| /* |
| * Copyright (C) 2009 The Android Open Source Project |
| * Copyright (c) 2012-2013, The Linux Foundation. All rights reserved. |
| * |
| * 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 "AudioPolicyManagerALSA" |
| //#define LOG_NDEBUG 0 |
| //#define LOG_NDDEBUG 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() |
| #define APM_AUDIO_IN_DEVICE_VIRTUAL_ALL AUDIO_DEVICE_IN_REMOTE_SUBMIX |
| |
| #include <utils/Log.h> |
| |
| #include "AudioPolicyManagerALSA.h" |
| #include <hardware/audio_effect.h> |
| #include <hardware/audio.h> |
| #include <hardware_legacy/audio_policy_conf.h> |
| #include <math.h> |
| #include <media/mediarecorder.h> |
| #include <stdio.h> |
| #include <cutils/properties.h> |
| |
| namespace android_audio_legacy { |
| |
| // ---------------------------------------------------------------------------- |
| // AudioPolicyManagerALSA |
| // ---------------------------------------------------------------------------- |
| |
| AudioParameter param; |
| |
| void AudioPolicyManager::setStrategyMute(routing_strategy strategy, |
| bool on, |
| audio_io_handle_t output, |
| int delayMs, |
| audio_devices_t device) |
| { |
| ALOGVV("setStrategyMute() strategy %d, mute %d, output %d", strategy, on, output); |
| for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) { |
| if (getStrategy((AudioSystem::stream_type)stream) == strategy) { |
| setStreamMute(stream, on, output, delayMs, device); |
| } |
| } |
| } |
| |
| uint32_t AudioPolicyManager::checkDeviceMuteStrategies(AudioOutputDescriptor *outputDesc, |
| audio_devices_t prevDevice, |
| uint32_t delayMs) |
| { |
| // mute/unmute strategies using an incompatible device combination |
| // if muting, wait for the audio in pcm buffer to be drained before proceeding |
| // if unmuting, unmute only after the specified delay |
| if (outputDesc->isDuplicated()) { |
| return 0; |
| } |
| |
| uint32_t muteWaitMs = 0; |
| audio_devices_t device = outputDesc->device(); |
| bool shouldMute = (outputDesc->refCount() != 0) && |
| (AudioSystem::popCount(device) >= 2); |
| // temporary mute output if device selection changes to avoid volume bursts due to |
| // different per device volumes |
| bool tempMute = (outputDesc->refCount() != 0) && (device != prevDevice); |
| |
| for (size_t i = 0; i < NUM_STRATEGIES; i++) { |
| audio_devices_t curDevice = getDeviceForStrategy((routing_strategy)i, false /*fromCache*/); |
| bool mute = shouldMute && (curDevice & device) && (curDevice != device); |
| bool doMute = false; |
| |
| if (mute && !outputDesc->mStrategyMutedByDevice[i]) { |
| doMute = true; |
| outputDesc->mStrategyMutedByDevice[i] = true; |
| } else if (!mute && outputDesc->mStrategyMutedByDevice[i]){ |
| doMute = true; |
| outputDesc->mStrategyMutedByDevice[i] = false; |
| } |
| if (doMute || tempMute) { |
| for (size_t j = 0; j < mOutputs.size(); j++) { |
| AudioOutputDescriptor *desc = mOutputs.valueAt(j); |
| if ((desc->supportedDevices() & outputDesc->supportedDevices()) |
| == AUDIO_DEVICE_NONE) { |
| continue; |
| } |
| audio_io_handle_t curOutput = mOutputs.keyAt(j); |
| ALOGVV("checkDeviceMuteStrategies() %s strategy %d (curDevice %04x) on output %d", |
| mute ? "muting" : "unmuting", i, curDevice, curOutput); |
| setStrategyMute((routing_strategy)i, mute, curOutput, mute ? 0 : delayMs * 4); |
| if (desc->strategyRefCount((routing_strategy)i) != 0) { |
| if (tempMute) { |
| if ((desc != outputDesc) && (desc->device() == device)) { |
| ALOGD("avoid tempmute on curOutput %d as device is same", curOutput); |
| } else { |
| setStrategyMute((routing_strategy)i, true, curOutput); |
| |
| //Change latency for tunnel/LPA player to make sure no noise on device switch |
| //Routing to BTA2DP, USB device , Proxy(WFD) will take time, increasing latency time |
| if((desc->mFlags & AUDIO_OUTPUT_FLAG_LPA) || (desc->mFlags & AUDIO_OUTPUT_FLAG_TUNNEL) |
| || (device & AUDIO_DEVICE_OUT_USB_DEVICE) || (device & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP) |
| || (device & AUDIO_DEVICE_OUT_PROXY) || (device & AUDIO_DEVICE_OUT_FM)) |
| { |
| setStrategyMute((routing_strategy)i, false, curOutput,desc->latency()*4 ,device); |
| } |
| else |
| setStrategyMute((routing_strategy)i, false, curOutput,desc->latency()*2 ,device); |
| } |
| } |
| if (tempMute || mute) { |
| if (muteWaitMs < desc->latency()) { |
| muteWaitMs = desc->latency(); |
| } |
| } |
| } |
| } |
| } |
| } |
| |
| // FIXME: should not need to double latency if volume could be applied immediately by the |
| // audioflinger mixer. We must account for the delay between now and the next time |
| // the audioflinger thread for this output will process a buffer (which corresponds to |
| // one buffer size, usually 1/2 or 1/4 of the latency). |
| muteWaitMs *= 2; |
| // wait for the PCM output buffers to empty before proceeding with the rest of the command |
| if (muteWaitMs > delayMs) { |
| muteWaitMs -= delayMs; |
| usleep(muteWaitMs * 1000); |
| return muteWaitMs; |
| } |
| return 0; |
| } |
| |
| void AudioPolicyManager::setStreamMute(int stream, |
| bool on, |
| audio_io_handle_t output, |
| int delayMs, |
| audio_devices_t device) |
| { |
| StreamDescriptor &streamDesc = mStreams[stream]; |
| AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output); |
| if (device == AUDIO_DEVICE_NONE) { |
| device = outputDesc->device(); |
| } |
| |
| ALOGVV("setStreamMute() stream %d, mute %d, output %d, mMuteCount %d device %04x", |
| stream, on, output, outputDesc->mMuteCount[stream], device); |
| |
| if (on) { |
| if (outputDesc->mMuteCount[stream] > 0) { |
| ALOGV("setStreamMute() muting already muted stream!"); |
| return; |
| } |
| if (outputDesc->mMuteCount[stream] == 0) { |
| if (streamDesc.mCanBeMuted && |
