| /* |
| * Copyright (c) 2013-2014, 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 "AudioPolicyManager" |
| //#define LOG_NDEBUG 0 |
| |
| //#define VERY_VERBOSE_LOGGING |
| #ifdef VERY_VERBOSE_LOGGING |
| #define ALOGVV ALOGV |
| #else |
| #define ALOGVV(a...) do { } while(0) |
| #endif |
| |
| // A device mask for all audio input devices that are considered "virtual" when evaluating |
| // active inputs in getActiveInput() |
| #ifdef AUDIO_EXTN_FM_ENABLED |
| #define APM_AUDIO_IN_DEVICE_VIRTUAL_ALL (AUDIO_DEVICE_IN_REMOTE_SUBMIX | AUDIO_DEVICE_IN_FM_RX_A2DP) |
| #else |
| #define APM_AUDIO_IN_DEVICE_VIRTUAL_ALL AUDIO_DEVICE_IN_REMOTE_SUBMIX |
| #endif |
| // A device mask for all audio output devices that are considered "remote" when evaluating |
| // active output devices in isStreamActiveRemotely() |
| #define APM_AUDIO_OUT_DEVICE_REMOTE_ALL AUDIO_DEVICE_OUT_REMOTE_SUBMIX |
| // A device mask for all audio input and output devices where matching inputs/outputs on device |
| // type alone is not enough: the address must match too |
| #define APM_AUDIO_DEVICE_MATCH_ADDRESS_ALL (AUDIO_DEVICE_IN_REMOTE_SUBMIX | \ |
| AUDIO_DEVICE_OUT_REMOTE_SUBMIX) |
| |
| #include <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" |
| |
| namespace android { |
| |
| // ---------------------------------------------------------------------------- |
| // AudioPolicyInterface implementation |
| // ---------------------------------------------------------------------------- |
| |
| status_t AudioPolicyManagerCustom::setDeviceConnectionState(audio_devices_t device, |
| audio_policy_dev_state_t state, |
| const char *device_address) |
| { |
| String8 address = (device_address == NULL) ? String8("") : String8(device_address); |
| |
| ALOGV("setDeviceConnectionState() device: %x, state %d, address %s", |
| device, state, address.string()); |
| |
| // connect/disconnect only 1 device at a time |
| if (!audio_is_output_device(device) && !audio_is_input_device(device)) return BAD_VALUE; |
| |
| // handle output devices |
| if (audio_is_output_device(device)) { |
| SortedVector <audio_io_handle_t> outputs; |
| |
| sp<DeviceDescriptor> devDesc = new DeviceDescriptor(String8(""), device); |
| devDesc->mAddress = address; |
| 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); |
| } |
| } |
| #endif |
| if (index >= 0) { |
| sp<HwModule> module = getModuleForDevice(device); |
| if (module == 0) { |
| ALOGD("setDeviceConnectionState() could not find HW module for device %08x", |
| device); |
| mAvailableOutputDevices.remove(devDesc); |
| return INVALID_OPERATION; |
| } |
| mAvailableOutputDevices[index]->mId = nextUniqueId(); |
| mAvailableOutputDevices[index]->mModule = module; |
| } else { |
| return NO_MEMORY; |
| } |
| |
| if (checkOutputsForDevice(devDesc, state, outputs, address) != NO_ERROR) { |
| mAvailableOutputDevices.remove(devDesc); |
| return INVALID_OPERATION; |
| } |
| // 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()); |
| 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); |
| |
| // Set Disconnect to HALs |
| AudioParameter param = AudioParameter(address); |
| 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, address); |
| } 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<AudioOutputDescriptor> desc = mOutputs.valueFor(outputs[i]); |
| // close unused outputs after device disconnection or direct outputs that have been |
| // opened by checkOutputsForDevice() to query dynamic parameters |
| if ((state == 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(); |
| audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/); |
| if (mPhoneState == AUDIO_MODE_IN_CALL) { |
| updateCallRouting(newDevice); |
| } |
| |
| #ifdef AUDIO_EXTN_FM_ENABLED |
| if(device == AUDIO_DEVICE_OUT_FM) { |
| if (state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) { |
| mOutputs.valueFor(mPrimaryOutput)->changeRefCount(AUDIO_STREAM_MUSIC, 1); |
| newDevice = (audio_devices_t)(getNewOutputDevice(mPrimaryOutput, false) | AUDIO_DEVICE_OUT_FM); |
| } else { |
| mOutputs.valueFor(mPrimaryOutput)->changeRefCount(AUDIO_STREAM_MUSIC, -1); |
| } |
| |
| AudioParameter param = AudioParameter(); |
| param.addInt(String8("handle_fm"), (int)newDevice); |
| ALOGV("setDeviceConnectionState() setParameters handle_fm"); |
| mpClientInterface->setParameters(mPrimaryOutput, param.toString()); |
| } |
| #endif |
| for (size_t i = 0; i < mOutputs.size(); i++) { |
| audio_io_handle_t output = mOutputs.keyAt(i); |
| if ((mPhoneState != AUDIO_MODE_IN_CALL) || (output != mPrimaryOutput)) { |
| audio_devices_t newDevice = getNewOutputDevice(mOutputs.keyAt(i), |
| 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 = !mOutputs.valueAt(i)->isDuplicated() |
| && (!deviceDistinguishesOnAddress(device) |
| // always force when disconnecting (a non-duplicated device) |
| || (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE)); |
| setOutputDevice(output, 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; |
| |
| sp<DeviceDescriptor> devDesc = new DeviceDescriptor(String8(""), device); |
| devDesc->mAddress = address; |
| 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 = getModuleForDevice(device); |
| if (module == NULL) { |
| ALOGW("setDeviceConnectionState(): could not find HW module for device %08x", |
| device); |
| return INVALID_OPERATION; |
| } |
| if (checkInputsForDevice(device, state, inputs, address) != NO_ERROR) { |
| return INVALID_OPERATION; |
| } |
| |
| index = mAvailableInputDevices.add(devDesc); |
| if (index >= 0) { |
| mAvailableInputDevices[index]->mId = nextUniqueId(); |
| mAvailableInputDevices[index]->mModule = module; |
| } else { |
| return NO_MEMORY; |
| } |
| } 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(address); |
| param.addInt(String8(AUDIO_PARAMETER_DEVICE_DISCONNECT), device); |
| mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString()); |
| |
| checkInputsForDevice(device, state, inputs, address); |
| mAvailableInputDevices.remove(devDesc); |
| |
| } break; |
| |
| default: |
| ALOGE("setDeviceConnectionState() invalid state: %x", state); |
| return BAD_VALUE; |
| } |
| |
| closeAllInputs(); |
| |
| if (mPhoneState == AUDIO_MODE_IN_CALL) { |
| audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/); |
| updateCallRouting(newDevice); |
| } |
| |
| mpClientInterface->onAudioPortListUpdate(); |
| return NO_ERROR; |
| } // end if is input device |
| |
| ALOGW("setDeviceConnectionState() invalid device: %x", device); |
| return BAD_VALUE; |
| } |
| |
| audio_policy_dev_state_t AudioPolicyManagerCustom::getDeviceConnectionState(audio_devices_t device, |
| const char *device_address) |
| { |
| audio_policy_dev_state_t state = AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE; |
| sp<DeviceDescriptor> devDesc = new DeviceDescriptor(String8(""), device); |
| devDesc->mAddress = (device_address == NULL) ? String8("") : String8(device_address); |
| ssize_t index; |
| DeviceVector *deviceVector; |
| |
| if (audio_is_output_device(device)) { |
| deviceVector = &mAvailableOutputDevices; |
| } else if (audio_is_input_device(device)) { |
| deviceVector = &mAvailableInputDevices; |
| } else { |
| ALOGW("getDeviceConnectionState() invalid device type %08x", device); |
| return AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE; |
| } |
| |
| index = deviceVector->indexOf(devDesc); |
| if (index >= 0) { |
| return AUDIO_POLICY_DEVICE_STATE_AVAILABLE; |
| } else { |
| return AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE; |
| } |
| } |
| |
| void AudioPolicyManagerCustom::setPhoneState(audio_mode_t state) |
| { |
| ALOGD("setPhoneState() state %d", state); |
| audio_devices_t newDevice = AUDIO_DEVICE_NONE; |
| |
| if (state < 0 || state >= AUDIO_MODE_CNT) { |
| ALOGW("setPhoneState() invalid state %d", state); |
| return; |
| } |
| |
| if (state == mPhoneState ) { |
| ALOGW("setPhoneState() setting same state %d", state); |
| return; |
| } |
| |
| // if leaving call state, handle special case of active streams |
| // pertaining to sonification strategy see handleIncallSonification() |
| if (isInCall()) { |
| ALOGV("setPhoneState() in call state management: new state is %d", state); |
| for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) { |
| handleIncallSonification((audio_stream_type_t)stream, false, true); |
| } |
| } |
| |
| // store previous phone state for management of sonification strategy below |
