |
| |
| // No need to evaluate playback routing when connecting a remote submix |
| // output device used by a dynamic policy of type recorder as no |
| // playback use case is affected. |
| bool doCheckForDeviceAndOutputChanges = true; |
| if (device->type() == AUDIO_DEVICE_OUT_REMOTE_SUBMIX |
| && strncmp(device_address, "0", AUDIO_DEVICE_MAX_ADDRESS_LEN) != 0) { |
| for (audio_io_handle_t output : outputs) { |
| sp<SwAudioOutputDescriptor> desc = mOutputs.valueFor(output); |
| if (desc->mPolicyMix != nullptr |
| && desc->mPolicyMix->mMixType == MIX_TYPE_RECORDERS |
| && strncmp(device_address, |
| desc->mPolicyMix->mDeviceAddress.string(), |
| AUDIO_DEVICE_MAX_ADDRESS_LEN) == 0) { |
| doCheckForDeviceAndOutputChanges = false; |
| break; |
| } |
| } |
| } |
| |
| auto checkCloseOutputs = [&]() { |
| // outputs must be closed after checkOutputForAllStrategies() is executed |
| if (!outputs.isEmpty()) { |
| for (audio_io_handle_t output : outputs) { |
| sp<SwAudioOutputDescriptor> desc = mOutputs.valueFor(output); |
| // 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(output); |
| } |
| } |
| // check A2DP again after closing A2DP output to reset mA2dpSuspended if needed |
| return true; |
| } |
| return false; |
| }; |
| |
| if (doCheckForDeviceAndOutputChanges) { |
| checkForDeviceAndOutputChanges(checkCloseOutputs); |
| } else { |
| checkCloseOutputs(); |
| } |
| |
| // handle FM device connection state to trigger FM AFE loopback |
| //TODO CP Begin |
| #if 0 |
| if (mApmConfigs->isFMPowerOptEnabled() && |
| deviceType == AUDIO_DEVICE_OUT_FM && hasPrimaryOutput()) { |
| audio_devices_t newDevice = AUDIO_DEVICE_NONE; |
| if (state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) { |
| /* |
| when mPrimaryOutput->start() is called for FM it would check if isActive() is true |
| or not as streamActiveCount=0 so isActive() would return false and curActiveCount will be |
| 1 and then the streamActiveCount will be increased by 1 for FM case.Updating curActiveCount |
| is important as in case of adding other tracks when FM is still active isActive() |
| will always be true as streamActiveCount will always be > 0,Hence curActiveCount will never |
| update for them. However ,when fm stops and the track stops too streamActiveCount will be 0 |
| isActive will false,it will check if curActiveCount < 1 as curActiveCount was never |
| updated so LOG_FATAL will cause the AudioServer to die.Hence this start() call will |
| ensure that curActiveCount is updated at least once when FM starts prior to other |
| tracks and on calling of stop() LOG_FATAL is not called. |
| */ |
| mPrimaryOutput->start(); |
| for (const std::pair<sp<TrackClientDescriptor>, size_t>& client_pair : mPrimaryOutput->getActiveClients()) { |
| if (client_pair.first->stream() == AUDIO_STREAM_MUSIC) { |
| mPrimaryOutput->changeStreamActiveCount(client_pair.first, 1); |
| break; |
| } |
| } |
| newDevice = (audio_devices_t)(getNewOutputDevice(mPrimaryOutput, false)|AUDIO_DEVICE_OUT_FM); |
| mFMIsActive = true; |
| mPrimaryOutput->mDevice = newDevice & ~AUDIO_DEVICE_OUT_FM; |
| } else { |
| newDevice = (audio_devices_t)(getNewOutputDevice(mPrimaryOutput, false)); |
| mFMIsActive = false; |
| for (const std::pair<sp<TrackClientDescriptor>, size_t>& client_pair : mPrimaryOutput->getActiveClients()) { |
| if (client_pair.first->stream() == AUDIO_STREAM_MUSIC) { |
| mPrimaryOutput->changeStreamActiveCount(client_pair.first, -1); |
| break; |
| } |
| } |
| /* |
| mPrimaryOutput->stop() is called as because of calling of start() |
| in FM case curActiveCount is getting updated and hence stop() is |
| called so that curActiveCount gets decremented and if any tracks |
| are added after FM stops they may get curActiveCount=0 ,ouptput |
| curActiveCount can be properly updated |
| */ |
| mPrimaryOutput->stop(); |
| } |
| AudioParameter param = AudioParameter(); |
| param.addInt(String8("handle_fm"), (int)newDevice); |
| mpClientInterface->setParameters(mPrimaryOutput->mIoHandle, param.toString()); |
| } |
| #endif |
| //TODO CP End |
| |
| if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL && hasPrimaryOutput()) { |
| DeviceVector newDevices = getNewOutputDevices(mPrimaryOutput, false /*fromCache*/); |
| updateCallRouting(newDevices); |
| } |
| |
| for (size_t i = 0; i < mOutputs.size(); i++) { |
| sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i); |
| if ((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) || (desc != mPrimaryOutput)) { |
| DeviceVector newDevices = getNewOutputDevices(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(deviceType) |
| // always force when disconnecting (a non-duplicated device) |
| || (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE)); |
| setOutputDevices(desc, newDevices, force, 0); |
| } |
| } |
| |
| if (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) { |
| cleanUpForDevice(device); |
| } |
| |
| mpClientInterface->onAudioPortListUpdate(); |
| return NO_ERROR; |
| } // end if is output device |
| |
| // handle input devices |
| if (audio_is_input_device(deviceType)) { |
| SortedVector <audio_io_handle_t> inputs; |
| |
| ssize_t index = mAvailableInputDevices.indexOf(device); |
| switch (state) |
| { |
| // handle input device connection |
| case AUDIO_POLICY_DEVICE_STATE_AVAILABLE: { |
| if (index >= 0) { |
| ALOGW("setDeviceConnectionState() device already connected: %d", deviceType); |
| return INVALID_OPERATION; |
| } |
| sp<HwModule> module = mHwModules.getModuleForDevice(device, AUDIO_FORMAT_DEFAULT); |
| if (module == NULL) { |
| ALOGW("setDeviceConnectionState(): could not find HW module for device %08x", |
| deviceType); |
| return INVALID_OPERATION; |
| } |
| |
| // Before checking intputs, broadcast connect event to allow HAL to retrieve dynamic |
| // parameters on newly connected devices (instead of opening the inputs...) |
| broadcastDeviceConnectionState(device, state); |
| |
| if (checkInputsForDevice(device, state, inputs) != NO_ERROR) { |
| broadcastDeviceConnectionState(device, AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE); |
| |
| mHwModules.cleanUpForDevice(device); |
| |
| return INVALID_OPERATION; |
| } |
| |
| index = mAvailableInputDevices.add(device); |
| if (index >= 0) { |
| mAvailableInputDevices[index]->attach(module); |
| } else { |
| return NO_MEMORY; |
| } |
| |
| // Propagate device availability to Engine |
| mEngine->setDeviceConnectionState(device, state); |
| } break; |
| |
| // handle input device disconnection |
| case AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE: { |
| if (index < 0) { |
| ALOGW("setDeviceConnectionState() device not connected: %d", deviceType); |
| return INVALID_OPERATION; |
| } |
| |
| ALOGV("setDeviceConnectionState() disconnecting input device %x", deviceType); |
| |
| // Set Disconnect to HALs |
| broadcastDeviceConnectionState(device, state); |
| |
| checkInputsForDevice(device, state, inputs); |
| mAvailableInputDevices.remove(device); |
| |
| // Propagate device availability to Engine |