| ((stream != AudioSystem::ENFORCED_AUDIBLE) || |
| (mForceUse[AudioSystem::FOR_SYSTEM] == AudioSystem::FORCE_NONE))) { |
| checkAndSetVolume(stream, 0, output, device, delayMs); |
| } |
| } |
| // increment mMuteCount after calling checkAndSetVolume() so that volume change is not ignored |
| outputDesc->mMuteCount[stream]++; |
| } else { |
| if (outputDesc->mMuteCount[stream] == 0) { |
| ALOGV("setStreamMute() unmuting non muted stream!"); |
| return; |
| } |
| if (--outputDesc->mMuteCount[stream] == 0) { |
| checkAndSetVolume(stream, |
| streamDesc.getVolumeIndex(device), |
| output, |
| device, |
| delayMs); |
| } |
| } |
| } |
| |
| |
| status_t AudioPolicyManager::setDeviceConnectionState(audio_devices_t device, |
| AudioSystem::device_connection_state state, |
| const char *device_address) |
| { |
| SortedVector <audio_io_handle_t> outputs; |
| |
| ALOGV("setDeviceConnectionState() device: %x, state %d, address %s", device, state, device_address); |
| |
| // connect/disconnect only 1 device at a time |
| if (!audio_is_output_device(device) && !audio_is_input_device(device)) return BAD_VALUE; |
| |
| if (strlen(device_address) >= MAX_DEVICE_ADDRESS_LEN) { |
| ALOGE("setDeviceConnectionState() invalid address: %s", device_address); |
| return BAD_VALUE; |
| } |
| |
| // handle output devices |
| if (audio_is_output_device(device)) { |
| |
| //Use QCOM's a2dp and usb audio solution, no need to check here |
| /*if (!mHasA2dp && audio_is_a2dp_device(device)) { |
| ALOGE("setDeviceConnectionState() invalid A2DP device: %x", device); |
| return BAD_VALUE; |
| } |
| if (!mHasUsb && audio_is_usb_device(device)) { |
| ALOGE("setDeviceConnectionState() invalid USB audio device: %x", device); |
| return BAD_VALUE; |
| }*/ |
| if (!mHasRemoteSubmix && audio_is_remote_submix_device((audio_devices_t)device)) { |
| ALOGE("setDeviceConnectionState() invalid remote submix audio device: %x", device); |
| return BAD_VALUE; |
| } |
| |
| // 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 AudioSystem::DEVICE_STATE_AVAILABLE: |
| if (mAvailableOutputDevices & device) { |
| ALOGW("setDeviceConnectionState() device already connected: %x", device); |
| return INVALID_OPERATION; |
| } |
| ALOGV("setDeviceConnectionState() connecting device %x", device); |
| |
| if (checkOutputsForDevice(device, state, outputs) != NO_ERROR) { |
| return INVALID_OPERATION; |
| } |
| ALOGV("setDeviceConnectionState() checkOutputsForDevice() returned %d outputs", |
| outputs.size()); |
| // register new device as available |
| mAvailableOutputDevices = (audio_devices_t)(mAvailableOutputDevices | device); |
| |
| if (audio_is_a2dp_device(device)) { |
| AudioParameter param; |
| param.add(String8("a2dp_connected"), String8("true")); |
| mpClientInterface->setParameters(0, param.toString()); |
| } |
| if ( audio_is_usb_device(device)) { |
| AudioParameter param; |
| param.add(String8("usb_connected"), String8("true")); |
| mpClientInterface->setParameters(0, param.toString()); |
| } |
| if (!outputs.isEmpty()) { |
| String8 paramStr; |
| if (audio_is_a2dp_device(device)) { |
| // handle A2DP device connection |
| AudioParameter param; |
| param.add(String8(AUDIO_PARAMETER_A2DP_SINK_ADDRESS), String8(device_address)); |
| paramStr = param.toString(); |
| mA2dpDeviceAddress = String8(device_address, MAX_DEVICE_ADDRESS_LEN); |
| mA2dpSuspended = false; |
| } else if (audio_is_bluetooth_sco_device(device)) { |
| // handle SCO device connection |
| mScoDeviceAddress = String8(device_address, MAX_DEVICE_ADDRESS_LEN); |
| } else if (audio_is_usb_device(device)) { |
| // handle USB device connection |
| mUsbCardAndDevice = String8(device_address, MAX_DEVICE_ADDRESS_LEN); |
| paramStr = mUsbCardAndDevice; |
| } |
| // not currently handling multiple simultaneous submixes: ignoring remote submix |
| // case and address |
| if (!paramStr.isEmpty()) { |
| for (size_t i = 0; i < outputs.size(); i++) { |
| mpClientInterface->setParameters(outputs[i], paramStr); |
| } |
| } |
| } |
| break; |
| // handle output device disconnection |
| case AudioSystem::DEVICE_STATE_UNAVAILABLE: { |
| if (!(mAvailableOutputDevices & device)) { |
| ALOGW("setDeviceConnectionState() device not connected: %x", device); |
| return INVALID_OPERATION; |
| } |
| |
| ALOGV("setDeviceConnectionState() disconnecting device %x", device); |
| // remove device from available output devices |
| mAvailableOutputDevices = (audio_devices_t)(mAvailableOutputDevices & ~device); |
| |
| checkOutputsForDevice(device, state, outputs); |
| if (audio_is_a2dp_device(device)) { |
| // handle A2DP device disconnection |
| mA2dpDeviceAddress = ""; |
| mA2dpSuspended = false; |
| |
| AudioParameter param; |
| param.add(String8("a2dp_connected"), String8("false")); |
| mpClientInterface->setParameters(0, param.toString()); |
| |
| } else if (audio_is_bluetooth_sco_device(device)) { |
| // handle SCO device disconnection |
| mScoDeviceAddress = ""; |
| } else if (audio_is_usb_device(device)) { |
| // handle USB device disconnection |
| mUsbCardAndDevice = ""; |
| |
| AudioParameter param; |
| param.add(String8("usb_connected"), String8("false")); |
| mpClientInterface->setParameters(0, param.toString()); |
| } |
| // not currently handling multiple simultaneous submixes: ignoring remote submix |
| // case and address |
| } break; |
| |
| default: |
| ALOGE("setDeviceConnectionState() invalid state: %x", state); |
| return BAD_VALUE; |
| } |
| |
| checkA2dpSuspend(); |
| checkOutputForAllStrategies(); |
| // outputs must be closed after checkOutputForAllStrategies() is executed |
| if (!outputs.isEmpty()) { |
| for (size_t i = 0; i < outputs.size(); i++) { |
| // close unused outputs after device disconnection or direct outputs that have been |