| int oldState = mPhoneState; |
| mPhoneState = state; |
| bool force = false; |
| |
| // are we entering or starting a call |
| if (!isStateInCall(oldState) && isStateInCall(state)) { |
| ALOGV(" Entering call in setPhoneState()"); |
| // force routing command to audio hardware when starting a call |
| // even if no device change is needed |
| force = true; |
| for (int j = 0; j < DEVICE_CATEGORY_CNT; j++) { |
| mStreams[AUDIO_STREAM_DTMF].mVolumeCurve[j] = |
| sVolumeProfiles[AUDIO_STREAM_VOICE_CALL][j]; |
| } |
| } else if (isStateInCall(oldState) && !isStateInCall(state)) { |
| ALOGV(" Exiting call in setPhoneState()"); |
| // force routing command to audio hardware when exiting a call |
| // even if no device change is needed |
| force = true; |
| for (int j = 0; j < DEVICE_CATEGORY_CNT; j++) { |
| mStreams[AUDIO_STREAM_DTMF].mVolumeCurve[j] = |
| sVolumeProfiles[AUDIO_STREAM_DTMF][j]; |
| } |
| } else if (isStateInCall(state) && (state != oldState)) { |
| ALOGV(" Switching between telephony and VoIP in setPhoneState()"); |
| // force routing command to audio hardware when switching between telephony and VoIP |
| // even if no device change is needed |
| force = true; |
| } |
| |
| // check for device and output changes triggered by new phone state |
| checkA2dpSuspend(); |
| checkOutputForAllStrategies(); |
| updateDevicesAndOutputs(); |
| |
| sp<AudioOutputDescriptor> hwOutputDesc = mOutputs.valueFor(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("SetPhoneState: Voice call state = %d", voice_call_state); |
| } |
| |
| if (mode_in_call && voice_call_state) { |
| ALOGD("Entering to call mode oldState :: %d state::%d ",oldState, state); |
| mvoice_call_state = voice_call_state; |
| if (prop_playback_enabled) { |
| //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++) { |
| ALOGV(" Invalidate on call mode for stream :: %d ", i); |
| //FIXME see fixme on name change |
| mpClientInterface->invalidateStream((audio_stream_type_t)i); |
| } |
| } |
| |
| if (prop_rec_enabled) { |
| //Close all active inputs |
| audio_io_handle_t activeInput = 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("FOUND active input during call active: %d",activeDesc->mInputSource); |
| break; |
| |
| case AUDIO_SOURCE_VOICE_COMMUNICATION: |
| if(prop_voip_enabled) { |
| ALOGD("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("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 = getActiveInput(); |
| if (activeInput != 0) { |
| sp<AudioInputDescriptor> activeDesc = mInputs.valueFor(activeInput); |
| if (AUDIO_SOURCE_VOICE_COMMUNICATION == activeDesc->mInputSource) { |
| ALOGD("CLOSING VoIP on call setup : %d",activeDesc->mInputSource); |
| stopInput(activeInput, activeDesc->mSessions.itemAt(0)); |
| releaseInput(activeInput, activeDesc->mSessions.itemAt(0)); |
| } |
| } |
| } |
| |
| //suspend PCM (deep-buffer) output & close compress & direct tracks |
| for (size_t i = 0; i < mOutputs.size(); i++) { |
| sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(i); |
| if ( (outputDesc == NULL) || (outputDesc->mProfile == NULL)) { |
| ALOGD("ouput desc / profile is NULL"); |
| continue; |
| } |
| if (((!outputDesc->isDuplicated() &&outputDesc->mProfile->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY)) |
| && prop_playback_enabled) { |
| ALOGD(" 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(" 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(" calling closeOutput on call mode for DIRECT 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("EXITING from call mode oldState :: %d state::%d \n",oldState, state); |
| mvoice_call_state = 0; |
| //restore PCM (deep-buffer) output after call termination |
| for (size_t i = 0; i < mOutputs.size(); i++) { |
| sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(i); |
| if ( (outputDesc == NULL) || (outputDesc->mProfile == NULL)) { |
| ALOGD("ouput desc / profile is NULL"); |
| continue; |
| } |
| if (!outputDesc->isDuplicated() && outputDesc->mProfile->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY) { |
| ALOGD("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++) { |
| ALOGD("Invalidate after call ends for stream :: %d ", i); |
| //FIXME see fixme on name change |
| 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 == mPhoneState) { |
| 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<AudioOutputDescriptor> 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) && |
| (mPhoneState == 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<AudioOutputDescriptor> desc = mOutputs.valueAt(i); |
| // mute media and sonification strategies and delay device switch by the largest |
| // latency of any output where either strategy is active. |
| // This avoid sending the ring tone or music tail into the earpiece or headset. |
| if ((desc->isStrategyActive(STRATEGY_MEDIA, |
| SONIFICATION_HEADSET_MUSIC_DELAY, |
| sysTime) || |
| desc->isStrategyActive(STRATEGY_SONIFICATION, |
| SONIFICATION_HEADSET_MUSIC_DELAY, |
| sysTime)) && |
| (delayMs < (int)desc->mLatency*2)) { |
| delayMs = desc->mLatency*2; |
| } |
| setStrategyMute(STRATEGY_MEDIA, true, mOutputs.keyAt(i)); |
| setStrategyMute(STRATEGY_MEDIA, false, mOutputs.keyAt(i), MUTE_TIME_MS, |
| getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/)); |
| setStrategyMute(STRATEGY_SONIFICATION, true, mOutputs.keyAt(i)); |
| setStrategyMute(STRATEGY_SONIFICATION, false, mOutputs.keyAt(i), MUTE_TIME_MS, |
| getDeviceForStrategy(STRATEGY_SONIFICATION, true /*fromCache*/)); |
| } |
| } |
| |
| // Note that despite the fact that getNewOutputDevice() is called on the primary output, |
| // the device returned is not necessarily reachable via this output |
| audio_devices_t rxDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/); |
| // force routing command to audio hardware when ending call |
| // even if no device change is needed |
| if (isStateInCall(oldState) && rxDevice == AUDIO_DEVICE_NONE) { |
| rxDevice = hwOutputDesc->device(); |
| } |
| |
| if (state == AUDIO_MODE_IN_CALL) { |
| updateCallRouting(rxDevice, delayMs); |
| } else if (oldState == AUDIO_MODE_IN_CALL) { |
| if (mCallRxPatch != 0) { |
| mpClientInterface->releaseAudioPatch(mCallRxPatch->mAfPatchHandle, 0); |
| mCallRxPatch.clear(); |
| } |
| if (mCallTxPatch != 0) { |
| mpClientInterface->releaseAudioPatch(mCallTxPatch->mAfPatchHandle, 0); |
| mCallTxPatch.clear(); |
| } |
| setOutputDevice(mPrimaryOutput, rxDevice, force, 0); |
| } else { |
| setOutputDevice(mPrimaryOutput, rxDevice, force, 0); |
| } |
| |
| //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) { |
| newDevice = getNewOutputDevice(output, false /*fromCache*/); |
| setOutputDevice(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 (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) { |
| handleIncallSonification((audio_stream_type_t)stream, true, true); |
| } |
| } |
| |
| // Flag that ringtone volume must be limited to music volume until we exit MODE_RINGTONE |
| if (state == AUDIO_MODE_RINGTONE && |
| isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY)) { |
| mLimitRingtoneVolume = true; |
| } else { |
| mLimitRingtoneVolume = false; |
| } |
| } |
| |
| 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, mPhoneState); |
| |
| bool forceVolumeReeval = false; |
| switch(usage) { |
| case AUDIO_POLICY_FORCE_FOR_COMMUNICATION: |
| if (config != AUDIO_POLICY_FORCE_SPEAKER && config != AUDIO_POLICY_FORCE_BT_SCO && |
| config != AUDIO_POLICY_FORCE_NONE) { |
| ALOGW("setForceUse() invalid config %d for FOR_COMMUNICATION", config); |
| return; |
| } |
| forceVolumeReeval = true; |
| mForceUse[usage] = config; |
| break; |
| case AUDIO_POLICY_FORCE_FOR_MEDIA: |
| if (config != AUDIO_POLICY_FORCE_HEADPHONES && config != AUDIO_POLICY_FORCE_BT_A2DP && |
| #ifdef AUDIO_EXTN_FM_ENABLED |
| config != AUDIO_POLICY_FORCE_SPEAKER && |
| #endif |
| config != AUDIO_POLICY_FORCE_WIRED_ACCESSORY && |
| config != AUDIO_POLICY_FORCE_ANALOG_DOCK && |
| config != AUDIO_POLICY_FORCE_DIGITAL_DOCK && config != AUDIO_POLICY_FORCE_NONE && |
| config != AUDIO_POLICY_FORCE_NO_BT_A2DP) { |
| ALOGW("setForceUse() invalid config %d for FOR_MEDIA", config); |
| return; |
| } |
| mForceUse[usage] = config; |
| break; |
| case AUDIO_POLICY_FORCE_FOR_RECORD: |
| if (config != AUDIO_POLICY_FORCE_BT_SCO && config != AUDIO_POLICY_FORCE_WIRED_ACCESSORY && |
| config != AUDIO_POLICY_FORCE_NONE) { |