| mEngine->setDeviceConnectionState(device, state); |
| } break; |
| |
| default: |
| ALOGE("setDeviceConnectionState() invalid state: %x", state); |
| return BAD_VALUE; |
| } |
| |
| closeAllInputs(); |
| /*audio policy: fix call volume over USB*/ |
| // As the input device list can impact the output device selection, update |
| // getDeviceForStrategy() cache |
| updateDevicesAndOutputs(); |
| |
| if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL && hasPrimaryOutput()) { |
| DeviceVector newDevices = getNewOutputDevices(mPrimaryOutput, false /*fromCache*/); |
| updateCallRouting(newDevices); |
| } |
| |
| if (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) { |
| cleanUpForDevice(device); |
| } |
| |
| mpClientInterface->onAudioPortListUpdate(); |
| return NO_ERROR; |
| } // end if is input device |
| |
| ALOGW("setDeviceConnectionState() invalid device: %x", deviceType); |
| return BAD_VALUE; |
| } |
| |
| |
| //TODO CP Begin |
| #if 0 |
| |
| void AudioPolicyManagerCustom::chkDpConnAndAllowedForVoice() |
| { |
| String8 value; |
| bool connAndAllowed = false; |
| String8 valueStr = mpClientInterface->getParameters((audio_io_handle_t)0, |
| String8("dp_for_voice")); |
| |
| AudioParameter result = AudioParameter(valueStr); |
| if (result.get(String8("dp_for_voice"), value) == NO_ERROR) { |
| connAndAllowed = value.contains("true"); |
| } |
| mEngine->setDpConnAndAllowedForVoice(connAndAllowed); |
| } |
| #endif |
| //TODO CP End |
| |
| bool AudioPolicyManagerCustom::isInvalidationOfMusicStreamNeeded(const audio_attributes_t &attr) |
| { |
| if (followsSameRouting(attr, attributes_initializer(AUDIO_USAGE_MEDIA))) { |
| for (size_t i = 0; i < mOutputs.size(); i++) { |
| sp<SwAudioOutputDescriptor> newOutputDesc = mOutputs.valueAt(i); |
| if (newOutputDesc->mFormat == AUDIO_FORMAT_DSD) |
| return false; |
| } |
| } |
| return true; |
| } |
| |
| |
| void AudioPolicyManagerCustom::checkOutputForAttributes(const audio_attributes_t &attr) |
| { |
| |
| DeviceVector oldDevices = mEngine->getOutputDevicesForAttributes(attr, 0, true /*fromCache*/); |
| DeviceVector newDevices = mEngine->getOutputDevicesForAttributes(attr, 0, false /*fromCache*/); |
| SortedVector<audio_io_handle_t> srcOutputs = getOutputsForDevices(oldDevices, mPreviousOutputs); |
| SortedVector<audio_io_handle_t> dstOutputs = getOutputsForDevices(newDevices, mOutputs); |
| |
| // also take into account external policy-related changes: add all outputs which are |
| // associated with policies in the "before" and "after" output vectors |
| ALOGVV("%s(): policy related outputs", __func__); |
| for (size_t i = 0 ; i < mPreviousOutputs.size() ; i++) { |
| const sp<SwAudioOutputDescriptor> desc = mPreviousOutputs.valueAt(i); |
| if (desc != 0 && desc->mPolicyMix != NULL) { |
| srcOutputs.add(desc->mIoHandle); |
| ALOGVV(" previous outputs: adding %d", desc->mIoHandle); |
| } |
| } |
| for (size_t i = 0 ; i < mOutputs.size() ; i++) { |
| const sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i); |
| if (desc != 0 && desc->mPolicyMix != NULL) { |
| dstOutputs.add(desc->mIoHandle); |
| ALOGVV(" new outputs: adding %d", desc->mIoHandle); |
| } |
| } |
| |
| if ((srcOutputs != dstOutputs) && isInvalidationOfMusicStreamNeeded(attr)) { |
| AudioPolicyManager::checkOutputForAttributes(attr); |
| } |
| } |
| |
| // 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) |
| { |
| DeviceVector primaryInputDevices = availablePrimaryModuleInputDevices(); |
| 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); |
| |
| if (mMasterMono) { |
| return false; // no offloading if mono is set. |
| } |
| |
| if (mApmConfigs->isVoiceConcEnabled()) { |
| if (mApmConfigs->isVoicePlayConcDisabled() && isInCall()) { |
| ALOGD("\n copl: blocking compress offload on call mode\n"); |
| return false; |
| } |
| } |
| if (mApmConfigs->isVoiceDSDConcDisabled() && |
| isInCall() && (offloadInfo.format == AUDIO_FORMAT_DSD)) { |
| ALOGD("blocking DSD compress offload on call mode"); |
| return false; |
| } |
| if (mApmConfigs->isRecPlayConcEnabled()) { |
| if (mApmConfigs->isRecPlayConcDisabled() && |
| ((true == mIsInputRequestOnProgress) || |
| (mInputs.activeInputsCountOnDevices(primaryInputDevices) > 0))) { |
| ALOGD("copl: blocking compress offload for record concurrency"); |
| 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 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; |
| } |
| |
| //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; |
| } |
| |
| if (mApmConfigs->isExtnFormatsEnabled()) { |
| //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; |
| } |
| |
| // check against wma std bit rate restriction |
| if ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA) { |
| int32_t sr_id = -1; |
| uint32_t min_bitrate, max_bitrate; |
| for (int i = 0; i < WMA_STD_NUM_FREQ; i++) { |
| if (offloadInfo.sample_rate == wmaStdSampleRateTbl[i]) { |
| sr_id = i; |
| break; |
| } |
| } |
| if ((sr_id < 0) || (popcount(offloadInfo.channel_mask) > 2) |
| || (popcount(offloadInfo.channel_mask) <= 0)) { |
| ALOGE("invalid sample rate or channel count"); |
| return false; |
| } |
| |
| min_bitrate = wmaStdMinAvgByteRateTbl[sr_id][popcount(offloadInfo.channel_mask) - 1]; |
| max_bitrate = wmaStdMaxAvgByteRateTbl[sr_id][popcount(offloadInfo.channel_mask) - 1]; |
| if ((offloadInfo.bit_rate > max_bitrate) || (offloadInfo.bit_rate < min_bitrate)) { |
| ALOGD("offload disabled for WMA clips with unsupported bit rate"); |
| ALOGD("bit_rate %d, max_bitrate %d, min_bitrate %d", offloadInfo.bit_rate, max_bitrate, min_bitrate); |
| return false; |
| } |
| } |
| |
| // Safely choose the min bitrate as threshold and leave the restriction to NT decoder as we can't distinguish wma pro and wma lossless here. |
| if ((((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA_PRO) && (offloadInfo.bit_rate > MAX_BITRATE_WMA_PRO)) || |
| (((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA_PRO) && (offloadInfo.bit_rate > MAX_BITRATE_WMA_LOSSLESS))) { |
| ALOGD("offload disabled for WMA_PRO/WMA_LOSSLESS clips with bit rate over maximum supported value"); |
| return false; |
| } |
| } |
| |
| //TODO: enable audio offloading with video when ready |
| if (offloadInfo.has_video && !mApmConfigs->isAudioOffloadVideoEnabled()) { |
| ALOGV("isOffloadSupported: has_video == true, returning false"); |
| return false; |
| } |
| |
| if (offloadInfo.has_video && offloadInfo.is_streaming && |
| !mApmConfigs->isAVStreamingOffloadEnabled()) { |
| ALOGW("offload disabled by vendor.audio.av.streaming.offload.enable %d", |
| mApmConfigs->isAVStreamingOffloadEnabled()); |
| return false; |
| } |
| |
| //If duration is less than minimum value defined in property, return false |
| if (mApmConfigs->getAudioOffloadMinDuration() > 0) { |
| if (offloadInfo.duration_us < (mApmConfigs->getAudioOffloadMinDuration() * 1000000 )) { |
| ALOGV("Offload denied by duration < audio.offload.min.duration.secs(=%u)", mApmConfigs->getAudioOffloadMinDuration()); |
| 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. 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_FLAC) || |
| ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_VORBIS)) |
| return false; |
| |
| if (mApmConfigs->isExtnFormatsEnabled()) { |
| if (((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_DSD) || |
| ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC_ADTS)) |
| return false; |
| } |
| } |
| |
| // Do not allow offloading if one non offloadable effect is enabled. This prevents from |
| // creating an offloaded track and tearing it down immediately after start when audioflinger |
| // detects there is an active non offloadable effect. |
| // FIXME: We should check the audio session here but we do not have it in this context. |
| // This may prevent offloading in rare situations where effects are left active by apps |
| // in the background. |
| if (mEffects.isNonOffloadableEffectEnabled()) { |
| return false; |
| } |
| |
| // See if there is a profile to support this. |
| // AUDIO_DEVICE_NONE |
| sp<IOProfile> profile = getProfileForOutput(DeviceVector() /*ignore device */, |
| offloadInfo.sample_rate, |
| offloadInfo.format, |
| offloadInfo.channel_mask, |
| AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD, |
| true /*directOnly*/); |
| ALOGV("isOffloadSupported() profile %sfound", profile != 0 ? "" : "NOT "); |
| return (profile != 0); |
| } |
| |
| void AudioPolicyManagerCustom::setPhoneState(audio_mode_t state) |
| { |
| ALOGD("setPhoneState() state %d", state); |
| // store previous phone state for management of sonification strategy below |
| int oldState = mEngine->getPhoneState(); |
| |
| if (mEngine->setPhoneState(state) != NO_ERROR) { |
| ALOGW("setPhoneState() invalid or same state %d", state); |
| return; |
| } |
| /// Opens: can these line be executed after the switch of volume curves??? |
| if (isStateInCall(oldState)) { |
| ALOGV("setPhoneState() in call state management: new state is %d", state); |
| |
| // 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 |
| checkForDeviceAndOutputChanges(); |
| |
| sp<SwAudioOutputDescriptor> hwOutputDesc = mPrimaryOutput; |
| if (mApmConfigs->isVoiceConcEnabled()) { |
| bool prop_playback_enabled = mApmConfigs->isVoicePlayConcDisabled(); |
| bool prop_rec_enabled = mApmConfigs->isVoiceRecConcDisabled(); |
| bool prop_voip_enabled = mApmConfigs->isVoiceVOIPConcDisabled(); |
| |
| 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 |
| Vector<sp <AudioInputDescriptor> > activeInputs = mInputs.getActiveInputs(); |
| if (activeInputs.size() != 0) { |
| for (size_t i = 0; i < activeInputs.size(); i++) { |
| sp<AudioInputDescriptor> activeInput = activeInputs[i]; |
| switch(activeInput->source()) { |
| 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", |
| activeInput->source()); |
| break; |
| |
| case AUDIO_SOURCE_VOICE_COMMUNICATION: |
| if (prop_voip_enabled) { |
| ALOGD("voice_conc:CLOSING VoIP input source on call setup :%d ", |
| activeInput->source()); |
| RecordClientVector activeClients = activeInput->clientsList(true /*activeOnly*/); |
| for (const auto& activeClient : activeClients) { |
| closeClient(activeClient->portId()); |
| } |
| } |
| break; |
| |
| default: |
| ALOGD("voice_conc:CLOSING input on call setup for inputSource: %d", |
| activeInput->source()); |
| RecordClientVector activeClients = activeInput->clientsList(true /*activeOnly*/); |
| for (const auto& activeClient : activeClients) { |
| closeClient(activeClient->portId()); |
| } |
| break; |
| } |
| } |
| } |
| } else if (prop_voip_enabled) { |
| Vector<sp <AudioInputDescriptor> > activeInputs = mInputs.getActiveInputs(); |
| if (activeInputs.size() != 0) { |
| for (size_t i = 0; i < activeInputs.size(); i++) { |
| sp<AudioInputDescriptor> activeInput = activeInputs[i]; |
| if (AUDIO_SOURCE_VOICE_COMMUNICATION == activeInput->source()) { |
| ALOGD("voice_conc:CLOSING VoIP on call setup : %d",activeInput->source()); |
| RecordClientVector activeClients = activeInput->clientsList(true /*activeOnly*/); |
| for (const auto& activeClient : activeClients) { |
| closeClient(activeClient->portId()); |
| } |
| } |
| } |
| } |
| } |
| if (prop_playback_enabled) { |
| // Move tracks associated to this strategy from previous output to new output |
| for (int i = AUDIO_STREAM_SYSTEM; i < AUDIO_STREAM_FOR_POLICY_CNT; i++) { |
| ALOGV("voice_conc:Invalidate on call mode for stream :: %d ", i); |
| 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; |
| } |
| |
| bool isFastFallBackNeeded = |
| ((AUDIO_OUTPUT_FLAG_DEEP_BUFFER | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT_PCM) & outputDesc->mProfile->getFlags()); |
| |
| if ((AUDIO_OUTPUT_FLAG_FAST == mFallBackflag) && isFastFallBackNeeded) { |
| if (((!outputDesc->isDuplicated() && outputDesc->mProfile->getFlags() & 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->getFlags() & 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->getFlags() & 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->getFlags() & AUDIO_OUTPUT_FLAG_VOIP_RX) { |
| if (prop_voip_enabled) { |
| ALOGD("voice_conc:calling closeOutput on call mode for DIRECT output"); |
| closeOutput(mOutputs.keyAt(i)); |
| } |
| } |
| else if (prop_playback_enabled |
| && (outputDesc->mProfile->getFlags() & AUDIO_OUTPUT_FLAG_DIRECT)) { |
| 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->getFlags() & 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 < AUDIO_STREAM_FOR_POLICY_CNT; i++) { |
| ALOGV("voice_conc:Invalidate on call mode for stream :: %d ", i); |
| 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); |
| } |
| } |
| } |
| } |
| |
| sp<SwAudioOutputDescriptor> outputDesc = NULL; |
| for (size_t i = 0; i < mOutputs.size(); i++) { |
| outputDesc = mOutputs.valueAt(i); |
| if ((outputDesc == NULL) || (outputDesc->mProfile == NULL)) { |
| ALOGD("voice_conc:ouput desc / profile is NULL"); |
| continue; |
| } |
| |
| if (mApmConfigs->isVoiceDSDConcDisabled() && |
| (outputDesc->mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) && |
| (outputDesc->mFormat == AUDIO_FORMAT_DSD)) { |
| ALOGD("voice_conc:calling closeOutput on call mode for DSD COMPRESS output"); |
| closeOutput(mOutputs.keyAt(i)); |
| // call invalidate for music, so that DSD compress will fallback to deep-buffer. |
| mpClientInterface->invalidateStream(AUDIO_STREAM_MUSIC); |
| } |
| |
| } |
| |
| if (mApmConfigs->isRecPlayConcEnabled()) { |
| if (mApmConfigs->isRecPlayConcDisabled()) { |
| 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->getFlags() & 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); |
| } |
| } |
| } |
| mPrevPhoneState = oldState; |
| int delayMs = 0; |
| if (isStateInCall(state)) { |
| nsecs_t sysTime = systemTime(); |
| auto musicStrategy = streamToStrategy(AUDIO_STREAM_MUSIC); |
| auto sonificationStrategy = streamToStrategy(AUDIO_STREAM_ALARM); |
| |
| 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 ((desc->isStrategyActive(musicStrategy, |
| SONIFICATION_HEADSET_MUSIC_DELAY, |
| sysTime) || |
| desc->isStrategyActive(sonificationStrategy, |
| SONIFICATION_HEADSET_MUSIC_DELAY, |