| // opened by checkOutputsForDevice() to query dynamic parameters |
| if (state == AudioSystem::DEVICE_STATE_UNAVAILABLE) { |
| closeOutput(outputs[i]); |
| } |
| } |
| } |
| |
| updateDevicesAndOutputs(); |
| #ifdef QCOM_PROXY_DEVICE_ENABLED |
| if (state == AudioSystem::DEVICE_STATE_AVAILABLE && |
| audio_is_a2dp_device(device) && |
| (mAvailableOutputDevices & AUDIO_DEVICE_OUT_PROXY)) { |
| ALOGV("Delay the proxy device open"); |
| return NO_ERROR; |
| } |
| #endif |
| |
| audio_devices_t newDevice = getNewDevice(mPrimaryOutput, false /*fromCache*/); |
| #ifdef QCOM_FM_ENABLED |
| if(device == AUDIO_DEVICE_OUT_FM) { |
| if (state == AudioSystem::DEVICE_STATE_AVAILABLE) { |
| ALOGV("setDeviceConnectionState() changeRefCount Inc"); |
| mOutputs.valueFor(mPrimaryOutput)->changeRefCount(AudioSystem::FM, 1); |
| newDevice = (audio_devices_t)(AudioPolicyManagerBase::getNewDevice(mPrimaryOutput, false) | AUDIO_DEVICE_OUT_FM); |
| } |
| else { |
| ALOGV("setDeviceConnectionState() changeRefCount Dec"); |
| mOutputs.valueFor(mPrimaryOutput)->changeRefCount(AudioSystem::FM, -1); |
| } |
| |
| AudioParameter param = AudioParameter(); |
| param.addInt(String8(AudioParameter::keyHandleFm), (int)newDevice); |
| ALOGV("setDeviceConnectionState() setParameters handle_fm"); |
| mpClientInterface->setParameters(mPrimaryOutput, param.toString()); |
| } |
| #endif |
| for (int i = mOutputs.size() -1; i >= 0; i--) { |
| audio_devices_t newDevice = getNewDevice(mOutputs.keyAt(i), true /*fromCache*/); |
| #ifdef QCOM_ANC_HEADSET_ENABLED |
| if(device == AUDIO_DEVICE_OUT_ANC_HEADPHONE || |
| device == AUDIO_DEVICE_OUT_ANC_HEADSET) { |
| if(newDevice == 0){ |
| newDevice = getDeviceForStrategy(STRATEGY_MEDIA, false); |
| } |
| } |
| #endif |
| setOutputDevice(mOutputs.keyAt(i), |
| getNewDevice(mOutputs.keyAt(i), true /*fromCache*/), |
| true, |
| 0); |
| } |
| |
| if (device == AUDIO_DEVICE_OUT_WIRED_HEADSET) { |
| device = AUDIO_DEVICE_IN_WIRED_HEADSET; |
| } else if (device == AUDIO_DEVICE_OUT_BLUETOOTH_SCO || |
| device == AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET || |
| device == AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT) { |
| device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET; |
| } else if(device == AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET){ |
| device = AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET; |
| #ifdef QCOM_ANC_HEADSET_ENABLED |
| } else if(device == AUDIO_DEVICE_OUT_ANC_HEADSET){ |
| device = AUDIO_DEVICE_IN_ANC_HEADSET; //wait for actual ANC device |
| #endif |
| } else { |
| return NO_ERROR; |
| } |
| } |
| // handle input devices |
| if (audio_is_input_device(device)) { |
| |
| switch (state) |
| { |
| // handle input device connection |
| case AudioSystem::DEVICE_STATE_AVAILABLE: { |
| if (mAvailableInputDevices & device) { |
| ALOGW("setDeviceConnectionState() device already connected: %d", device); |
| return INVALID_OPERATION; |
| } |
| mAvailableInputDevices = mAvailableInputDevices | (device & ~AUDIO_DEVICE_BIT_IN); |
| } |
| break; |
| |
| // handle input device disconnection |
| case AudioSystem::DEVICE_STATE_UNAVAILABLE: { |
| if (!(mAvailableInputDevices & device)) { |
| ALOGW("setDeviceConnectionState() device not connected: %d", device); |
| return INVALID_OPERATION; |
| } |
| mAvailableInputDevices = (audio_devices_t) (mAvailableInputDevices & ~device); |
| } break; |
| |
| default: |
| ALOGE("setDeviceConnectionState() invalid state: %x", state); |
| return BAD_VALUE; |
| } |
| |
| audio_io_handle_t activeInput = getActiveInput(); |
| if (activeInput != 0) { |
| AudioInputDescriptor *inputDesc = mInputs.valueFor(activeInput); |
| audio_devices_t newDevice = getDeviceForInputSource(inputDesc->mInputSource); |
| if ((newDevice != AUDIO_DEVICE_NONE) && (newDevice != inputDesc->mDevice)) { |
| ALOGV("setDeviceConnectionState() changing device from %x to %x for input %d", |
| inputDesc->mDevice, newDevice, activeInput); |
| inputDesc->mDevice = newDevice; |
| AudioParameter param = AudioParameter(); |
| param.addInt(String8(AudioParameter::keyRouting), (int)newDevice); |
| mpClientInterface->setParameters(activeInput, param.toString()); |
| } |
| } |
| |
| return NO_ERROR; |
| } |
| |
| ALOGW("setDeviceConnectionState() invalid device: %x", device); |
| return BAD_VALUE; |
| } |
| |
| |
| AudioSystem::device_connection_state AudioPolicyManager::getDeviceConnectionState(audio_devices_t device, |
| const char *device_address) |
| { |
| AudioSystem::device_connection_state state = AudioSystem::DEVICE_STATE_UNAVAILABLE; |
| String8 address = String8(device_address); |
| if (audio_is_output_device(device)) { |
| if (device & mAvailableOutputDevices) { |
| if (audio_is_a2dp_device(device) && |
| ((address != "" && mA2dpDeviceAddress != address))) { |
| return state; |
| } |
| if (audio_is_bluetooth_sco_device(device) && |
| address != "" && mScoDeviceAddress != address) { |
| return state; |
| } |
| if (audio_is_usb_device(device) && |
| ((address != "" && mUsbCardAndDevice != address))) { |
| ALOGE("getDeviceConnectionState() invalid device: %x", device); |
| return state; |
| } |
| if (audio_is_remote_submix_device((audio_devices_t)device) && !mHasRemoteSubmix) { |
| return state; |
| } |
| state = AudioSystem::DEVICE_STATE_AVAILABLE; |
| } |
| } else if (audio_is_input_device(device)) { |
| if (device & mAvailableInputDevices) { |
| state = AudioSystem::DEVICE_STATE_AVAILABLE; |
| } |
| } |
| |
| return state; |
| } |
| |
| |
| void AudioPolicyManager::setPhoneState(int state) |
| { |
| ALOGV("setPhoneState() state %d", state); |
| audio_devices_t newDevice = AUDIO_DEVICE_NONE; |
| if (state < 0 || state >= AudioSystem::NUM_MODES) { |
| 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 < AudioSystem::NUM_STREAM_TYPES; stream++) { |