| ALOGW("setForceUse() invalid config %d for FOR_RECORD", config); |
| return; |
| } |
| mForceUse[usage] = config; |
| break; |
| case AUDIO_POLICY_FORCE_FOR_DOCK: |
| if (config != AUDIO_POLICY_FORCE_NONE && config != AUDIO_POLICY_FORCE_BT_CAR_DOCK && |
| config != AUDIO_POLICY_FORCE_BT_DESK_DOCK && |
| config != AUDIO_POLICY_FORCE_WIRED_ACCESSORY && |
| config != AUDIO_POLICY_FORCE_ANALOG_DOCK && |
| config != AUDIO_POLICY_FORCE_DIGITAL_DOCK) { |
| ALOGW("setForceUse() invalid config %d for FOR_DOCK", config); |
| } |
| forceVolumeReeval = true; |
| mForceUse[usage] = config; |
| break; |
| case AUDIO_POLICY_FORCE_FOR_SYSTEM: |
| if (config != AUDIO_POLICY_FORCE_NONE && |
| config != AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) { |
| ALOGW("setForceUse() invalid config %d for FOR_SYSTEM", config); |
| } |
| forceVolumeReeval = true; |
| mForceUse[usage] = config; |
| break; |
| case AUDIO_POLICY_FORCE_FOR_HDMI_SYSTEM_AUDIO: |
| if (config != AUDIO_POLICY_FORCE_NONE && |
| config != AUDIO_POLICY_FORCE_HDMI_SYSTEM_AUDIO_ENFORCED) { |
| ALOGW("setForceUse() invalid config %d forHDMI_SYSTEM_AUDIO", config); |
| } |
| mForceUse[usage] = config; |
| break; |
| default: |
| ALOGW("setForceUse() invalid usage %d", usage); |
| break; |
| } |
| |
| // check for device and output changes triggered by new force usage |
| checkA2dpSuspend(); |
| checkOutputForAllStrategies(); |
| updateDevicesAndOutputs(); |
| if (mPhoneState == AUDIO_MODE_IN_CALL) { |
| audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, true /*fromCache*/); |
| updateCallRouting(newDevice); |
| } |
| for (size_t i = 0; i < mOutputs.size(); i++) { |
| audio_io_handle_t output = mOutputs.keyAt(i); |
| audio_devices_t newDevice = getNewOutputDevice(output, true /*fromCache*/); |
| if ((mPhoneState != AUDIO_MODE_IN_CALL) || (output != mPrimaryOutput)) { |
| setOutputDevice(output, newDevice, (newDevice != AUDIO_DEVICE_NONE)); |
| } |
| if (forceVolumeReeval && (newDevice != AUDIO_DEVICE_NONE)) { |
| applyStreamVolumes(output, newDevice, 0, true); |
| } |
| } |
| |
| audio_io_handle_t activeInput = getActiveInput(); |
| if (activeInput != 0) { |
| setInputDevice(activeInput, getNewInputDevice(activeInput)); |
| } |
| |
| } |
| |
| audio_io_handle_t AudioPolicyManagerCustom::getOutputForDevice( |
| audio_devices_t device, |
| 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 AudioOutputDescriptor(NULL); |
| 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 |
| |
| #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 == mPhoneState) || |
| ((AUDIO_MODE_IN_CALL == mPrevPhoneState) |
| && (AUDIO_MODE_IN_COMMUNICATION == mPhoneState))) |
| { |
| if(AUDIO_OUTPUT_FLAG_VOIP_RX & flags) { |
| if(prop_voip_enabled) { |
| ALOGD(" IN call mode returing no output .. for VoIP usecase flags: %x ", flags ); |
| // flags = (AudioSystem::output_flags)AUDIO_OUTPUT_FLAG_FAST; |
| return 0; |
| } |
| } |
| else { |
| ALOGD(" IN call mode adding ULL flags .. flags: %x ", flags ); |
| flags = AUDIO_OUTPUT_FLAG_FAST; |
| } |
| } |
| } 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 == mPhoneState) || |
| ((AUDIO_MODE_IN_CALL == mPrevPhoneState) |
| && (AUDIO_MODE_IN_COMMUNICATION == mPhoneState))) |
| { |
| if(AUDIO_OUTPUT_FLAG_VOIP_RX & flags) { |
| ALOGD(" IN call mode returing no output .. for VoIP usecase flags: %x ", flags ); |
| // flags = (AudioSystem::output_flags)AUDIO_OUTPUT_FLAG_FAST; |
| return 0; |
| } |
| } |
| } |
| #endif |
| |
| #ifdef WFD_CONCURRENCY |
| audio_devices_t availableOutputDeviceTypes = mAvailableOutputDevices.types(); |
| if ((availableOutputDeviceTypes & AUDIO_DEVICE_OUT_PROXY) |
| && (stream != AUDIO_STREAM_MUSIC)) { |
| ALOGD(" WFD mode adding ULL flags 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 |
| |
| #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) || (activeInputsCount() > 0))) { |
| if (AUDIO_MODE_IN_COMMUNICATION == mPhoneState) { |
| if (AUDIO_OUTPUT_FLAG_VOIP_RX & flags) { |
| // allow VoIP using voice path |
| // Do nothing |
| } else if((flags & AUDIO_OUTPUT_FLAG_FAST) != 0) { |
| ALOGD(" 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(" 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 |
| // 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); |
| } |
| |
| if ((format == AUDIO_FORMAT_PCM_16_BIT) &&(popcount(channelMask) > 2)) { |
| ALOGV("owerwrite flag(%x) for PCM16 multi-channel(CM:%x) playback", flags ,channelMask); |
| flags = AUDIO_OUTPUT_FLAG_DIRECT; |
| } |
| |
| 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) || |
| !isNonOffloadableEffectEnabled()) && |
| flags & AUDIO_OUTPUT_FLAG_DIRECT) { |
| profile = getProfileForDirectOutput(device, |
| samplingRate, |
| format, |
| channelMask, |
| (audio_output_flags_t)flags); |
| } |
| |
| if (profile != 0) { |
| sp<AudioOutputDescriptor> outputDesc = NULL; |
| |
| for (size_t i = 0; i < mOutputs.size(); i++) { |
| sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i); |
| if (!desc->isDuplicated() && (profile == desc->mProfile)) { |
| outputDesc = desc; |
| // reuse direct output if currently open and configured with same parameters |
| if ((samplingRate == outputDesc->mSamplingRate) && |
| (format == outputDesc->mFormat) && |
| (channelMask == outputDesc->mChannelMask)) { |
| outputDesc->mDirectOpenCount++; |
| ALOGV("getOutput() reusing direct output %d", mOutputs.keyAt(i)); |
| return mOutputs.keyAt(i); |
| } |
| } |
| } |
| // close direct output if currently open and configured with different parameters |
| if (outputDesc != NULL) { |
| closeOutput(outputDesc->mIoHandle); |
| } |
| outputDesc = new AudioOutputDescriptor(profile); |
| 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->mModule->mHandle, |
| &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); |
| } |
| return AUDIO_IO_HANDLE_NONE; |
| } |
| outputDesc->mSamplingRate = config.sample_rate; |
| outputDesc->mChannelMask = config.channel_mask; |
| outputDesc->mFormat = config.format; |
| outputDesc->mRefCount[stream] = 0; |
| outputDesc->mStopTime[stream] = 0; |
| outputDesc->mDirectOpenCount = 1; |
| |
| audio_io_handle_t srcOutput = getOutputForEffect(); |
| addOutput(output, outputDesc); |
| audio_io_handle_t dstOutput = getOutputForEffect(); |
| if (dstOutput == output) { |
| mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, srcOutput, dstOutput); |
| } |
| mPreviousOutputs = mOutputs; |
| ALOGV("getOutput() returns new direct output %d", output); |
| mpClientInterface->onAudioPortListUpdate(); |
| return output; |
| } |
| |
| non_direct_output: |
| |
| // ignoring channel mask due to downmix capability in mixer |
| |
| // open a non direct output |
| |
| // for non direct outputs, only PCM is supported |
| if (audio_is_linear_pcm(format)) { |
| // get which output is suitable for the specified stream. The actual |
| // routing change will happen when startOutput() will be called |
| SortedVector<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("getOutput() returns output %d", output); |
| |
| return output; |
| } |
| |
| |
| status_t AudioPolicyManagerCustom::stopOutput(audio_io_handle_t output, |
| audio_stream_type_t stream, |
| int session) |
| { |
| ALOGV("stopOutput() output %d, stream %d, session %d", output, stream, session); |
| ssize_t index = mOutputs.indexOfKey(output); |
| if (index < 0) { |
| ALOGW("stopOutput() unknown output %d", output); |
| return BAD_VALUE; |
| } |
| |
| sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(index); |
| |
| // handle special case for sonification while in call |
| if ((isInCall()) && (outputDesc->mRefCount[stream] == 1)) { |
| handleIncallSonification(stream, false, false); |
| } |
| |
| if (outputDesc->mRefCount[stream] > 0) { |
| // decrement usage count of this stream on the output |
| outputDesc->changeRefCount(stream, -1); |
| // store time at which the stream was stopped - see isStreamActive() |
| if (outputDesc->mRefCount[stream] == 0) { |
| outputDesc->mStopTime[stream] = systemTime(); |
| audio_devices_t newDevice = getNewOutputDevice(output, false /*fromCache*/); |
| // delay the device switch by twice the latency because stopOutput() is executed when |
| // the track stop() command is received and at that time the audio track buffer can |
| // still contain data that needs to be drained. The latency only covers the audio HAL |
| // and kernel buffers. Also the latency does not always include additional delay in the |