| sysTime)) && |
| (delayMs < (int)desc->latency()*2)) { |
| delayMs = desc->latency()*2; |
| } |
| setStrategyMute(musicStrategy, true, desc); |
| setStrategyMute(musicStrategy, false, desc, MUTE_TIME_MS, |
| mEngine->getOutputDevicesForAttributes(attributes_initializer(AUDIO_USAGE_MEDIA), |
| nullptr, true /*fromCache*/).types()); |
| setStrategyMute(sonificationStrategy, true, desc); |
| setStrategyMute(sonificationStrategy, false, desc, MUTE_TIME_MS, |
| mEngine->getOutputDevicesForAttributes(attributes_initializer(AUDIO_USAGE_ALARM), |
| nullptr, true /*fromCache*/).types()); |
| } |
| } |
| |
| 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 |
| DeviceVector rxDevices = getNewOutputDevices(mPrimaryOutput, false /*fromCache*/); |
| // force routing command to audio hardware when ending call |
| // even if no device change is needed |
| if (isStateInCall(oldState) && rxDevices.isEmpty()) { |
| rxDevices = mPrimaryOutput->devices(); |
| } |
| |
| if (state == AUDIO_MODE_IN_CALL) { |
| updateCallRouting(rxDevices, 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(); |
| } |
| setOutputDevices(mPrimaryOutput, rxDevices, force, 0); |
| } else { |
| setOutputDevices(mPrimaryOutput, rxDevices, 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) { |
| DeviceVector newDevices = getNewOutputDevices(mOutputs.valueFor(output), false /*fromCache*/); |
| setOutputDevices(mOutputs.valueFor(output), newDevices, !newDevices.isEmpty()); |
| } |
| } |
| if (isStateInCall(state)) { |
| ALOGV("setPhoneState() in call state management: new state is %d", state); |
| |
| // 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 |
| mLimitRingtoneVolume = (state == AUDIO_MODE_RINGTONE && |
| isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY)); |
| } |
| |
| void AudioPolicyManagerCustom::setForceUse(audio_policy_force_use_t usage, |
| audio_policy_forced_cfg_t config) |
| { |
| ALOGD("setForceUse() usage %d, config %d, mPhoneState %d", usage, config, mEngine->getPhoneState()); |
| if (config == mEngine->getForceUse(usage)) { |
| return; |
| } |
| |
| 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 |
| checkForDeviceAndOutputChanges(); |
| |
| /*audio policy: workaround for truncated touch sounds*/ |
| //FIXME: workaround for truncated touch sounds |
| // to be removed when the problem is handled by system UI |
| uint32_t delayMs = 0; |
| uint32_t waitMs = 0; |
| if (usage == AUDIO_POLICY_FORCE_FOR_COMMUNICATION) { |
| delayMs = TOUCH_SOUND_FIXED_DELAY_MS; |
| } |
| if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL && hasPrimaryOutput()) { |
| DeviceVector newDevices = getNewOutputDevices(mPrimaryOutput, true /*fromCache*/); |
| if (forceVolumeReeval && !newDevices.isEmpty()) { |
| applyStreamVolumes(mPrimaryOutput, newDevices.types(), delayMs, true); |
| } |
| waitMs = updateCallRouting(newDevices, delayMs); |
| } |
| // 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); |
| DeviceVector newDevices = getNewOutputDevices(outputDesc, true /*fromCache*/); |
| if ((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) || (outputDesc != mPrimaryOutput)) { |
| waitMs = setOutputDevices(outputDesc, newDevices, !newDevices.isEmpty(), |
| delayMs); |
| |
| if (forceVolumeReeval && !newDevices.isEmpty()) { |
| applyStreamVolumes(outputDesc, newDevices.types(), waitMs, true); |
| } |
| } |
| } |
| |
| for (const auto& activeDesc : mInputs.getActiveInputs()) { |
| // Skip for hotword recording as the input device switch |
| // is handled within sound trigger HAL |
| if (activeDesc->isSoundTrigger() && |
| activeDesc->source() == AUDIO_SOURCE_HOTWORD) { |
| continue; |
| } |
| auto newDevice = getNewInputDevice(activeDesc); |
| // Force new input selection if the new device can not be reached via current input |
| if (activeDesc->mProfile->getSupportedDevices().contains(newDevice)) { |
| setInputDevice(activeDesc->mIoHandle, newDevice); |
| } else { |
| closeInput(activeDesc->mIoHandle); |
| } |
| } |
| } |
| |
| status_t AudioPolicyManagerCustom::stopSource(const sp<SwAudioOutputDescriptor>& outputDesc, |
| const sp<TrackClientDescriptor>& client) |
| { |
| audio_stream_type_t stream = client->stream(); |
| auto clientVolSrc = client->volumeSource(); |
| |
| 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); |
| |
| if (outputDesc->getActivityCount(clientVolSrc) > 0) { |
| if (outputDesc->getActivityCount(clientVolSrc) == 1) { |
| // Automatically disable the remote submix input when output is stopped on a |
| // re routing mix of type MIX_TYPE_RECORDERS |
| if (audio_is_remote_submix_device(outputDesc->devices().types()) && |
| outputDesc->mPolicyMix != NULL && |
| outputDesc->mPolicyMix->mMixType == MIX_TYPE_RECORDERS) { |
| setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX, |
| AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE, |
| outputDesc->mPolicyMix->mDeviceAddress, |
| "remote-submix", AUDIO_FORMAT_DEFAULT); |
| } |
| } |
| bool forceDeviceUpdate = false; |
| if (client->hasPreferredDevice(true)) { |
| checkStrategyRoute(client->strategy(), AUDIO_IO_HANDLE_NONE); |
| forceDeviceUpdate = true; |
| } |
| |
| // decrement usage count of this stream on the output |
| outputDesc->setClientActive(client, false); |
| |
| // store time at which the stream was stopped - see isStreamActive() |
| if (outputDesc->getActivityCount(clientVolSrc) == 0 || forceDeviceUpdate) { |
| outputDesc->setStopTime(client, systemTime()); |
| DeviceVector prevDevices = outputDesc->devices(); |
| DeviceVector newDevices = getNewOutputDevices(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 ...) |
| setOutputDevices(outputDesc, newDevices, 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<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i); |
| if (desc != outputDesc && |
| desc->isActive() && |
| outputDesc->sharesHwModuleWith(desc) && |
| (newDevices != desc->devices())) { |
| DeviceVector dev = getNewOutputDevices(mOutputs.valueFor(curOutput), false /*fromCache*/); |
| bool force = prevDevices != dev; |
| uint32_t delayMs; |
| if (dev == prevDevices) { |
| delayMs = 0; |
| } else { |
| delayMs = outputDesc->latency()*2; |
| } |
| setOutputDevices(desc, |
| dev, |
| force, |
| delayMs); |
| /*audio policy: fix media volume after ringtone*/ |
| // re-apply device specific volume if not done by setOutputDevice() |
| if (!force) { |
| applyStreamVolumes(desc, dev.types(), delayMs); |
| } |
| } |
| } |
| // update the outputs if stopping one with a stream that can affect notification routing |
| handleNotificationRoutingForStream(stream); |
| } |
| |
| if (stream == AUDIO_STREAM_ENFORCED_AUDIBLE && |
| mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) { |
| setStrategyMute(streamToStrategy(AUDIO_STREAM_RING), false, outputDesc); |
| } |
| |
| if (followsSameRouting(client->attributes(), attributes_initializer(AUDIO_USAGE_MEDIA))) { |
| selectOutputForMusicEffects(); |
| } |
| return NO_ERROR; |
| } else { |
| ALOGW("stopOutput() refcount is already 0"); |
| return INVALID_OPERATION; |