| handleIncallSonification(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; |
| } 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; |
| } 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 |
| // Need to update A2DP suspend first then getNewDevice(from cache) |
| checkA2dpSuspend(); |
| checkOutputForAllStrategies(); |
| updateDevicesAndOutputs(); |
| newDevice = getNewDevice(mPrimaryOutput, false /*fromCache*/); |
| |
| AudioOutputDescriptor *hwOutputDesc = mOutputs.valueFor(mPrimaryOutput); |
| |
| // force routing command to audio hardware when ending call |
| // even if no device change is needed |
| if (isStateInCall(oldState) && newDevice == AUDIO_DEVICE_NONE) { |
| newDevice = hwOutputDesc->device(); |
| } |
| |
| // when changing from ring tone to in call mode, mute the ringing tone |
| // immediately and delay the route change to avoid sending the ring tone |
| // tail into the earpiece or headset. |
| int delayMs = 0; |
| if (isStateInCall(state) && oldState == AudioSystem::MODE_RINGTONE) { |
| // delay the device change command by twice the output latency to have some margin |
| // and be sure that audio buffers not yet affected by the mute are out when |
| // we actually apply the route change |
| delayMs = hwOutputDesc->mLatency*2; |
| setStreamMute(AudioSystem::RING, true, mPrimaryOutput); |
| } |
| |
| // Ignore the delay to enable voice call on this target as the enabling the |
| // voice call has enough delay to make sure the ringtone audio completely |
| // played out |
| if (state == AUDIO_MODE_IN_CALL && |
| #ifdef QCOM_CSDCLIENT_ENABLED |
| platform_is_Fusion3() && |
| #endif |
| oldState == AUDIO_MODE_RINGTONE) { |
| delayMs = 40; |
| } |
| |
| if (isStateInCall(state)) { |
| for (size_t i = 0; i < mOutputs.size(); i++) { |
| AudioOutputDescriptor *desc = mOutputs.valueAt(i); |
| //take the biggest latency for all outputs |
| if (delayMs < desc->mLatency*2) { |
| delayMs = desc->mLatency*2; |
| } |
| //mute STRATEGY_MEDIA on all outputs |
| if (desc->strategyRefCount(STRATEGY_MEDIA) != 0) { |
| setStrategyMute(STRATEGY_MEDIA, true, mOutputs.keyAt(i)); |
| setStrategyMute(STRATEGY_MEDIA, false, mOutputs.keyAt(i), MUTE_TIME_MS, |
| getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/)); |
| } |
| } |
| } |
| |
| // change routing is necessary |
| setOutputDevice(mPrimaryOutput, newDevice, force, delayMs); |
| |
| //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 = getNewDevice(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); |
| // unmute the ringing tone after a sufficient delay if it was muted before |
| // setting output device above |
| if (oldState == AudioSystem::MODE_RINGTONE) { |
| setStreamMute(AudioSystem::RING, false, mPrimaryOutput, MUTE_TIME_MS); |
| } |
| for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) { |
| handleIncallSonification(stream, true, true); |
| } |
| } |
| |
| // Flag that ringtone volume must be limited to music volume until we exit MODE_RINGTONE |
| if (state == AudioSystem::MODE_RINGTONE && |
| isStreamActive(AudioSystem::MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY)) { |
| mLimitRingtoneVolume = true; |
| } else { |
| mLimitRingtoneVolume = false; |
| } |
| } |
| |
| |
| void AudioPolicyManager::setForceUse(AudioSystem::force_use usage, AudioSystem::forced_config config) |
| { |
| ALOGV("setForceUse() usage %d, config %d, mPhoneState %d", usage, config, mPhoneState); |
| |
| bool forceVolumeReeval = false; |
| switch(usage) { |
| case AudioSystem::FOR_COMMUNICATION: |
| if (config != AudioSystem::FORCE_SPEAKER && config != AudioSystem::FORCE_BT_SCO && |
| config != AudioSystem::FORCE_NONE) { |
| ALOGW("setForceUse() invalid config %d for FOR_COMMUNICATION", config); |
| return; |
| } |
| forceVolumeReeval = true; |
| mForceUse[usage] = config; |
| break; |
| case AudioSystem::FOR_MEDIA: |
| if (config != AudioSystem::FORCE_HEADPHONES && config != AudioSystem::FORCE_BT_A2DP && |
| config != AudioSystem::FORCE_WIRED_ACCESSORY && |
| config != AudioSystem::FORCE_ANALOG_DOCK && |
| config != AudioSystem::FORCE_DIGITAL_DOCK && config != AudioSystem::FORCE_NONE && |
| config != AudioSystem::FORCE_NO_BT_A2DP && config != AudioSystem::FORCE_SPEAKER) { |
| ALOGW("setForceUse() invalid config %d for FOR_MEDIA", config); |
| return; |
| } |
| mForceUse[usage] = config; |
| break; |
| case AudioSystem::FOR_RECORD: |
| if (config != AudioSystem::FORCE_BT_SCO && config != AudioSystem::FORCE_WIRED_ACCESSORY && |
| config != AudioSystem::FORCE_NONE) { |
| ALOGW("setForceUse() invalid config %d for FOR_RECORD", config); |
| return; |
| } |
| mForceUse[usage] = config; |
| break; |
| case AudioSystem::FOR_DOCK: |
| if (config != AudioSystem::FORCE_NONE && config != AudioSystem::FORCE_BT_CAR_DOCK && |
| config != AudioSystem::FORCE_BT_DESK_DOCK && |
| config != AudioSystem::FORCE_WIRED_ACCESSORY && |
| config != AudioSystem::FORCE_ANALOG_DOCK && |
| config != AudioSystem::FORCE_DIGITAL_DOCK) { |
| ALOGW("setForceUse() invalid config %d for FOR_DOCK", config); |
| } |
| forceVolumeReeval = true; |
| mForceUse[usage] = config; |
| break; |
| case AudioSystem::FOR_SYSTEM: |
| if (config != AudioSystem::FORCE_NONE && |
| config != AudioSystem::FORCE_SYSTEM_ENFORCED) { |
| ALOGW("setForceUse() invalid config %d for FOR_SYSTEM", config); |
| } |
| forceVolumeReeval = true; |
| 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(); |
| for (int i = mOutputs.size() -1; i >= 0; i--) { |
| audio_io_handle_t output = mOutputs.keyAt(i); |
| audio_devices_t newDevice = getNewDevice(output, true /*fromCache*/); |
| 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) { |
| AudioInputDescriptor *inputDesc = mInputs.valueFor(activeInput); |