| // audio path (audio DSP, CODEC ...) |
| setOutputDevice(output, newDevice, false, outputDesc->mLatency*2); |
| |
| // force restoring the device selection on other active outputs if it differs from the |
| // one being selected for this output |
| for (size_t i = 0; i < mOutputs.size(); i++) { |
| audio_io_handle_t curOutput = mOutputs.keyAt(i); |
| sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i); |
| if (curOutput != output && |
| desc->isActive() && |
| outputDesc->sharesHwModuleWith(desc) && |
| (newDevice != desc->device())) { |
| setOutputDevice(curOutput, |
| getNewOutputDevice(curOutput, false /*fromCache*/), |
| true, |
| outputDesc->mLatency*2); |
| } |
| } |
| // update the outputs if stopping one with a stream that can affect notification routing |
| handleNotificationRoutingForStream(stream); |
| } |
| return NO_ERROR; |
| } else { |
| ALOGW("stopOutput() refcount is already 0 for output %d", output); |
| return INVALID_OPERATION; |
| } |
| } |
| |
| audio_io_handle_t AudioPolicyManagerCustom::getInput(audio_source_t inputSource, |
| uint32_t samplingRate, |
| audio_format_t format, |
| audio_channel_mask_t channelMask, |
| audio_session_t session, |
| audio_input_flags_t flags) |
| { |
| ALOGV("getInput() inputSource %d, samplingRate %d, format %d, channelMask %x, session %d, " |
| "flags %#x", |
| inputSource, samplingRate, format, channelMask, session, flags); |
| |
| audio_devices_t device = getDeviceForInputSource(inputSource); |
| |
| if (device == AUDIO_DEVICE_NONE) { |
| ALOGW("getInput() could not find device for inputSource %d", inputSource); |
| return AUDIO_IO_HANDLE_NONE; |
| } |
| |
| /*The below code is intentionally not ported. |
| It's not needed to update the channel mask based on source because |
| the source is sent to audio HAL through set_parameters(). |
| For example, if source = VOICE_CALL, does not mean we need to capture two channels. |
| If the sound recorder app selects AMR as encoding format but source as RX+TX, |
| we need both in ONE channel. So we use the channels set by the app and use source |
| to tell the driver what needs to captured (RX only, TX only, or RX+TX ).*/ |
| // adapt channel selection to input source |
| /*switch (inputSource) { |
| case AUDIO_SOURCE_VOICE_UPLINK: |
| channelMask = AUDIO_CHANNEL_IN_VOICE_UPLINK; |
| break; |
| case AUDIO_SOURCE_VOICE_DOWNLINK: |
| channelMask = AUDIO_CHANNEL_IN_VOICE_DNLINK; |
| break; |
| case AUDIO_SOURCE_VOICE_CALL: |
| channelMask = AUDIO_CHANNEL_IN_VOICE_UPLINK | AUDIO_CHANNEL_IN_VOICE_DNLINK; |
| break; |
| default: |
| break; |
| }*/ |
| |
| #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 == mPhoneState) || |
| ((AUDIO_MODE_IN_CALL == mPrevPhoneState) && |
| (AUDIO_MODE_IN_COMMUNICATION == mPhoneState))) |
| { |
| switch(inputSource) { |
| case AUDIO_SOURCE_VOICE_UPLINK: |
| case AUDIO_SOURCE_VOICE_DOWNLINK: |
| case AUDIO_SOURCE_VOICE_CALL: |
| ALOGD("Creating input during incall mode for inputSource: %d ",inputSource); |
| break; |
| |
| case AUDIO_SOURCE_VOICE_COMMUNICATION: |
| if(prop_voip_enabled) { |
| ALOGD("BLOCKING VoIP request during incall mode for inputSource: %d ",inputSource); |
| return 0; |
| } |
| break; |
| default: |
| ALOGD("BLOCKING input during incall mode for inputSource: %d ",inputSource); |
| return 0; |
| } |
| } |
| }//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 == mPhoneState) || |
| ((AUDIO_MODE_IN_CALL == mPrevPhoneState) && |
| (AUDIO_MODE_IN_COMMUNICATION == mPhoneState))) |
| { |
| if(inputSource == AUDIO_SOURCE_VOICE_COMMUNICATION) { |
| ALOGD("BLOCKING VoIP request during incall mode for inputSource: %d ",inputSource); |
| return 0; |
| } |
| } |
| } |
| |
| #endif |
| |
| audio_io_handle_t input = AUDIO_IO_HANDLE_NONE; |
| bool isSoundTrigger = false; |
| audio_source_t halInputSource = inputSource; |
| if (inputSource == AUDIO_SOURCE_HOTWORD) { |
| ssize_t index = mSoundTriggerSessions.indexOfKey(session); |
| if (index >= 0) { |
| input = mSoundTriggerSessions.valueFor(session); |
| isSoundTrigger = true; |
| flags = (audio_input_flags_t)(flags | AUDIO_INPUT_FLAG_HW_HOTWORD); |
| ALOGV("SoundTrigger capture on session %d input %d", session, input); |
| } else { |
| halInputSource = AUDIO_SOURCE_VOICE_RECOGNITION; |
| } |
| } |
| |
| sp<IOProfile> profile = getInputProfile(device, |
| samplingRate, |
| format, |
| channelMask, |
| flags); |
| if (profile == 0) { |
| //retry without flags |
| audio_input_flags_t log_flags = flags; |
| flags = AUDIO_INPUT_FLAG_NONE; |
| profile = getInputProfile(device, |
| samplingRate, |
| format, |
| channelMask, |
| flags); |
| if (profile == 0) { |
| ALOGW("getInput() could not find profile for device 0x%X, samplingRate %u, format %#x, " |
| "channelMask 0x%X, flags %#x", |
| device, samplingRate, format, channelMask, log_flags); |
| return AUDIO_IO_HANDLE_NONE; |
| } |
| } |
| |
| if (profile->mModule->mHandle == 0) { |
| ALOGE("getInput(): HW module %s not opened", profile->mModule->mName); |
| return AUDIO_IO_HANDLE_NONE; |
| } |
| |
| audio_config_t config = AUDIO_CONFIG_INITIALIZER; |
| config.sample_rate = samplingRate; |
| config.channel_mask = channelMask; |
| config.format = format; |
| |
| status_t status = mpClientInterface->openInput(profile->mModule->mHandle, |
| &input, |
| &config, |
| &device, |
| String8(""), |
| halInputSource, |
| flags); |
| |
| // only accept input with the exact requested set of parameters |
| if (status != NO_ERROR || |
| (samplingRate != config.sample_rate) || |
| (format != config.format) || |
| (channelMask != config.channel_mask)) { |
| ALOGW("getInput() failed opening input: samplingRate %d, format %d, channelMask %x", |
| samplingRate, format, channelMask); |
| if (input != AUDIO_IO_HANDLE_NONE) { |
| mpClientInterface->closeInput(input); |
| } |
| return AUDIO_IO_HANDLE_NONE; |
| } |
| |
| sp<AudioInputDescriptor> inputDesc = new AudioInputDescriptor(profile); |
| inputDesc->mInputSource = inputSource; |
| inputDesc->mRefCount = 0; |
| inputDesc->mOpenRefCount = 1; |
| inputDesc->mSamplingRate = samplingRate; |
| inputDesc->mFormat = format; |
| inputDesc->mChannelMask = channelMask; |
| inputDesc->mDevice = device; |
| inputDesc->mSessions.add(session); |
| inputDesc->mIsSoundTrigger = isSoundTrigger; |
| |
| addInput(input, inputDesc); |
| mpClientInterface->onAudioPortListUpdate(); |
| return input; |
| } |
| |
| 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 (!isVirtualInputDevice(inputDesc->mDevice)) { |
| |
| // for a non-virtual input device, check if there is another (non-virtual) active input |
| audio_io_handle_t activeInput = 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; |
| } |
| } |
| } |
| |
| #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) &&(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) { |
| 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<AudioOutputDescriptor> 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) { |
| if (activeInputsCount() == 0) { |
| SoundTrigger::setCaptureState(true); |
| } |
| setInputDevice(input, getNewInputDevice(input), true /* force */); |
| |
| // Automatically enable the remote submix output when input is started. |
| // For remote submix (a virtual device), we open only one input per capture request. |
| if (audio_is_remote_submix_device(inputDesc->mDevice)) { |
| setDeviceConnectionState(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, |
| AUDIO_POLICY_DEVICE_STATE_AVAILABLE, AUDIO_REMOTE_SUBMIX_DEVICE_ADDRESS); |
| } |
| } |
| |
| ALOGV("AudioPolicyManagerCustom::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) |
| { |
| ALOGV("stopInput() input %d", input); |
| ssize_t index = mInputs.indexOfKey(input); |
| if (index < 0) { |
| ALOGW("stopInput() unknown input %d", input); |
| return BAD_VALUE; |
| } |
| sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(index); |
| |
| index = inputDesc->mSessions.indexOf(session); |
| if (index < 0) { |
| ALOGW("stopInput() unknown session %d on input %d", session, input); |
| return BAD_VALUE; |
| } |
| |
| if (inputDesc->mRefCount == 0) { |
| ALOGW("stopInput() input %d already stopped", input); |
| return INVALID_OPERATION; |
| } |
| |
| inputDesc->mRefCount--; |
| if (inputDesc->mRefCount == 0) { |
| |
| // automatically disable the remote submix output when input is stopped |
| if (audio_is_remote_submix_device(inputDesc->mDevice)) { |