| } |
| } |
| |
| status_t AudioPolicyManagerCustom::startSource(const sp<SwAudioOutputDescriptor>& outputDesc, |
| const sp<TrackClientDescriptor>& client, |
| uint32_t *delayMs) |
| { |
| // cannot start playback of STREAM_TTS if any other output is being used |
| uint32_t beaconMuteLatency = 0; |
| audio_stream_type_t stream = client->stream(); |
| auto clientVolSrc = client->volumeSource(); |
| auto clientStrategy = client->strategy(); |
| auto clientAttr = client->attributes(); |
| |
| 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 (!mTtsOutputAvailable && mOutputs.isAnyOutputActive( |
| streamToVolumeSource(AUDIO_STREAM_TTS) /*sourceToIgnore*/)) { |
| return INVALID_OPERATION; |
| } else { |
| beaconMuteLatency = handleEventForBeacon(STARTING_BEACON); |
| } |
| } else { |
| // some playback other than beacon starts |
| beaconMuteLatency = handleEventForBeacon(STARTING_OUTPUT); |
| } |
| |
| // force device change if the output is inactive and no audio patch is already present. |
| // check active before incrementing usage count |
| bool force = !outputDesc->isActive() && |
| (outputDesc->getPatchHandle() == AUDIO_PATCH_HANDLE_NONE); |
| |
| DeviceVector devices; |
| AudioMix *policyMix = NULL; |
| const char *address = NULL; |
| if (outputDesc->mPolicyMix != NULL) { |
| policyMix = outputDesc->mPolicyMix; |
| audio_devices_t newDeviceType; |
| address = policyMix->mDeviceAddress.string(); |
| if ((policyMix->mRouteFlags & MIX_ROUTE_FLAG_LOOP_BACK) == MIX_ROUTE_FLAG_LOOP_BACK) { |
| newDeviceType = AUDIO_DEVICE_OUT_REMOTE_SUBMIX; |
| } else { |
| newDeviceType = policyMix->mDeviceType; |
| } |
| sp device = mAvailableOutputDevices.getDevice(newDeviceType, String8(address), |
| AUDIO_FORMAT_DEFAULT); |
| ALOG_ASSERT(device, "%s: no device found t=%u, a=%s", __func__, newDeviceType, address); |
| devices.add(device); |
| } |
| |
| // requiresMuteCheck is false when we can bypass mute strategy. |
| // It covers a common case when there is no materially active audio |
| // and muting would result in unnecessary delay and dropped audio. |
| const uint32_t outputLatencyMs = outputDesc->latency(); |
| bool requiresMuteCheck = outputDesc->isActive(outputLatencyMs * 2); // account for drain |
| |
| // 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->setClientActive(client, true); |
| |
| if (client->hasPreferredDevice(true)) { |
| devices = getNewOutputDevices(outputDesc, false /*fromCache*/); |
| if (devices != outputDesc->devices()) { |
| checkStrategyRoute(clientStrategy, outputDesc->mIoHandle); |
| } |
| } |
| |
| if (followsSameRouting(clientAttr, attributes_initializer(AUDIO_USAGE_MEDIA))) { |
| selectOutputForMusicEffects(); |
| } |
| |
| if (outputDesc->getActivityCount(clientVolSrc) == 1 || !devices.isEmpty()) { |
| // starting an output being rerouted? |
| if (devices.isEmpty()) { |
| devices = getNewOutputDevices(outputDesc, false /*fromCache*/); |
| } |
| bool shouldWait = |
| (followsSameRouting(clientAttr, attributes_initializer(AUDIO_USAGE_ALARM)) || |
| followsSameRouting(clientAttr, attributes_initializer(AUDIO_USAGE_NOTIFICATION)) || |
| (beaconMuteLatency > 0)); |
| uint32_t waitMs = beaconMuteLatency; |
| for (size_t i = 0; i < mOutputs.size(); i++) { |
| sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i); |
| if (desc != outputDesc) { |
| // An output has a shared device if |
| // - managed by the same hw module |
| // - supports the currently selected device |
| const bool sharedDevice = outputDesc->sharesHwModuleWith(desc) |
| && (!desc->filterSupportedDevices(devices).isEmpty()); |
| |
| // force a device change if any other output is: |
| // - managed by the same hw module |
| // - supports currently selected device |
| // - has a current device selection that differs from selected device. |
| // - has an active audio patch |
| // In this case, the audio HAL must receive the new device selection so that it can |
| // change the device currently selected by the other output. |
| if (sharedDevice && |
| desc->devices() != devices && |
| desc->getPatchHandle() != AUDIO_PATCH_HANDLE_NONE) { |
| 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 |
| const uint32_t latencyMs = desc->latency(); |
| const bool isActive = desc->isActive(latencyMs * 2); // account for drain |
| |
| if (shouldWait && isActive && (waitMs < latencyMs)) { |
| waitMs = latencyMs; |
| } |
| |
| // Require mute check if another output is on a shared device |
| // and currently active to have proper drain and avoid pops. |
| // Note restoring AudioTracks onto this output needs to invoke |
| // a volume ramp if there is no mute. |
| requiresMuteCheck |= sharedDevice && isActive; |
| } |
| } |
| const uint32_t muteWaitMs = |
| setOutputDevices(outputDesc, devices, force, 0, NULL, requiresMuteCheck); |
| |
| // apply volume rules for current stream and device if necessary |
| checkAndSetVolume(stream, |
| getVolumeCurves(stream).getVolumeIndex(outputDesc->devices().types()), |
| outputDesc, |
| outputDesc->devices().types()); |
| |
| // 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 (followsSameRouting(clientAttr, attributes_initializer(AUDIO_USAGE_ALARM))) { |
| mpClientInterface->invalidateStream(AUDIO_STREAM_ACCESSIBILITY); |
| } |
| if (waitMs > muteWaitMs) { |
| *delayMs = waitMs - muteWaitMs; |
| } |
| |
| } |
| |
| if (stream == AUDIO_STREAM_ENFORCED_AUDIBLE && |
| mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) { |
| setStrategyMute(streamToStrategy(AUDIO_STREAM_ALARM), true, outputDesc); |
| } |
| |
| // Automatically enable the remote submix input when output is started on a re routing mix |
| // of type MIX_TYPE_RECORDERS |
| if (audio_is_remote_submix_device(devices.types()) && policyMix != NULL && |
| policyMix->mMixType == MIX_TYPE_RECORDERS) { |
| setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX, |
| AUDIO_POLICY_DEVICE_STATE_AVAILABLE, |
| address, |
| "remote-submix", |
| AUDIO_FORMAT_DEFAULT); |
| } |
| |
| return NO_ERROR; |
| } |
| |
| status_t AudioPolicyManagerCustom::checkAndSetVolume(audio_stream_type_t stream, |
| int index, |
| const sp<AudioOutputDescriptor>& 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->isMuted(streamToVolumeSource(stream))) { |
| ALOGVV("%s() stream %d muted count %d", __func__, stream, outputDesc->getMuteCount(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->devices().types(); |
| } |
| |
| 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)getVolumeCurves(stream).getVolumeIndexMax(); |
| } else { |
| voiceVolume = 1.0; |
| } |
| |
| if (voiceVolume != mLastVoiceVolume) { |
| mpClientInterface->setVoiceVolume(voiceVolume, delayMs); |
| mLastVoiceVolume = voiceVolume; |
| } |
| } else if (mApmConfigs->isFMPowerOptEnabled() && |
| 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); |
| } |
| } |
| |
| return NO_ERROR; |
| } |
| |
| bool static tryForDirectPCM(audio_output_flags_t flags) |
| { |
| bool trackDirectPCM = false; // Output request for track created by other apps |
| |
| if (flags == AUDIO_OUTPUT_FLAG_NONE) { |