| audio_devices_t newDevice = getDeviceForInputSource(inputDesc->mInputSource); |
| if ((newDevice != AUDIO_DEVICE_NONE) && (newDevice != inputDesc->mDevice)) { |
| ALOGV("setForceUse() changing device from %x to %x for input %d", |
| inputDesc->mDevice, newDevice, activeInput); |
| inputDesc->mDevice = newDevice; |
| AudioParameter param = AudioParameter(); |
| param.addInt(String8(AudioParameter::keyRouting), (int)newDevice); |
| mpClientInterface->setParameters(activeInput, param.toString()); |
| } |
| } |
| |
| } |
| |
| audio_io_handle_t AudioPolicyManager::getOutput(AudioSystem::stream_type stream, |
| uint32_t samplingRate, |
| uint32_t format, |
| uint32_t channelMask, |
| AudioSystem::output_flags flags) |
| { |
| audio_io_handle_t output = 0; |
| uint32_t latency = 0; |
| routing_strategy strategy = getStrategy((AudioSystem::stream_type)stream); |
| audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/); |
| ALOGV("getOutput() stream %d, samplingRate %d, format %d, channelMask %x, flags %x", |
| stream, samplingRate, format, channelMask, flags); |
| |
| #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"); |
| AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(NULL); |
| outputDesc->mDevice = mTestDevice; |
| outputDesc->mSamplingRate = mTestSamplingRate; |
| outputDesc->mFormat = mTestFormat; |
| outputDesc->mChannelMask = mTestChannels; |
| outputDesc->mLatency = mTestLatencyMs; |
| outputDesc->mFlags = (audio_output_flags_t)(mDirectOutput ? AudioSystem::OUTPUT_FLAG_DIRECT : 0); |
| outputDesc->mRefCount[stream] = 0; |
| mTestOutputs[mCurOutput] = mpClientInterface->openOutput(0, &outputDesc->mDevice, |
| &outputDesc->mSamplingRate, |
| &outputDesc->mFormat, |
| &outputDesc->mChannelMask, |
| &outputDesc->mLatency, |
| outputDesc->mFlags); |
| if (mTestOutputs[mCurOutput]) { |
| 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 |
| |
| IOProfile *profile = getProfileForDirectOutput(device, |
| samplingRate, |
| format, |
| channelMask, |
| (audio_output_flags_t)flags); |
| |
| if (profile != NULL) { |
| |
| ALOGV("getOutput() opening direct output device %x", device); |
| |
| AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(profile); |
| outputDesc->mDevice = device; |
| outputDesc->mSamplingRate = samplingRate; |
| outputDesc->mFormat = (audio_format_t)format; |
| outputDesc->mChannelMask = (audio_channel_mask_t)channelMask; |
| outputDesc->mLatency = 0; |
| outputDesc->mFlags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT);; |
| outputDesc->mRefCount[stream] = 0; |
| outputDesc->mStopTime[stream] = 0; |
| output = mpClientInterface->openOutput(profile->mModule->mHandle, |
| &outputDesc->mDevice, |
| &outputDesc->mSamplingRate, |
| &outputDesc->mFormat, |
| &outputDesc->mChannelMask, |
| &outputDesc->mLatency, |
| outputDesc->mFlags); |
| |
| // only accept an output with the requested parameters |
| if (output == 0 || |
| (samplingRate != 0 && samplingRate != outputDesc->mSamplingRate) || |
| (format != 0 && format != outputDesc->mFormat) || |
| (channelMask != 0 && channelMask != outputDesc->mChannelMask)) { |
| 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 != 0) { |
| mpClientInterface->closeOutput(output); |
| } |
| delete outputDesc; |
| return 0; |
| } |
| addOutput(output, outputDesc); |
| ALOGV("getOutput() returns direct output %d", output); |
| return output; |
| } |
| |
| // ignoring channel mask due to downmix capability in mixer |
| |
| // open a non direct output |
| |
| // 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); |
| |
| output = selectOutput(outputs, flags); |
| |
| 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 AudioPolicyManager::startOutput(audio_io_handle_t output, |
| AudioSystem::stream_type stream, |
| int session) |
| { |
| ALOGV("startOutput() output %d, stream %d, session %d", output, stream, session); |
| ssize_t index = mOutputs.indexOfKey(output); |
| if (index < 0) { |
| ALOGW("startOutput() unknow output %d", output); |
| return BAD_VALUE; |
| } |
| |
| AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index); |
| |
| // increment usage count for this stream on the requested output: |
| // NOTE that the usage count is the same for duplicated output and hardware output which is |
| // necessary for a correct control of hardware output routing by startOutput() and stopOutput() |
| outputDesc->changeRefCount(stream, 1); |
| |
| if (outputDesc->mRefCount[stream] == 1) { |
| audio_devices_t newDevice = getNewDevice(output, false /*fromCache*/); |
| routing_strategy strategy = getStrategy(stream); |
| bool shouldWait = (strategy == STRATEGY_SONIFICATION) || |
| (strategy == STRATEGY_SONIFICATION_RESPECTFUL); |
| uint32_t waitMs = 0; |
| uint32_t muteWaitMs = 0; |
| bool force = false; |
| for (size_t i = 0; i < mOutputs.size(); i++) { |
| AudioOutputDescriptor *desc = mOutputs.valueAt(i); |
| if (desc != outputDesc) { |
| // force a device change if any other output is managed by the same hw |
| // module and has a current device selection that differs from selected device. |
| // In this case, the audio HAL must receive the new device selection so that it can |
| // change the device currently selected by the other active output. |
| if (outputDesc->sharesHwModuleWith(desc) && |
| desc->device() != newDevice) { |
| force = true; |
| } |
| // wait for audio on other active outputs to be presented when starting |
| // a notification so that audio focus effect can propagate. |
| if (shouldWait && (desc->refCount() != 0) && (waitMs < desc->latency())) { |
| waitMs = desc->latency(); |
| } |
| } |
| } |
| |
| #ifdef QCOM_FM_ENABLED |
| if(stream == AudioSystem::FM && output == getA2dpOutput()) { |
| muteWaitMs = setOutputDevice(output, newDevice, true); |
| } else |
| #endif |