| setDeviceConnectionState(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, |
| AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE, AUDIO_REMOTE_SUBMIX_DEVICE_ADDRESS); |
| } |
| |
| resetInputDevice(input); |
| |
| if (activeInputsCount() == 0) { |
| SoundTrigger::setCaptureState(false); |
| } |
| } |
| |
| #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) && (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) { |
| ALOGD(" Invalidate on stopInput for stream :: %d ", i); |
| //FIXME see fixme on name change |
| mpClientInterface->invalidateStream((audio_stream_type_t)i); |
| } |
| } |
| } |
| #endif |
| return NO_ERROR; |
| } |
| |
| status_t AudioPolicyManagerCustom::setStreamVolumeIndex(audio_stream_type_t stream, |
| int index, |
| audio_devices_t device) |
| { |
| |
| if ((index < mStreams[stream].mIndexMin) || (index > mStreams[stream].mIndexMax)) { |
| return BAD_VALUE; |
| } |
| if (!audio_is_output_device(device)) { |
| return BAD_VALUE; |
| } |
| |
| // Force max volume if stream cannot be muted |
| if (!mStreams[stream].mCanBeMuted) index = mStreams[stream].mIndexMax; |
| |
| ALOGV("setStreamVolumeIndex() stream %d, device %04x, index %d", |
| stream, device, index); |
| |
| // if device is AUDIO_DEVICE_OUT_DEFAULT set default value and |
| // clear all device specific values |
| if (device == AUDIO_DEVICE_OUT_DEFAULT) { |
| mStreams[stream].mIndexCur.clear(); |
| } |
| mStreams[stream].mIndexCur.add(device, index); |
| |
| // compute and apply stream volume on all outputs according to connected device |
| status_t status = NO_ERROR; |
| for (size_t i = 0; i < mOutputs.size(); i++) { |
| audio_devices_t curDevice = |
| getDeviceForVolume(mOutputs.valueAt(i)->device()); |
| #ifdef AUDIO_EXTN_FM_ENABLED |
| audio_devices_t availableOutputDeviceTypes = mAvailableOutputDevices.types(); |
| if (((device == AUDIO_DEVICE_OUT_DEFAULT) && |
| ((availableOutputDeviceTypes & AUDIO_DEVICE_OUT_FM) != AUDIO_DEVICE_OUT_FM)) || |
| (device == curDevice)) { |
| #else |
| if ((device == AUDIO_DEVICE_OUT_DEFAULT) || (device == curDevice)) { |
| #endif |
| status_t volStatus = checkAndSetVolume(stream, index, mOutputs.keyAt(i), curDevice); |
| if (volStatus != NO_ERROR) { |
| status = volStatus; |
| } |
| } |
| } |
| return status; |
| } |
| |
| // 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) |
| { |
| ALOGD("copl: isOffloadSupported: SR=%u, CM=0x%x, Format=0x%x, StreamType=%d," |
| " BitRate=%u, duration=%lld 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) || (activeInputsCount() > 0))) { |
| ALOGD("copl: blocking compress offload for record concurrency"); |
| return false; |
| } |
| #endif |
| // Check if stream type is music, then only allow offload as of now. |
| if (offloadInfo.stream_type != AUDIO_STREAM_MUSIC) |
| { |
| ALOGD("isOffloadSupported: stream_type != MUSIC, returning false"); |
| return false; |
| } |
| |
| char propValue[PROPERTY_VALUE_MAX]; |
| bool pcmOffload = false; |
| #ifdef PCM_OFFLOAD_ENABLED |
| if (audio_is_offload_pcm(offloadInfo.format)) { |
| if(property_get("audio.offload.pcm.enable", propValue, NULL)) { |
| bool prop_enabled = atoi(propValue) || !strncmp("true", propValue, 4); |
| if (prop_enabled) { |
| ALOGW("PCM offload property is enabled"); |
| pcmOffload = true; |
| } |
| } |
| if (!pcmOffload) { |
| ALOGD("PCM offload disabled by property audio.offload.pcm.enable"); |
| return false; |
| } |
| } |
| #endif |
| |
| if (!pcmOffload) { |
| // Check if offload has been disabled |
| if (property_get("audio.offload.disable", propValue, "0")) { |
| if (atoi(propValue) != 0) { |
| ALOGD("offload disabled by audio.offload.disable=%s", propValue ); |
| return false; |
| } |
| } |
| |
| //check if it's multi-channel AAC (includes sub formats), 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 and FLAC format"); |
| return false; |
| } |
| |
| if (offloadInfo.has_video) |
| { |
| if(property_get("av.offload.enable", propValue, NULL)) { |
| bool prop_enabled = atoi(propValue) || !strncmp("true", propValue, 4); |
| if (!prop_enabled) { |
| ALOGW("offload disabled by av.offload.enable = %s ", propValue ); |
| return false; |
| } |
| } else { |
| return false; |
| } |
| |
| if(offloadInfo.is_streaming) { |
| if (property_get("av.streaming.offload.enable", propValue, NULL)) { |
| bool prop_enabled = atoi(propValue) || !strncmp("true", propValue, 4); |
| if (!prop_enabled) { |
| ALOGW("offload disabled by av.streaming.offload.enable = %s ", propValue ); |
| return false; |
| } |
| } else { |
| //Do not offload AV streamnig if the property is not defined |
| return false; |
| } |
| } |
| ALOGD("copl: isOffloadSupported: has_video == true, property\ |
| set to enable offload"); |
| } |
| } |
| |
| //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 )) { |
| ALOGD("copl: Offload denied by duration < audio.offload.min.duration.secs(=%s)", propValue); |
| return false; |
| } |
| } else if (offloadInfo.duration_us < OFFLOAD_DEFAULT_MIN_DURATION_SECS * 1000000) { |
| ALOGD("copl: 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_FLAC) || |
| ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_VORBIS) || |
| ((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) || |
| pcmOffload) |
| return false; |
| } |
| |
| // Do not allow offloading if one non offloadable effect is enabled. This prevents from |
| // creating an offloaded track and tearing it down immediately after start when audioflinger |
| // detects there is an active non offloadable effect. |
| // FIXME: We should check the audio session here but we do not have it in this context. |
| // This may prevent offloading in rare situations where effects are left active by apps |
| // in the background. |
| if (isNonOffloadableEffectEnabled()) { |
| 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); |
| ALOGD("copl: isOffloadSupported() profile %sfound", profile != 0 ? "" : "NOT "); |
| return (profile != 0); |
| } |
| |
| uint32_t AudioPolicyManagerCustom::nextUniqueId() |
| { |
| return android_atomic_inc(&mNextUniqueId); |
| } |
| |
| AudioPolicyManagerCustom::routing_strategy AudioPolicyManagerCustom::getStrategy( |
| audio_stream_type_t stream) { |
| // stream to strategy mapping |
| switch (stream) { |
| case AUDIO_STREAM_VOICE_CALL: |
| case AUDIO_STREAM_BLUETOOTH_SCO: |
| return STRATEGY_PHONE; |
| case AUDIO_STREAM_RING: |
| case AUDIO_STREAM_ALARM: |
| return STRATEGY_SONIFICATION; |
| case AUDIO_STREAM_NOTIFICATION: |
| return STRATEGY_SONIFICATION_RESPECTFUL; |
| case AUDIO_STREAM_DTMF: |
| return STRATEGY_DTMF; |
| default: |
| ALOGE("unknown stream type"); |
| case AUDIO_STREAM_SYSTEM: |
| // NOTE: SYSTEM stream uses MEDIA strategy because muting music and switching outputs |
| // while key clicks are played produces a poor result |
| case AUDIO_STREAM_TTS: |
| case AUDIO_STREAM_MUSIC: |
| #ifdef AUDIO_EXTN_INCALL_MUSIC_ENABLED |
| case AUDIO_STREAM_INCALL_MUSIC: |
| #endif |
| return STRATEGY_MEDIA; |
| case AUDIO_STREAM_ENFORCED_AUDIBLE: |
| return STRATEGY_ENFORCED_AUDIBLE; |
| } |
| } |
| |
| void AudioPolicyManagerCustom::handleNotificationRoutingForStream(audio_stream_type_t stream) { |
| switch(stream) { |
| case AUDIO_STREAM_MUSIC: |
| checkOutputForStrategy(STRATEGY_SONIFICATION_RESPECTFUL); |
| updateDevicesAndOutputs(); |
| break; |
| default: |
| break; |
| } |
| } |
| |
| audio_devices_t AudioPolicyManagerCustom::getDeviceForStrategy(routing_strategy strategy, |
| bool fromCache) |
| { |
| uint32_t device = AUDIO_DEVICE_NONE; |
| |
| if (fromCache) { |
| ALOGVV("getDeviceForStrategy() from cache strategy %d, device %x", |
| strategy, mDeviceForStrategy[strategy]); |
| return mDeviceForStrategy[strategy]; |
| } |
| audio_devices_t availableOutputDeviceTypes = mAvailableOutputDevices.types(); |
| switch (strategy) { |
| |
| case STRATEGY_SONIFICATION_RESPECTFUL: |
| if (isInCall()) { |
| device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/); |
| } else if (isStreamActiveRemotely(AUDIO_STREAM_MUSIC, |
| SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY)) { |
| // while media is playing on a remote device, use the the sonification behavior. |
| // Note that we test this usecase before testing if media is playing because |
| // the isStreamActive() method only informs about the activity of a stream, not |
| // if it's for local playback. Note also that we use the same delay between both tests |