| if (AudioPolicyManagerCustom::mApmConfigs != NULL) |
| trackDirectPCM = AudioPolicyManagerCustom::mApmConfigs->isAudioTrackOffloadEnabled(); |
| } |
| return trackDirectPCM; |
| } |
| |
| 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, |
| const audio_config_t *config, |
| audio_output_flags_t *flags, |
| audio_port_handle_t *selectedDeviceId, |
| audio_port_handle_t *portId, |
| std::vector<audio_io_handle_t> *secondaryOutputs) |
| { |
| audio_offload_info_t tOffloadInfo = AUDIO_INFO_INITIALIZER; |
| audio_config_t tConfig; |
| |
| uint32_t bitWidth = (audio_bytes_per_sample(config->format) * 8); |
| |
| memcpy(&tConfig, config, sizeof(audio_config_t)); |
| if ((*flags == AUDIO_OUTPUT_FLAG_DIRECT || tryForDirectPCM(*flags)) && |
| (!memcmp(&config->offload_info, &tOffloadInfo, sizeof(audio_offload_info_t)))) { |
| tConfig.offload_info.sample_rate = config->sample_rate; |
| tConfig.offload_info.channel_mask = config->channel_mask; |
| tConfig.offload_info.format = config->format; |
| tConfig.offload_info.stream_type = *stream; |
| tConfig.offload_info.bit_width = bitWidth; |
| if (attr != NULL) { |
| ALOGV("found attribute .. setting usage %d ", attr->usage); |
| tConfig.offload_info.usage = attr->usage; |
| } else { |
| ALOGI("%s:: attribute is NULL .. no usage set", __func__); |
| } |
| } |
| |
| return AudioPolicyManager::getOutputForAttr(attr, output, session, stream, |
| (uid_t)uid, &tConfig, |
| flags, |
| (audio_port_handle_t*)selectedDeviceId, |
| portId, |
| secondaryOutputs); |
| } |
| |
| audio_io_handle_t AudioPolicyManagerCustom::getOutputForDevices( |
| const DeviceVector &devices, |
| audio_session_t session, |
| audio_stream_type_t stream, |
| const audio_config_t *config, |
| audio_output_flags_t *flags) |
| { |
| audio_io_handle_t output = AUDIO_IO_HANDLE_NONE; |
| status_t status; |
| |
| if (stream < AUDIO_STREAM_MIN || stream >= AUDIO_STREAM_CNT) { |
| ALOGE("%s: invalid stream %d", __func__, stream); |
| return AUDIO_IO_HANDLE_NONE; |
| } |
| |
| 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 (mApmConfigs->isCompressVOIPEnabled()) { |
| if (stream == AUDIO_STREAM_VOICE_CALL && |
| audio_is_linear_pcm(config->format)) { |
| // let voice stream to go with primary output by default |
| // in case direct voip is bypassed |
| bool use_primary_out = true; |
| |
| if ((config->channel_mask == 1) && |
| (config->sample_rate == 8000 || config->sample_rate == 16000 || |
| config->sample_rate == 32000 || config->sample_rate == 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 voipOutCount = 1, voipSampleRate = 1; |
| |
| String8 valueStr = mpClientInterface->getParameters((audio_io_handle_t)0, |
| String8("voip_out_stream_count")); |
| AudioParameter 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 ((voipOutCount == 0) && |
| ((voipSampleRate == 0) || (voipSampleRate == config->sample_rate))) { |
| if (mApmConfigs->useVoicePathForPCMVOIP() |
| && (config->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"); |
| use_primary_out = false; |
| } |
| } |
| } |
| |
| if (use_primary_out) { |
| *flags = (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_FAST|AUDIO_OUTPUT_FLAG_PRIMARY); |
| } |
| } |
| } else { |
| if (stream == AUDIO_STREAM_VOICE_CALL && |
| audio_is_linear_pcm(config->format)) { |
| //check if VoIP output is not opened already |
| bool voip_pcm_already_in_use = false; |
| for (size_t i = 0; i < mOutputs.size(); i++) { |
| sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i); |
| if (desc->mFlags == (AUDIO_OUTPUT_FLAG_VOIP_RX | AUDIO_OUTPUT_FLAG_DIRECT)) { |
| //close voip output if currently open by the same client with different device |
| if (desc->mDirectClientSession == session && |
| desc->devices() != devices) { |
| closeOutput(desc->mIoHandle); |
| } else { |
| voip_pcm_already_in_use = true; |
| ALOGD("VoIP PCM already in use"); |
| } |
| break; |
| } |
| } |
| |
| if (!voip_pcm_already_in_use) { |
| *flags = (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_VOIP_RX | |
| AUDIO_OUTPUT_FLAG_DIRECT); |
| ALOGV("Set VoIP and Direct output flags for PCM format"); |
| } |
| } |
| } /* compress_voip_enabled */ |
| |
| //IF VOIP is going to be started at the same time as when |
| //vr is enabled, get VOIP to fallback to low latency |
| String8 vr_value; |
| String8 value_Str; |
| bool is_vr_mode_on = false; |
| AudioParameter ret; |
| |
| value_Str = mpClientInterface->getParameters((audio_io_handle_t)0, |
| String8("vr_audio_mode_on")); |
| ret = AudioParameter(value_Str); |
| if (ret.get(String8("vr_audio_mode_on"), vr_value) == NO_ERROR) { |
| is_vr_mode_on = vr_value.contains("true"); |
| ALOGI("VR mode is %d, switch to primary output if request is for fast|raw", |
| is_vr_mode_on); |
| } |
| |
| if (is_vr_mode_on) { |
| //check the flags being requested for, and clear FAST|RAW |
| *flags = (audio_output_flags_t)(*flags & |
| (~(AUDIO_OUTPUT_FLAG_FAST|AUDIO_OUTPUT_FLAG_RAW))); |
| |
| } |
| |
| if (mApmConfigs->isVoiceConcEnabled()) { |
| bool prop_play_enabled = false, prop_voip_enabled = false; |
| prop_play_enabled = mApmConfigs->isVoicePlayConcDisabled(); |
| prop_voip_enabled = mApmConfigs->isVoiceVOIPConcDisabled(); |
| |
| bool isDeepBufferFallBackNeeded = |
| ((AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT_PCM) & *flags); |
| bool isFastFallBackNeeded = |
| ((AUDIO_OUTPUT_FLAG_DEEP_BUFFER | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT_PCM) & *flags); |
| |
| 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 (isFastFallBackNeeded && |
| (AUDIO_OUTPUT_FLAG_FAST == mFallBackflag)) { |
| ALOGD("voice_conc:IN call mode adding ULL flags .. flags: %x ", *flags ); |
| *flags = AUDIO_OUTPUT_FLAG_FAST; |
| } else if (isDeepBufferFallBackNeeded && |
| (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; |
| } |
| } |
| } |
| } |
| if (mApmConfigs->isRecPlayConcEnabled()) { |
| bool prop_rec_play_enabled = mApmConfigs->isRecPlayConcDisabled(); |
| DeviceVector primaryInputDevices = availablePrimaryModuleInputDevices(); |
| if ((prop_rec_play_enabled) && |
| ((true == mIsInputRequestOnProgress) || |
| (mInputs.activeInputsCountOnDevices(primaryInputDevices) > 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; |
| } |
| } |
| |
| /* |
| * 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. |
| */ |
| if (mApmConfigs->isAFEProxyEnabled()) { |
| 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; |
| } |
| } |
| |
| // 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); |
| } |
| |
| // Do internal direct magic here |
| bool offload_disabled = property_get_bool("audio.offload.disable", false); |
| if ((*flags == AUDIO_OUTPUT_FLAG_NONE) && |
| (stream == AUDIO_STREAM_MUSIC) && |
| ( !offload_disabled) && |
| ((config->offload_info.usage == AUDIO_USAGE_MEDIA) || |
| (config->offload_info.usage == AUDIO_USAGE_GAME))) { |
| audio_output_flags_t old_flags = *flags; |
| *flags = (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_DIRECT); |
| ALOGD("Force Direct Flag .. old flags(0x%x)", old_flags); |