| { |
| muteWaitMs = setOutputDevice(output, newDevice, force); |
| } |
| |
| // handle special case for sonification while in call |
| if (isInCall()) { |
| handleIncallSonification(stream, true, false); |
| } |
| |
| // apply volume rules for current stream and device if necessary |
| checkAndSetVolume(stream, |
| mStreams[stream].getVolumeIndex(newDevice), |
| output, |
| newDevice); |
| |
| // update the outputs if starting an output with a stream that can affect notification |
| // routing |
| handleNotificationRoutingForStream(stream); |
| if (waitMs > muteWaitMs) { |
| usleep((waitMs - muteWaitMs) * 2 * 1000); |
| } |
| } else { |
| // handle special case for sonification while in call |
| if (isInCall()) { |
| handleIncallSonification(stream, true, false); |
| } |
| } |
| return NO_ERROR; |
| } |
| |
| status_t AudioPolicyManager::stopOutput(audio_io_handle_t output, |
| AudioSystem::stream_type 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() unknow output %d", output); |
| return BAD_VALUE; |
| } |
| |
| AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index); |
| |
| // handle special case for sonification while in call |
| if (isInCall()) { |
| handleIncallSonification(stream, false, false); |
| } |
| |
| if (outputDesc->mRefCount[stream] > 0) { |
| // decrement usage count of this stream on the output |
| outputDesc->changeRefCount(stream, -1); |
| // store time at which the stream was stopped - see isStreamActive() |
| if (outputDesc->mRefCount[stream] == 0) { |
| outputDesc->mStopTime[stream] = systemTime(); |
| |
| if ((outputDesc->mRefCount[AUDIO_STREAM_RING]!= 0) && (stream == AUDIO_STREAM_VOICE_CALL)) { |
| // When AUDIO_STREAM_RING is present, Send Mute on RING |
| // if it gets stopOutput on AUDIO_STREAM_VOICE_CALL |
| setStreamMute(AudioSystem::RING, true, mPrimaryOutput); |
| } |
| |
| audio_devices_t newDevice = getNewDevice(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); |
| AudioOutputDescriptor *desc = mOutputs.valueAt(i); |
| if (curOutput != output && |
| desc->refCount() != 0 && |
| outputDesc->sharesHwModuleWith(desc) && |
| newDevice != desc->device()) { |
| setOutputDevice(curOutput, |
| getNewDevice(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 AudioPolicyManager::getInput(int inputSource, |
| uint32_t samplingRate, |
| uint32_t format, |
| uint32_t channelMask, |
| AudioSystem::audio_in_acoustics acoustics) |
| { |
| audio_io_handle_t input = 0; |
| audio_devices_t device = getDeviceForInputSource(inputSource); |
| |
| ALOGV("getInput() inputSource %d, samplingRate %d, format %d, channelMask %x, acoustics %x", |
| inputSource, samplingRate, format, channelMask, acoustics); |
| |
| if (device == AUDIO_DEVICE_NONE) { |
| ALOGW("getInput() could not find device for inputSource %d", inputSource); |
| return 0; |
| } |
| |
| // adapt channel selection to input source |
| switch(inputSource) { |
| case AUDIO_SOURCE_VOICE_UPLINK: |
| channelMask |= AudioSystem::CHANNEL_IN_VOICE_UPLINK; |
| break; |
| case AUDIO_SOURCE_VOICE_DOWNLINK: |
| channelMask |= AudioSystem::CHANNEL_IN_VOICE_DNLINK; |
| break; |
| case AUDIO_SOURCE_VOICE_CALL: |
| channelMask |= (AudioSystem::CHANNEL_IN_VOICE_UPLINK | AudioSystem::CHANNEL_IN_VOICE_DNLINK); |
| break; |
| default: |
| break; |
| } |
| |
| IOProfile *profile = getInputProfile(device, |
| samplingRate, |
| format, |
| channelMask); |
| if (profile == NULL) { |
| ALOGW("getInput() could not find profile for device %04x, samplingRate %d, format %d," |
| "channelMask %04x", |
| device, samplingRate, format, channelMask); |
| return 0; |
| } |
| |
| if (profile->mModule->mHandle == 0) { |
| ALOGE("getInput(): HW module %s not opened", profile->mModule->mName); |
| return 0; |
| } |
| |
| AudioInputDescriptor *inputDesc = new AudioInputDescriptor(profile); |
| |
| inputDesc->mInputSource = inputSource; |
| inputDesc->mDevice = device; |
| inputDesc->mSamplingRate = samplingRate; |
| inputDesc->mFormat = (audio_format_t)format; |
| inputDesc->mChannelMask = (audio_channel_mask_t)channelMask; |
| inputDesc->mRefCount = 0; |
| input = mpClientInterface->openInput(profile->mModule->mHandle, |
| &inputDesc->mDevice, |
| &inputDesc->mSamplingRate, |
| &inputDesc->mFormat, |
| &inputDesc->mChannelMask); |
| |
| // only accept input with the exact requested set of parameters |
| if (input == 0 || |
| (samplingRate != inputDesc->mSamplingRate) || |
| (format != inputDesc->mFormat) || |
| (channelMask != inputDesc->mChannelMask)) { |
| ALOGV("getInput() failed opening input: samplingRate %d, format %d, channelMask %d", |
| samplingRate, format, channelMask); |
| if (input != 0) { |
| mpClientInterface->closeInput(input); |
| } |
| delete inputDesc; |
| return 0; |
| } |
| mInputs.add(input, inputDesc); |
| return input; |
| } |
| |
| /* |
| Overwriting this function from base class to allow 2 acitve AudioRecord clients in case of FM. |
| One for FM A2DP playbck and other for FM recording. |
| */ |
| status_t AudioPolicyManager::startInput(audio_io_handle_t input) |
| { |
| ALOGV("startInput() input %d", input); |
| ssize_t index = mInputs.indexOfKey(input); |
| if (index < 0) { |
| ALOGW("startInput() unknow input %d", input); |
| return BAD_VALUE; |
| } |
| AudioInputDescriptor *inputDesc = mInputs.valueAt(index); |
| |
| /* |
| #ifdef AUDIO_POLICY_TEST |
| if (mTestInput == 0) |
| #endif //AUDIO_POLICY_TEST |
| { |
| // refuse 2 active AudioRecord clients at the same time |
| if (getActiveInput() != 0) { |
| ALOGW("startInput() input %d failed: other input already started", input); |
| return INVALID_OPERATION; |
| } |
| } |
| */ |
| |
| AudioParameter param = AudioParameter(); |
| param.addInt(String8(AudioParameter::keyRouting), (int)inputDesc->mDevice); |
| |