| device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/); |
| //user "safe" speaker if available instead of normal speaker to avoid triggering |
| //other acoustic safety mechanisms for notification |
| if (device == AUDIO_DEVICE_OUT_SPEAKER && (availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER_SAFE)) |
| device = AUDIO_DEVICE_OUT_SPEAKER_SAFE; |
| } else if (isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY)) { |
| // while media is playing (or has recently played), use the same device |
| device = getDeviceForStrategy(STRATEGY_MEDIA, false /*fromCache*/); |
| } else { |
| // when media is not playing anymore, fall back on the sonification behavior |
| device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/); |
| //user "safe" speaker if available instead of normal speaker to avoid triggering |
| //other acoustic safety mechanisms for notification |
| if (device == AUDIO_DEVICE_OUT_SPEAKER && (availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER_SAFE)) |
| device = AUDIO_DEVICE_OUT_SPEAKER_SAFE; |
| } |
| |
| break; |
| |
| case STRATEGY_DTMF: |
| if (!isInCall()) { |
| // when off call, DTMF strategy follows the same rules as MEDIA strategy |
| device = getDeviceForStrategy(STRATEGY_MEDIA, false /*fromCache*/); |
| break; |
| } |
| // when in call, DTMF and PHONE strategies follow the same rules |
| // FALL THROUGH |
| |
| case STRATEGY_PHONE: |
| // Force use of only devices on primary output if: |
| // - in call AND |
| // - cannot route from voice call RX OR |
| // - audio HAL version is < 3.0 and TX device is on the primary HW module |
| if (mPhoneState == AUDIO_MODE_IN_CALL) { |
| audio_devices_t txDevice = getDeviceForInputSource(AUDIO_SOURCE_VOICE_COMMUNICATION); |
| sp<AudioOutputDescriptor> hwOutputDesc = mOutputs.valueFor(mPrimaryOutput); |
| if (((mAvailableInputDevices.types() & |
| AUDIO_DEVICE_IN_TELEPHONY_RX & ~AUDIO_DEVICE_BIT_IN) == 0) || |
| (((txDevice & availablePrimaryInputDevices() & ~AUDIO_DEVICE_BIT_IN) != 0) && |
| (hwOutputDesc->getAudioPort()->mModule->mHalVersion < |
| AUDIO_DEVICE_API_VERSION_3_0))) { |
| availableOutputDeviceTypes = availablePrimaryOutputDevices(); |
| } |
| } |
| // for phone strategy, we first consider the forced use and then the available devices by order |
| // of priority |
| switch (mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION]) { |
| case AUDIO_POLICY_FORCE_BT_SCO: |
| if (!isInCall() || strategy != STRATEGY_DTMF) { |
| device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT; |
| if (device) break; |
| } |
| device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET; |
| if (device) break; |
| device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_SCO; |
| if (device) break; |
| // if SCO device is requested but no SCO device is available, fall back to default case |
| // FALL THROUGH |
| |
| default: // FORCE_NONE |
| // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to A2DP |
| if (!isInCall() && |
| (mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] != AUDIO_POLICY_FORCE_NO_BT_A2DP) && |
| (getA2dpOutput() != 0) && !mA2dpSuspended) { |
| device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP; |
| if (device) break; |
| device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES; |
| if (device) break; |
| } |
| device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_WIRED_HEADPHONE; |
| if (device) break; |
| device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_WIRED_HEADSET; |
| if (device) break; |
| device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_DEVICE; |
| if (device) break; |
| if (mPhoneState != AUDIO_MODE_IN_CALL) { |
| device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_ACCESSORY; |
| if (device) break; |
| device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET; |
| if (device) break; |
| device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_AUX_DIGITAL; |
| if (device) break; |
| } |
| |
| // Allow voice call on USB ANLG DOCK headset |
| device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET; |
| if (device) break; |
| |
| device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_EARPIECE; |
| if (device) break; |
| device = mDefaultOutputDevice->mDeviceType; |
| if (device == AUDIO_DEVICE_NONE) { |
| ALOGE("getDeviceForStrategy() no device found for STRATEGY_PHONE"); |
| } |
| break; |
| |
| case AUDIO_POLICY_FORCE_SPEAKER: |
| // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to |
| // A2DP speaker when forcing to speaker output |
| if (!isInCall() && |
| (mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] != AUDIO_POLICY_FORCE_NO_BT_A2DP) && |
| (getA2dpOutput() != 0) && !mA2dpSuspended) { |
| device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER; |
| if (device) break; |
| } |
| if (mPhoneState != AUDIO_MODE_IN_CALL) { |
| device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_ACCESSORY; |
| if (device) break; |
| device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_DEVICE; |
| if (device) break; |
| device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET; |
| if (device) break; |
| device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_AUX_DIGITAL; |
| if (device) break; |
| device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET; |
| if (device) break; |
| } |
| device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_LINE; |
| if (device) break; |
| device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER; |
| if (device) break; |
| device = mDefaultOutputDevice->mDeviceType; |
| if (device == AUDIO_DEVICE_NONE) { |
| ALOGE("getDeviceForStrategy() no device found for STRATEGY_PHONE, FORCE_SPEAKER"); |
| } |
| break; |
| } |
| |
| if (isInCall() && (device == AUDIO_DEVICE_NONE)) { |
| // when in call, get the device for Phone strategy |
| device = getDeviceForStrategy(STRATEGY_PHONE, false /*fromCache*/); |
| break; |
| } |
| |
| #ifdef AUDIO_EXTN_FM_ENABLED |
| if (availableOutputDeviceTypes & AUDIO_DEVICE_OUT_FM) { |
| if (mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] == AUDIO_POLICY_FORCE_SPEAKER) { |
| device = AUDIO_DEVICE_OUT_SPEAKER; |
| } |
| } |
| #endif |
| break; |
| |
| case STRATEGY_SONIFICATION: |
| |
| // If incall, just select the STRATEGY_PHONE device: The rest of the behavior is handled by |
| // handleIncallSonification(). |
| if (isInCall()) { |
| device = getDeviceForStrategy(STRATEGY_PHONE, false /*fromCache*/); |
| break; |
| } |
| // FALL THROUGH |
| |
| case STRATEGY_ENFORCED_AUDIBLE: |
| // strategy STRATEGY_ENFORCED_AUDIBLE uses same routing policy as STRATEGY_SONIFICATION |
| // except: |
| // - when in call where it doesn't default to STRATEGY_PHONE behavior |
| // - in countries where not enforced in which case it follows STRATEGY_MEDIA |
| |
| if ((strategy == STRATEGY_SONIFICATION) || |
| (mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED)) { |
| device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER; |
| if (device == AUDIO_DEVICE_NONE) { |
| ALOGE("getDeviceForStrategy() speaker device not found for STRATEGY_SONIFICATION"); |
| } |
| } |
| // The second device used for sonification is the same as the device used by media strategy |
| // FALL THROUGH |
| |
| case STRATEGY_MEDIA: { |
| uint32_t device2 = AUDIO_DEVICE_NONE; |
| |
| if (isInCall() && (device == AUDIO_DEVICE_NONE)) { |
| // when in call, get the device for Phone strategy |
| device = getDeviceForStrategy(STRATEGY_PHONE, false /*fromCache*/); |
| break; |
| } |
| #ifdef AUDIO_EXTN_FM_ENABLED |
| if (mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] == AUDIO_POLICY_FORCE_SPEAKER) { |
| device = AUDIO_DEVICE_OUT_SPEAKER; |
| break; |
| } |
| #endif |
| |
| if (strategy != STRATEGY_SONIFICATION) { |
| // no sonification on remote submix (e.g. WFD) |
| device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_REMOTE_SUBMIX; |
| } |
| if ((device2 == AUDIO_DEVICE_NONE) && |
| (mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] != AUDIO_POLICY_FORCE_NO_BT_A2DP) && |
| (getA2dpOutput() != 0) && !mA2dpSuspended) { |
| device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP; |
| if (device2 == AUDIO_DEVICE_NONE) { |
| device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES; |
| } |
| if (device2 == AUDIO_DEVICE_NONE) { |
| device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER; |
| } |
| } |
| if (device2 == AUDIO_DEVICE_NONE) { |