| } else if (*flags == AUDIO_OUTPUT_FLAG_DIRECT && |
| (offload_disabled || stream != AUDIO_STREAM_MUSIC)) { |
| ALOGD("Offloading is disabled or Stream is not music --> Force Remove Direct Flag"); |
| *flags = (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_NONE); |
| } |
| |
| // check if direct output for pcm/track offload already exits |
| bool direct_pcm_already_in_use = false; |
| if (*flags == AUDIO_OUTPUT_FLAG_DIRECT) { |
| for (size_t i = 0; i < mOutputs.size(); i++) { |
| sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i); |
| if (desc->mFlags == AUDIO_OUTPUT_FLAG_DIRECT) { |
| direct_pcm_already_in_use = true; |
| ALOGD("Direct PCM already in use"); |
| break; |
| } |
| } |
| // prevent direct pcm for non-music stream blindly if direct pcm already in use |
| // for other music stream concurrency is handled after checking direct ouput usage |
| // and checking client |
| if (direct_pcm_already_in_use == true && stream != AUDIO_STREAM_MUSIC) { |
| ALOGD("disabling offload for non music stream as direct pcm is already in use"); |
| *flags = (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_NONE); |
| } |
| } |
| |
| 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 || *flags == AUDIO_OUTPUT_FLAG_DIRECT) && |
| mApmConfigs->isAudioDeepbufferMediaEnabled() && !isInCall()) { |
| forced_deep = true; |
| } |
| |
| if (stream == AUDIO_STREAM_TTS) { |
| *flags = AUDIO_OUTPUT_FLAG_TTS; |
| } else if (stream == AUDIO_STREAM_VOICE_CALL && |
| audio_is_linear_pcm(config->format)) { |
| *flags = (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_VOIP_RX | |
| AUDIO_OUTPUT_FLAG_DIRECT); |
| ALOGV("Set VoIP and Direct output flags for PCM format"); |
| } else if (devices.types() == AUDIO_DEVICE_OUT_TELEPHONY_TX && |
| stream == AUDIO_STREAM_MUSIC && |
| audio_is_linear_pcm(config->format) && |
| isInCall()) { |
| *flags = (audio_output_flags_t)AUDIO_OUTPUT_FLAG_INCALL_MUSIC; |
| } |
| |
| 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(config->format) && config->sample_rate <= SAMPLE_RATE_HZ_MAX && |
| audio_channel_count_from_out_mask(config->channel_mask) <= 2) { |
| goto non_direct_output; |
| } |
| |
| // Do not allow offloading if one non offloadable effect is enabled or MasterMono is enabled. |
| // This prevents 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)) && |
| (mEffects.isNonOffloadableEffectEnabled() || mMasterMono)) { |
| ALOGD("non offloadable effect is enabled, try with non direct output"); |
| goto non_direct_output; |
| } |
| |
| profile = getProfileForOutput(devices, |
| config->sample_rate, |
| config->format, |
| config->channel_mask, |
| (audio_output_flags_t)*flags, |
| true /* directOnly */); |
| |
| if (profile != 0) { |
| |
| if (!(*flags & AUDIO_OUTPUT_FLAG_DIRECT) && |
| (profile->getFlags() & AUDIO_OUTPUT_FLAG_DIRECT)) { |
| ALOGI("got Direct without requesting ... reject "); |
| profile = NULL; |
| goto non_direct_output; |
| } |
| |
| if ((*flags & AUDIO_OUTPUT_FLAG_MMAP_NOIRQ) == 0 || output != AUDIO_IO_HANDLE_NONE) { |
| sp<SwAudioOutputDescriptor> 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 (!mApmConfigs->isAudioMultipleOffloadEnable() && |
| ((*flags & AUDIO_OUTPUT_FLAG_DIRECT) != 0)) { |
| 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 by the same client |
| // and configured with same parameters |
| if ((config->sample_rate == desc->mSamplingRate) && |
| audio_formats_match(config->format, desc->mFormat) && |
| (config->channel_mask == desc->mChannelMask) && |
| (session == desc->mDirectClientSession)) { |
| desc->mDirectOpenCount++; |
| ALOGV("getOutputForDevice() reusing direct output %d for session %d", |
| mOutputs.keyAt(i), session); |
| return mOutputs.keyAt(i); |
| } |
| } |
| if (outputDesc != NULL) { |
| if (*flags == AUDIO_OUTPUT_FLAG_DIRECT && |
| direct_pcm_already_in_use == true && |
| session != outputDesc->mDirectClientSession) { |
| ALOGV("getOutput() do not reuse direct pcm output because current client (%d) " |
| "is not the same as requesting client (%d) for different output conf", |
| outputDesc->mDirectClientSession, session); |
| goto non_direct_output; |
| } |
| closeOutput(outputDesc->mIoHandle); |
| } |
| } |
| if (!profile->canOpenNewIo()) { |
| goto non_direct_output; |
| } |
| |
| outputDesc = |
| new SwAudioOutputDescriptor(profile, mpClientInterface); |
| DeviceVector outputDevices = mAvailableOutputDevices.getDevicesFromTypeMask(devices.types()); |
| String8 address = outputDevices.size() > 0 ? outputDevices.itemAt(0)->address() |
| : String8(""); |
| status = outputDesc->open(config, devices, stream, *flags, &output); |
| |
| // only accept an output with the requested parameters |
| if (status != NO_ERROR || |
| (config->sample_rate != 0 && config->sample_rate != outputDesc->mSamplingRate) || |
| (config->format != AUDIO_FORMAT_DEFAULT && |
| !audio_formats_match(config->format, outputDesc->mFormat)) || |
| (config->channel_mask != 0 && config->channel_mask != outputDesc->mChannelMask)) { |
| ALOGV("getOutputForDevice() failed opening direct output: output %d sample rate %d %d," |
| "format %d %d, channel mask %04x %04x", output, config->sample_rate, |
| outputDesc->mSamplingRate, config->format, outputDesc->mFormat, |
| config->channel_mask, outputDesc->mChannelMask); |
| if (output != AUDIO_IO_HANDLE_NONE) { |
| outputDesc->close(); |
| } |
| // fall back to mixer output if possible when the direct output could not be open |
| if (audio_is_linear_pcm(config->format) && config->sample_rate <= SAMPLE_RATE_HZ_MAX) { |
| goto non_direct_output; |
| } |
| return AUDIO_IO_HANDLE_NONE; |
| } |
| outputDesc->mDirectOpenCount = 1; |
| outputDesc->mDirectClientSession = session; |
| |
| addOutput(output, outputDesc); |
| mPreviousOutputs = mOutputs; |
| ALOGV("getOutputForDevice() returns new direct output %d", output); |
| mpClientInterface->onAudioPortListUpdate(); |
| return output; |
| } |
| } |
| } |
| |
| non_direct_output: |
| |
| // A request for HW A/V sync cannot fallback to a mixed output because time |
| // stamps are embedded in audio data |
| if ((*flags & (AUDIO_OUTPUT_FLAG_HW_AV_SYNC | AUDIO_OUTPUT_FLAG_MMAP_NOIRQ)) != 0) { |
| return AUDIO_IO_HANDLE_NONE; |
| } |
| |
| // 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(config->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 = getOutputsForDevices(devices, 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); |
| |
| if (forced_deep) { |
| *flags = (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_DEEP_BUFFER); |
| ALOGI("setting force DEEP buffer now "); |
| } else if (*flags == AUDIO_OUTPUT_FLAG_NONE) { |
| // no deep buffer playback is requested hence fallback to primary |
| *flags = (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_PRIMARY); |
| ALOGI("FLAG None hence request for a primary output"); |
| } |
| |
| output = selectOutput(outputs, *flags, config->format); |
| } |
| |
| ALOGW_IF((output == 0), "getOutputForDevice() could not find output for stream %d, " |
| "sampling rate %d, format %#x, channels %#x, flags %#x", |