| param.addInt(String8(AudioParameter::keyInputSource), (int)inputDesc->mInputSource); |
| |
| // use Voice Recognition mode or not for this input based on input source |
| int vr_enabled = inputDesc->mInputSource == AUDIO_SOURCE_VOICE_RECOGNITION ? 1 : 0; |
| param.addInt(String8("vr_mode"), vr_enabled); |
| ALOGV("AudioPolicyManager::startInput(%d), setting vr_mode to %d", inputDesc->mInputSource, vr_enabled); |
| |
| //to pass on if camcorder mode is enabled to HAL |
| int camcorder_enabled = inputDesc->mInputSource == AUDIO_SOURCE_CAMCORDER ? 1 : 0; |
| param.addInt(String8("camcorder_mode"), camcorder_enabled); |
| |
| mpClientInterface->setParameters(input, param.toString()); |
| |
| inputDesc->mRefCount = 1; |
| return NO_ERROR; |
| } |
| |
| status_t AudioPolicyManager::stopInput(audio_io_handle_t input) |
| { |
| ALOGV("stopInput() input %d", input); |
| ssize_t index = mInputs.indexOfKey(input); |
| if (index < 0) { |
| ALOGW("stopInput() unknow input %d", input); |
| return BAD_VALUE; |
| } |
| AudioInputDescriptor *inputDesc = mInputs.valueAt(index); |
| |
| if (inputDesc->mRefCount == 0) { |
| ALOGW("stopInput() input %d already stopped", input); |
| return INVALID_OPERATION; |
| } else { |
| AudioParameter param = AudioParameter(); |
| param.addInt(String8(AudioParameter::keyRouting), 0); |
| mpClientInterface->setParameters(input, param.toString()); |
| setOutputDevice(mPrimaryOutput, getNewDevice(mPrimaryOutput, true), true); |
| inputDesc->mRefCount = 0; |
| return NO_ERROR; |
| } |
| } |
| |
| |
| status_t AudioPolicyManager::setStreamVolumeIndex(AudioSystem::stream_type 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) && (AUDIO_DEVICE_OUT_DEFAULT != AUDIO_DEVICE_OUT_SPEAKER)) { |
| 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()); |
| if (device == curDevice) { |
| status_t volStatus = checkAndSetVolume(stream, index, mOutputs.keyAt(i), curDevice); |
| if (volStatus != NO_ERROR) { |
| status = volStatus; |
| } |
| } |
| } |
| return status; |
| } |
| |
| |
| status_t AudioPolicyManager::checkOutputsForDevice(audio_devices_t device, |
| AudioSystem::device_connection_state state, |
| SortedVector<audio_io_handle_t>& outputs) |
| { |
| AudioOutputDescriptor *desc; |
| |
| if (state == AudioSystem::DEVICE_STATE_AVAILABLE) { |
| // first list already open outputs that can be routed to this device |
| for (size_t i = 0; i < mOutputs.size(); i++) { |
| desc = mOutputs.valueAt(i); |
| if (!desc->isDuplicated() && (desc->mProfile->mSupportedDevices & device)) { |
| ALOGV("checkOutputsForDevice(): adding opened output %d", mOutputs.keyAt(i)); |
| outputs.add(mOutputs.keyAt(i)); |
| } |
| } |
| // then look for output profiles that can be routed to this device |
| SortedVector<IOProfile *> profiles; |
| for (size_t i = 0; i < mHwModules.size(); i++) |
| { |
| if (mHwModules[i]->mHandle == 0) { |
| continue; |
| } |
| for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++) |
| { |
| if (mHwModules[i]->mOutputProfiles[j]->mSupportedDevices & device) { |
| ALOGV("checkOutputsForDevice(): adding profile %d from module %d", j, i); |
| profiles.add(mHwModules[i]->mOutputProfiles[j]); |
| } |
| } |
| } |
| |
| if (profiles.isEmpty() && outputs.isEmpty()) { |
| ALOGW("checkOutputsForDevice(): No output available for device %04x", device); |
| return BAD_VALUE; |
| } |
| |
| // open outputs for matching profiles if needed. Direct outputs are also opened to |
| // query for dynamic parameters and will be closed later by setDeviceConnectionState() |
| for (ssize_t profile_index = 0; profile_index < (ssize_t)profiles.size(); profile_index++) { |
| IOProfile *profile = profiles[profile_index]; |
| |
| // nothing to do if one output is already opened for this profile |
| size_t j; |
| for (j = 0; j < mOutputs.size(); j++) { |
| desc = mOutputs.valueAt(j); |
| if (!desc->isDuplicated() && desc->mProfile == profile) { |
| break; |
| } |
| } |
| if (j != mOutputs.size()) { |
| continue; |
| } |
| |
| ALOGV("opening output for device %08x", device); |
| desc = new AudioOutputDescriptor(profile); |
| desc->mDevice = device; |
| audio_io_handle_t output = 0; |
| if (!(desc->mFlags & AUDIO_OUTPUT_FLAG_LPA || desc->mFlags & AUDIO_OUTPUT_FLAG_TUNNEL || |
| desc->mFlags & AUDIO_OUTPUT_FLAG_VOIP_RX)) { |
| output = mpClientInterface->openOutput(profile->mModule->mHandle, |
| &desc->mDevice, |
| &desc->mSamplingRate, |
| &desc->mFormat, |
| &desc->mChannelMask, |
| &desc->mLatency, |
| desc->mFlags); |
| } |
| if (output != 0) { |
| if (desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) { |
| String8 reply; |
| char *value; |
| if (profile->mSamplingRates[0] == 0) { |
| reply = mpClientInterface->getParameters(output, |
| String8(AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES)); |
| ALOGV("checkOutputsForDevice() direct output sup sampling rates %s", |
| reply.string()); |
| value = strpbrk((char *)reply.string(), "="); |
| if (value != NULL) { |
| loadSamplingRates(value, profile); |
| } |
| } |
| if (profile->mFormats[0] == 0) { |
| reply = mpClientInterface->getParameters(output, |
| String8(AUDIO_PARAMETER_STREAM_SUP_FORMATS)); |
| ALOGV("checkOutputsForDevice() direct output sup formats %s", |
| reply.string()); |
| value = strpbrk((char *)reply.string(), "="); |
| if (value != NULL) { |
| loadFormats(value, profile); |
| } |
| } |
| if (profile->mChannelMasks[0] == 0) { |
| reply = mpClientInterface->getParameters(output, |
| String8(AUDIO_PARAMETER_STREAM_SUP_CHANNELS)); |
| ALOGV("checkOutputsForDevice() direct output sup channel masks %s", |
| reply.string()); |
| value = strpbrk((char *)reply.string(), "="); |
| if (value != NULL) { |
| loadOutChannels(value + 1, profile); |
| } |
| } |
| if (((profile->mSamplingRates[0] == 0) && |