| device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_WIRED_HEADPHONE; |
| } |
| if ((device2 == AUDIO_DEVICE_NONE)) { |
| device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_LINE; |
| } |
| if (device2 == AUDIO_DEVICE_NONE) { |
| device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_WIRED_HEADSET; |
| } |
| if (device2 == AUDIO_DEVICE_NONE) { |
| device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_ACCESSORY; |
| } |
| if (device2 == AUDIO_DEVICE_NONE) { |
| device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_DEVICE; |
| } |
| if (device2 == AUDIO_DEVICE_NONE) { |
| device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET; |
| } |
| if ((strategy != STRATEGY_SONIFICATION) && (device == AUDIO_DEVICE_NONE) |
| && (device2 == AUDIO_DEVICE_NONE)) { |
| // no sonification on aux digital (e.g. HDMI) |
| device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_AUX_DIGITAL; |
| } |
| if ((device2 == AUDIO_DEVICE_NONE) && |
| (mForceUse[AUDIO_POLICY_FORCE_FOR_DOCK] == AUDIO_POLICY_FORCE_ANALOG_DOCK) |
| && (strategy != STRATEGY_SONIFICATION)) { |
| device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET; |
| } |
| #ifdef AUDIO_EXTN_FM_ENABLED |
| if ((strategy != STRATEGY_SONIFICATION) && (device == AUDIO_DEVICE_NONE) |
| && (device2 == AUDIO_DEVICE_NONE)) { |
| device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_FM_TX; |
| } |
| #endif |
| #ifdef AUDIO_EXTN_AFE_PROXY_ENABLED |
| if ((strategy != STRATEGY_SONIFICATION) && (device == AUDIO_DEVICE_NONE) |
| && (device2 == AUDIO_DEVICE_NONE)) { |
| // no sonification on WFD sink |
| device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_PROXY; |
| } |
| #endif |
| if (device2 == AUDIO_DEVICE_NONE) { |
| device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER; |
| } |
| int device3 = AUDIO_DEVICE_NONE; |
| if (strategy == STRATEGY_MEDIA) { |
| // ARC, SPDIF and AUX_LINE can co-exist with others. |
| device3 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_HDMI_ARC; |
| device3 |= (availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPDIF); |
| device3 |= (availableOutputDeviceTypes & AUDIO_DEVICE_OUT_AUX_LINE); |
| } |
| |
| device2 |= device3; |
| // device is DEVICE_OUT_SPEAKER if we come from case STRATEGY_SONIFICATION or |
| // STRATEGY_ENFORCED_AUDIBLE, AUDIO_DEVICE_NONE otherwise |
| device |= device2; |
| |
| // If hdmi system audio mode is on, remove speaker out of output list. |
| if ((strategy == STRATEGY_MEDIA) && |
| (mForceUse[AUDIO_POLICY_FORCE_FOR_HDMI_SYSTEM_AUDIO] == |
| AUDIO_POLICY_FORCE_HDMI_SYSTEM_AUDIO_ENFORCED)) { |
| device &= ~AUDIO_DEVICE_OUT_SPEAKER; |
| } |
| |
| if (device) break; |
| device = mDefaultOutputDevice->mDeviceType; |
| if (device == AUDIO_DEVICE_NONE) { |
| ALOGE("getDeviceForStrategy() no device found for STRATEGY_MEDIA"); |
| } |
| } break; |
| |
| default: |
| ALOGW("getDeviceForStrategy() unknown strategy: %d", strategy); |
| break; |
| } |
| |
| ALOGVV("getDeviceForStrategy() strategy %d, device %x", strategy, device); |
| return device; |
| } |
| |
| audio_devices_t AudioPolicyManagerCustom::getDeviceForInputSource(audio_source_t inputSource) |
| { |
| uint32_t device = AUDIO_DEVICE_NONE; |
| audio_devices_t availableDeviceTypes = mAvailableInputDevices.types() & |
| ~AUDIO_DEVICE_BIT_IN; |
| switch (inputSource) { |
| case AUDIO_SOURCE_VOICE_UPLINK: |
| if (availableDeviceTypes & AUDIO_DEVICE_IN_VOICE_CALL) { |
| device = AUDIO_DEVICE_IN_VOICE_CALL; |
| break; |
| } |
| break; |
| |
| case AUDIO_SOURCE_DEFAULT: |
| case AUDIO_SOURCE_MIC: |
| if (availableDeviceTypes & AUDIO_DEVICE_IN_BLUETOOTH_A2DP) { |
| device = AUDIO_DEVICE_IN_BLUETOOTH_A2DP; |
| } else if (availableDeviceTypes & AUDIO_DEVICE_IN_WIRED_HEADSET) { |
| device = AUDIO_DEVICE_IN_WIRED_HEADSET; |
| } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_DEVICE) { |
| device = AUDIO_DEVICE_IN_USB_DEVICE; |
| } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) { |
| device = AUDIO_DEVICE_IN_BUILTIN_MIC; |
| } |
| break; |
| |
| case AUDIO_SOURCE_VOICE_COMMUNICATION: |
| // Allow only use of devices on primary input if in call and HAL does not support routing |
| // to voice call path. |
| if ((mPhoneState == AUDIO_MODE_IN_CALL) && |
| (mAvailableOutputDevices.types() & AUDIO_DEVICE_OUT_TELEPHONY_TX) == 0) { |
| availableDeviceTypes = availablePrimaryInputDevices() & ~AUDIO_DEVICE_BIT_IN; |
| } |
| |
| switch (mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION]) { |
| case AUDIO_POLICY_FORCE_BT_SCO: |
| // if SCO device is requested but no SCO device is available, fall back to default case |
| if (availableDeviceTypes & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) { |
| device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET; |
| break; |
| } |
| // FALL THROUGH |
| |
| default: // FORCE_NONE |
| if (availableDeviceTypes & AUDIO_DEVICE_IN_WIRED_HEADSET) { |
| device = AUDIO_DEVICE_IN_WIRED_HEADSET; |
| } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_DEVICE) { |
| device = AUDIO_DEVICE_IN_USB_DEVICE; |
| } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) { |
| device = AUDIO_DEVICE_IN_BUILTIN_MIC; |
| } |
| break; |
| |
| case AUDIO_POLICY_FORCE_SPEAKER: |
| if (availableDeviceTypes & AUDIO_DEVICE_IN_BACK_MIC) { |
| device = AUDIO_DEVICE_IN_BACK_MIC; |
| } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) { |
| device = AUDIO_DEVICE_IN_BUILTIN_MIC; |
| } |
| break; |
| } |
| break; |
| |
| case AUDIO_SOURCE_VOICE_RECOGNITION: |
| case AUDIO_SOURCE_HOTWORD: |
| if (mForceUse[AUDIO_POLICY_FORCE_FOR_RECORD] == AUDIO_POLICY_FORCE_BT_SCO && |
| availableDeviceTypes & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) { |
| device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET; |
| } else if (availableDeviceTypes & AUDIO_DEVICE_IN_WIRED_HEADSET) { |
| device = AUDIO_DEVICE_IN_WIRED_HEADSET; |
| } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_DEVICE) { |
| device = AUDIO_DEVICE_IN_USB_DEVICE; |
| } else if (availableDeviceTypes & AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET) { |
| device = AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET; |
| } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) { |
| device = AUDIO_DEVICE_IN_BUILTIN_MIC; |
| } |
| break; |
| case AUDIO_SOURCE_CAMCORDER: |
| if (availableDeviceTypes & AUDIO_DEVICE_IN_BACK_MIC) { |
| device = AUDIO_DEVICE_IN_BACK_MIC; |
| } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) { |
| device = AUDIO_DEVICE_IN_BUILTIN_MIC; |
| } |
| break; |
| case AUDIO_SOURCE_VOICE_DOWNLINK: |
| case AUDIO_SOURCE_VOICE_CALL: |
| if (availableDeviceTypes & AUDIO_DEVICE_IN_VOICE_CALL) { |
| device = AUDIO_DEVICE_IN_VOICE_CALL; |
| } |
| break; |
| case AUDIO_SOURCE_REMOTE_SUBMIX: |
| if (availableDeviceTypes & AUDIO_DEVICE_IN_REMOTE_SUBMIX) { |
| device = AUDIO_DEVICE_IN_REMOTE_SUBMIX; |
| } |
| break; |
| #ifdef AUDIO_EXTN_FM_ENABLED |
| case AUDIO_SOURCE_FM_RX: |
| device = AUDIO_DEVICE_IN_FM_RX; |
| break; |
| case AUDIO_SOURCE_FM_RX_A2DP: |
| device = AUDIO_DEVICE_IN_FM_RX_A2DP; |
| break; |
| #endif |
| default: |
| ALOGW("getDeviceForInputSource() invalid input source %d", inputSource); |
| break; |
| } |
| ALOGV("getDeviceForInputSource()input source %d, device %08x", inputSource, device); |
| return device; |
| } |
| |
| bool AudioPolicyManagerCustom::isVirtualInputDevice(audio_devices_t device) |
| { |
| if ((device & AUDIO_DEVICE_BIT_IN) != 0) { |
| device &= ~AUDIO_DEVICE_BIT_IN; |
| if ((popcount(device) == 1) && ((device & ~APM_AUDIO_IN_DEVICE_VIRTUAL_ALL) == 0)) |
| return true; |
| } |
| return false; |
| } |
| |
| bool AudioPolicyManagerCustom::deviceDistinguishesOnAddress(audio_devices_t device) { |
| return ((device & APM_AUDIO_DEVICE_MATCH_ADDRESS_ALL) != 0); |
| } |
| |
| AudioPolicyManagerCustom::device_category AudioPolicyManagerCustom::getDeviceCategory(audio_devices_t device) |
| { |
| switch(getDeviceForVolume(device)) { |
| case AUDIO_DEVICE_OUT_EARPIECE: |
| return DEVICE_CATEGORY_EARPIECE; |
| case AUDIO_DEVICE_OUT_WIRED_HEADSET: |
| case AUDIO_DEVICE_OUT_WIRED_HEADPHONE: |
| case AUDIO_DEVICE_OUT_BLUETOOTH_SCO: |
| case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET: |
| case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP: |
| case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES: |
| #ifdef AUDIO_EXTN_FM_ENABLED |
| case AUDIO_DEVICE_OUT_FM: |