| stream, config->sample_rate, config->format, config->channel_mask, *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, |
| const audio_config_base_t *config, |
| audio_input_flags_t flags, |
| audio_port_handle_t *selectedDeviceId, |
| input_type_t *inputType, |
| audio_port_handle_t *portId) |
| { |
| audio_source_t inputSource; |
| inputSource = attr->source; |
| |
| if (mApmConfigs->isVoiceConcEnabled()) { |
| bool prop_rec_enabled = false, prop_voip_enabled = false; |
| prop_rec_enabled = mApmConfigs->isVoiceRecConcDisabled(); |
| prop_voip_enabled = mApmConfigs->isVoiceVOIPConcDisabled(); |
| |
| 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; |
| } |
| } |
| } |
| } |
| |
| |
| return AudioPolicyManager::getInputForAttr(attr, |
| input, |
| session, |
| uid, |
| config, |
| flags, |
| selectedDeviceId, |
| inputType, |
| portId); |
| } |
| |
| uint32_t AudioPolicyManagerCustom::activeNonSoundTriggerInputsCountOnDevices(audio_devices_t devices) const |
| { |
| uint32_t count = 0; |
| for (size_t i = 0; i < mInputs.size(); i++) { |
| const sp<AudioInputDescriptor> inputDescriptor = mInputs.valueAt(i); |
| if (inputDescriptor->isActive() && !inputDescriptor->isSoundTrigger() && |
| ((devices == AUDIO_DEVICE_IN_DEFAULT) || |
| ((inputDescriptor->getDeviceType() & devices & ~AUDIO_DEVICE_BIT_IN) != 0))) { |
| count++; |
| } |
| } |
| return count; |
| } |
| |
| status_t AudioPolicyManagerCustom::startInput(audio_port_handle_t portId) |
| { |
| ALOGV("%s portId %d", __FUNCTION__, portId); |
| |
| sp<AudioInputDescriptor> inputDesc = mInputs.getInputForClient(portId); |
| if (inputDesc == 0) { |
| ALOGW("%s no input for client %d", __FUNCTION__, portId); |
| return BAD_VALUE; |
| } |
| |
| audio_io_handle_t input = inputDesc->mIoHandle; |
| sp<RecordClientDescriptor> client = inputDesc->getClient(portId); |
| if (client->active()) { |
| ALOGW("%s input %d client %d already started", __FUNCTION__, input, client->portId()); |
| return INVALID_OPERATION; |
| } |
| |
| audio_session_t session = client->session(); |
| |
| ALOGV("%s input:%d, session:%d)", __FUNCTION__, input, session); |
| |
| |
| // FIXME: disable concurrent capture until UI is ready |
| #if 0 |
| if (!isConcurentCaptureAllowed(inputDesc, audioSession)) { |
| ALOGW("startInput(%d) failed: other input already started", input); |
| return INVALID_OPERATION; |
| } |
| |
| if (isInCall()) { |
| *concurrency |= API_INPUT_CONCURRENCY_CALL; |
| } |
| |
| if (mInputs.activeInputsCountOnDevices() != 0) { |
| *concurrency |= API_INPUT_CONCURRENCY_CAPTURE; |
| } |
| #endif |
| |
| if (mApmConfigs->isRecPlayConcEnabled()) { |
| mIsInputRequestOnProgress = true; |
| DeviceVector primaryInputDevices = availablePrimaryModuleInputDevices(); |
| if (mApmConfigs->isRecPlayConcDisabled() && (mInputs.activeInputsCountOnDevices(primaryInputDevices) == 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 < AUDIO_STREAM_FOR_POLICY_CNT; i++) { |
| // Do not call invalidate for ENFORCED_AUDIBLE (otherwise pops are seen for camcorder) |
| if (i != AUDIO_STREAM_ENFORCED_AUDIBLE) { |
| 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->getFlags() |
| & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { |
| // close compress sessions |
| ALOGD("calling closeOutput on record conc for COMPRESS output"); |
| closeOutput(mOutputs.keyAt(i)); |
| } |
| } |
| } |
| } |
| status_t status = inputDesc->start(); |
| if (status != NO_ERROR) { |
| return status; |
| } |
| |
| // increment activity count before calling getNewInputDevice() below as only active sessions |
| // are considered for device selection |
| inputDesc->setClientActive(client, true); |
| |
| // indicate active capture to sound trigger service if starting capture from a mic on |
| // primary HW module |
| sp<DeviceDescriptor> device = getNewInputDevice(inputDesc); |
| setInputDevice(input, device, true /* force */); |
| |
| if (inputDesc->activeCount() == 1) { |
| // 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->mDeviceAddress, |
| MIX_STATE_MIXING); |
| } |
| |
| DeviceVector primaryInputDevices = availablePrimaryModuleInputDevices(); |
| if ((primaryInputDevices.contains(device) && (device->type() & ~AUDIO_DEVICE_BIT_IN)) != 0) { |
| if (mApmConfigs->isVAConcEnabled()) { |
| if (activeNonSoundTriggerInputsCountOnDevices(primaryInputDevices.types()) == 1) |
| SoundTrigger::setCaptureState(true); |
| } else if (mInputs.activeInputsCountOnDevices(primaryInputDevices) == 1) |
| SoundTrigger::setCaptureState(true); |
| } |
| |
| // 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->getDeviceType())) { |
| String8 address = String8(""); |
| if (inputDesc->mPolicyMix == NULL) { |
| address = String8("0"); |
| } else if (inputDesc->mPolicyMix->mMixType == MIX_TYPE_PLAYERS) { |
| address = inputDesc->mPolicyMix->mDeviceAddress; |
| } |
| if (address != "") { |
| setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, |
| AUDIO_POLICY_DEVICE_STATE_AVAILABLE, |
| address, "remote-submix", AUDIO_FORMAT_DEFAULT); |
| } |
| } |
| } |
| |
| ALOGV("%s input %d source = %d exit", __FUNCTION__, input, client->source()); |
| |
| if (mApmConfigs->isRecPlayConcEnabled()) |
| mIsInputRequestOnProgress = false; |
| return NO_ERROR; |
| } |
| |
| status_t AudioPolicyManagerCustom::stopInput(audio_port_handle_t portId) |
| { |
| status_t status; |
| status = AudioPolicyManager::stopInput(portId); |
| sp<AudioInputDescriptor> inputDesc = mInputs.getInputForClient(portId); |
| if (inputDesc == 0) { |
| ALOGW("stopInput() no input for client %d", portId); |
| return BAD_VALUE; |
| } |
| sp<RecordClientDescriptor> client = inputDesc->getClient(portId); |
| audio_io_handle_t input = inputDesc->mIoHandle; |
| |
| ALOGV("stopInput() input %d", input); |
| DeviceVector primaryInputDevices = availablePrimaryModuleInputDevices(); |
| if (mApmConfigs->isVAConcEnabled()) { |
| sp<AudioInputDescriptor> inputDesc = mInputs.getInputForClient(portId); |
| if ((primaryInputDevices.contains(inputDesc->getDevice()) && |
| activeNonSoundTriggerInputsCountOnDevices(primaryInputDevices.types())) == 0) { |
| SoundTrigger::setCaptureState(false); |
| } |
| } |
| if (mApmConfigs->isRecPlayConcEnabled()) { |
| if (mApmConfigs->isRecPlayConcDisabled() && |
| (mInputs.activeInputsCountOnDevices(primaryInputDevices) == 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); |
| } |
| } |
| } |
| } |
| return status; |
| } |
| |
| AudioPolicyManagerCustom::AudioPolicyManagerCustom(AudioPolicyClientInterface *clientInterface) |
| : AudioPolicyManager(clientInterface), |
| mFallBackflag(AUDIO_OUTPUT_FLAG_NONE), |
| mHdmiAudioDisabled(false), |
| mHdmiAudioEvent(false), |
| mPrevPhoneState(0), |
| mIsInputRequestOnProgress(false), |
| mPrevFMVolumeDb(0.0f), |
| mFMIsActive(false) |
| { |
| if (mApmConfigs->useXMLAudioPolicyConf()) |
| ALOGD("USE_XML_AUDIO_POLICY_CONF is TRUE"); |
| else |
| ALOGD("USE_XML_AUDIO_POLICY_CONF is FALSE"); |
| |
| if (mApmConfigs->isVoiceConcEnabled()) |
| mFallBackflag = getFallBackPath(); |
| } |
| |
| } |