| #endif |
| return DEVICE_CATEGORY_HEADSET; |
| case AUDIO_DEVICE_OUT_LINE: |
| case AUDIO_DEVICE_OUT_AUX_DIGITAL: |
| /*USB? Remote submix?*/ |
| return DEVICE_CATEGORY_EXT_MEDIA; |
| case AUDIO_DEVICE_OUT_SPEAKER: |
| case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT: |
| case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER: |
| case AUDIO_DEVICE_OUT_USB_ACCESSORY: |
| case AUDIO_DEVICE_OUT_USB_DEVICE: |
| case AUDIO_DEVICE_OUT_REMOTE_SUBMIX: |
| #ifdef AUDIO_EXTN_AFE_PROXY_ENABLED |
| case AUDIO_DEVICE_OUT_PROXY: |
| #endif |
| default: |
| return DEVICE_CATEGORY_SPEAKER; |
| } |
| } |
| |
| float AudioPolicyManagerCustom::volIndexToAmpl(audio_devices_t device, const StreamDescriptor& streamDesc, |
| int indexInUi) |
| { |
| device_category deviceCategory = getDeviceCategory(device); |
| const VolumeCurvePoint *curve = streamDesc.mVolumeCurve[deviceCategory]; |
| |
| // the volume index in the UI is relative to the min and max volume indices for this stream type |
| int nbSteps = 1 + curve[VOLMAX].mIndex - |
| curve[VOLMIN].mIndex; |
| int volIdx = (nbSteps * (indexInUi - streamDesc.mIndexMin)) / |
| (streamDesc.mIndexMax - streamDesc.mIndexMin); |
| |
| // find what part of the curve this index volume belongs to, or if it's out of bounds |
| int segment = 0; |
| if (volIdx < curve[VOLMIN].mIndex) { // out of bounds |
| return 0.0f; |
| } else if (volIdx < curve[VOLKNEE1].mIndex) { |
| segment = 0; |
| } else if (volIdx < curve[VOLKNEE2].mIndex) { |
| segment = 1; |
| } else if (volIdx <= curve[VOLMAX].mIndex) { |
| segment = 2; |
| } else { // out of bounds |
| return 1.0f; |
| } |
| |
| // linear interpolation in the attenuation table in dB |
| float decibels = curve[segment].mDBAttenuation + |
| ((float)(volIdx - curve[segment].mIndex)) * |
| ( (curve[segment+1].mDBAttenuation - |
| curve[segment].mDBAttenuation) / |
| ((float)(curve[segment+1].mIndex - |
| curve[segment].mIndex)) ); |
| |
| float amplification = exp( decibels * 0.115129f); // exp( dB * ln(10) / 20 ) |
| |
| ALOGVV("VOLUME vol index=[%d %d %d], dB=[%.1f %.1f %.1f] ampl=%.5f", |
| curve[segment].mIndex, volIdx, |
| curve[segment+1].mIndex, |
| curve[segment].mDBAttenuation, |
| decibels, |
| curve[segment+1].mDBAttenuation, |
| amplification); |
| |
| return amplification; |
| } |
| |
| float AudioPolicyManagerCustom::computeVolume(audio_stream_type_t stream, |
| int index, |
| audio_io_handle_t output, |
| audio_devices_t device) |
| { |
| float volume = 1.0; |
| sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output); |
| StreamDescriptor &streamDesc = mStreams[stream]; |
| |
| if (device == AUDIO_DEVICE_NONE) { |
| device = outputDesc->device(); |
| } |
| |
| // if volume is not 0 (not muted), force media volume to max on digital output |
| if (stream == AUDIO_STREAM_MUSIC && |
| index != mStreams[stream].mIndexMin && |
| (device == AUDIO_DEVICE_OUT_AUX_DIGITAL || |
| #ifdef AUDIO_EXTN_AFE_PROXY_ENABLED |
| device == AUDIO_DEVICE_OUT_PROXY || |
| #endif |
| device == AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET)) { |
| return 1.0; |
| } |
| |
| #ifdef AUDIO_EXTN_INCALL_MUSIC_ENABLED |
| if (stream == AUDIO_STREAM_INCALL_MUSIC) { |
| return 1.0; |
| } |
| #endif |
| |
| volume = volIndexToAmpl(device, streamDesc, index); |
| |
| // if a headset is connected, apply the following rules to ring tones and notifications |
| // to avoid sound level bursts in user's ears: |
| // - always attenuate ring tones and notifications volume by 6dB |
| // - if music is playing, always limit the volume to current music volume, |
| // with a minimum threshold at -36dB so that notification is always perceived. |
| const routing_strategy stream_strategy = getStrategy(stream); |
| if ((device & (AUDIO_DEVICE_OUT_BLUETOOTH_A2DP | |
| AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES | |
| AUDIO_DEVICE_OUT_WIRED_HEADSET | |
| AUDIO_DEVICE_OUT_WIRED_HEADPHONE)) && |
| ((stream_strategy == STRATEGY_SONIFICATION) |
| || (stream_strategy == STRATEGY_SONIFICATION_RESPECTFUL) |
| || (stream == AUDIO_STREAM_SYSTEM) |
| || ((stream_strategy == STRATEGY_ENFORCED_AUDIBLE) && |
| (mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] == AUDIO_POLICY_FORCE_NONE))) && |
| streamDesc.mCanBeMuted) { |
| volume *= SONIFICATION_HEADSET_VOLUME_FACTOR; |
| // when the phone is ringing we must consider that music could have been paused just before |
| // by the music application and behave as if music was active if the last music track was |
| // just stopped |
| if (isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY) || |
| mLimitRingtoneVolume) { |
| audio_devices_t musicDevice = getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/); |
| float musicVol = computeVolume(AUDIO_STREAM_MUSIC, |
| mStreams[AUDIO_STREAM_MUSIC].getVolumeIndex(musicDevice), |
| output, |
| musicDevice); |
| float minVol = (musicVol > SONIFICATION_HEADSET_VOLUME_MIN) ? |
| musicVol : SONIFICATION_HEADSET_VOLUME_MIN; |
| if (volume > minVol) { |
| volume = minVol; |
| ALOGV("computeVolume limiting volume to %f musicVol %f", minVol, musicVol); |
| } |
| } |
| } |
| |
| return volume; |
| } |
| |
| status_t AudioPolicyManagerCustom::checkAndSetVolume(audio_stream_type_t stream, |
| int index, |
| audio_io_handle_t output, |
| audio_devices_t device, |
| int delayMs, |
| bool force) |
| { |
| |
| // do not change actual stream volume if the stream is muted |
| if (mOutputs.valueFor(output)->mMuteCount[stream] != 0) { |
| ALOGVV("checkAndSetVolume() stream %d muted count %d", |
| stream, mOutputs.valueFor(output)->mMuteCount[stream]); |
| return NO_ERROR; |
| } |
| |
| // do not change in call volume if bluetooth is connected and vice versa |
| if ((stream == AUDIO_STREAM_VOICE_CALL && |
| mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION] == AUDIO_POLICY_FORCE_BT_SCO) || |
| (stream == AUDIO_STREAM_BLUETOOTH_SCO && |
| mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION] != AUDIO_POLICY_FORCE_BT_SCO)) { |
| ALOGV("checkAndSetVolume() cannot set stream %d volume with force use = %d for comm", |
| stream, mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION]); |
| return INVALID_OPERATION; |
| } |
| |
| float volume = computeVolume(stream, index, output, device); |
| // We actually change the volume if: |
| // - the float value returned by computeVolume() changed |
| // - the force flag is set |
| if (volume != mOutputs.valueFor(output)->mCurVolume[stream] || |
| force) { |
| mOutputs.valueFor(output)->mCurVolume[stream] = volume; |
| ALOGVV("checkAndSetVolume() for output %d stream %d, volume %f, delay %d", output, stream, volume, delayMs); |
| // Force VOICE_CALL to track BLUETOOTH_SCO stream volume when bluetooth audio is |
| // enabled |
| if (stream == AUDIO_STREAM_BLUETOOTH_SCO) { |
| mpClientInterface->setStreamVolume(AUDIO_STREAM_VOICE_CALL, volume, output, delayMs); |
| #ifdef AUDIO_EXTN_FM_ENABLED |
| } else if (stream == AUDIO_STREAM_MUSIC && |
| output == mPrimaryOutput) { |
| if (volume >= 0) { |
| AudioParameter param = AudioParameter(); |
| param.addFloat(String8("fm_volume"), volume); |
| ALOGV("checkAndSetVolume setParameters volume, volume=:%f delay=:%d",volume,delayMs*2); |
| //Double delayMs to avoid sound burst while device switch. |
| mpClientInterface->setParameters(mPrimaryOutput, param.toString(), delayMs*2); |
| } |
| #endif |
| } |
| mpClientInterface->setStreamVolume(stream, volume, output, delayMs); |
| } |
| |
| 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[stream].mIndexMax; |
| } else { |
| voiceVolume = 1.0; |
| } |
| |
| if (voiceVolume != mLastVoiceVolume && ((output == mPrimaryOutput) || |
| isDirectOutput(output))) { |
| mpClientInterface->setVoiceVolume(voiceVolume, delayMs); |
| mLastVoiceVolume = voiceVolume; |
| } |
| } |
| |
| return NO_ERROR; |
| } |
| |
| bool AudioPolicyManagerCustom::isStateInCall(int state) { |
| return ((state == AUDIO_MODE_IN_CALL) || (state == AUDIO_MODE_IN_COMMUNICATION) || |
| ((state == AUDIO_MODE_RINGTONE) && (mPrevPhoneState == AUDIO_MODE_IN_CALL))); |
| } |
| |
| |
| extern "C" AudioPolicyInterface* createAudioPolicyManager( |
| AudioPolicyClientInterface *clientInterface) |
| { |
| return new AudioPolicyManager(clientInterface); |
| } |
| |
| extern "C" void destroyAudioPolicyManager(AudioPolicyInterface *interface) |
| { |
| delete interface; |
| } |
| |
| }; // namespace android |