| (profile->mSamplingRates.size() < 2)) || |
| ((profile->mFormats[0] == 0) && |
| (profile->mFormats.size() < 2)) || |
| ((profile->mFormats[0] == 0) && |
| (profile->mChannelMasks.size() < 2))) { |
| ALOGW("checkOutputsForDevice() direct output missing param"); |
| output = 0; |
| } else { |
| addOutput(output, desc); |
| } |
| } else { |
| audio_io_handle_t duplicatedOutput = 0; |
| // add output descriptor |
| addOutput(output, desc); |
| // set initial stream volume for device |
| applyStreamVolumes(output, device, 0, true); |
| |
| //TODO: configure audio effect output stage here |
| |
| // open a duplicating output thread for the new output and the primary output |
| duplicatedOutput = mpClientInterface->openDuplicateOutput(output, |
| mPrimaryOutput); |
| if (duplicatedOutput != 0) { |
| // add duplicated output descriptor |
| AudioOutputDescriptor *dupOutputDesc = new AudioOutputDescriptor(NULL); |
| dupOutputDesc->mOutput1 = mOutputs.valueFor(mPrimaryOutput); |
| dupOutputDesc->mOutput2 = mOutputs.valueFor(output); |
| dupOutputDesc->mSamplingRate = desc->mSamplingRate; |
| dupOutputDesc->mFormat = desc->mFormat; |
| dupOutputDesc->mChannelMask = desc->mChannelMask; |
| dupOutputDesc->mLatency = desc->mLatency; |
| addOutput(duplicatedOutput, dupOutputDesc); |
| applyStreamVolumes(duplicatedOutput, device, 0, true); |
| } else { |
| ALOGW("checkOutputsForDevice() could not open dup output for %d and %d", |
| mPrimaryOutput, output); |
| mpClientInterface->closeOutput(output); |
| mOutputs.removeItem(output); |
| output = 0; |
| } |
| } |
| } |
| if (output == 0) { |
| ALOGW("checkOutputsForDevice() could not open output for device %x", device); |
| delete desc; |
| profiles.removeAt(profile_index); |
| profile_index--; |
| } else { |
| outputs.add(output); |
| ALOGV("checkOutputsForDevice(): adding output %d", output); |
| } |
| } |
| |
| if (profiles.isEmpty()) { |
| ALOGW("checkOutputsForDevice(): No output available for device %04x", device); |
| return BAD_VALUE; |
| } |
| } else { |
| // check if one opened output is not needed any more after disconnecting one device |
| for (size_t i = 0; i < mOutputs.size(); i++) { |
| desc = mOutputs.valueAt(i); |
| if (!desc->isDuplicated() && |
| !(desc->mProfile->mSupportedDevices & mAvailableOutputDevices)) { |
| ALOGV("checkOutputsForDevice(): disconnecting adding output %d", mOutputs.keyAt(i)); |
| outputs.add(mOutputs.keyAt(i)); |
| } |
| } |
| for (size_t i = 0; i < mHwModules.size(); i++) |
| { |
| if (mHwModules[i]->mHandle == 0) { |
| continue; |
| } |
| for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++) |
| { |
| IOProfile *profile = mHwModules[i]->mOutputProfiles[j]; |
| if ((profile->mSupportedDevices & device) && |
| (profile->mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { |
| ALOGV("checkOutputsForDevice(): clearing direct output profile %d on module %d", |
| j, i); |
| if (profile->mSamplingRates[0] == 0) { |
| profile->mSamplingRates.clear(); |
| profile->mSamplingRates.add(0); |
| } |
| if (profile->mFormats[0] == 0) { |
| profile->mFormats.clear(); |
| profile->mFormats.add((audio_format_t)0); |
| } |
| if (profile->mChannelMasks[0] == 0) { |
| profile->mChannelMasks.clear(); |
| profile->mChannelMasks.add((audio_channel_mask_t)0); |
| } |
| } |
| } |
| } |
| } |
| return NO_ERROR; |
| } |
| |
| audio_devices_t AudioPolicyManager::getNewDevice(audio_io_handle_t output, bool fromCache) |
| { |
| audio_devices_t device = AUDIO_DEVICE_NONE; |
| |
| AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output); |
| AudioOutputDescriptor *primaryOutputDesc = mOutputs.valueFor(mPrimaryOutput); |
| // check the following by order of priority to request a routing change if necessary: |
| // 1: the strategy enforced audible is active on the output: |
| // use device for strategy enforced audible |
| // 2: we are in call or the strategy phone is active on the output: |
| // use device for strategy phone |
| // 3: the strategy sonification is active on the output: |
| // use device for strategy sonification |
| // 4: the strategy "respectful" sonification is active on the output: |
| // use device for strategy "respectful" sonification |
| // 5: the strategy media is active on the output: |
| // use device for strategy media |
| // 6: the strategy DTMF is active on the output: |
| // use device for strategy DTMF |
| if (outputDesc->isUsedByStrategy(STRATEGY_ENFORCED_AUDIBLE)) { |
| device = getDeviceForStrategy(STRATEGY_ENFORCED_AUDIBLE, fromCache); |
| } else if (isInCall() || |
| outputDesc->isUsedByStrategy(STRATEGY_PHONE)) { |
| device = getDeviceForStrategy(STRATEGY_PHONE, fromCache); |
| } else if (outputDesc->isUsedByStrategy(STRATEGY_SONIFICATION)){ |
| device = getDeviceForStrategy(STRATEGY_SONIFICATION, fromCache); |
| } else if (outputDesc->isUsedByStrategy(STRATEGY_SONIFICATION_RESPECTFUL)) { |
| device = getDeviceForStrategy(STRATEGY_SONIFICATION_RESPECTFUL, fromCache); |
| } else if (outputDesc->isUsedByStrategy(STRATEGY_MEDIA)) { |
| device = getDeviceForStrategy(STRATEGY_MEDIA, fromCache); |
| } else if (outputDesc->isUsedByStrategy(STRATEGY_DTMF)) { |
| device = getDeviceForStrategy(STRATEGY_DTMF, fromCache); |
| } |
| |
| ALOGV("getNewDevice() selected device %x", device); |
| return device; |
| } |
| |
| AudioPolicyManager::routing_strategy AudioPolicyManager::getStrategy( |
| AudioSystem::stream_type stream) { |
| // stream to strategy mapping |
| switch (stream) { |
| case AudioSystem::VOICE_CALL: |
| case AudioSystem::BLUETOOTH_SCO: |
| return STRATEGY_PHONE; |
| case AudioSystem::RING: |
| case AudioSystem::ALARM: |
| return STRATEGY_SONIFICATION; |
| case AudioSystem::NOTIFICATION: |
| return STRATEGY_SONIFICATION_RESPECTFUL; |
| case AudioSystem::DTMF: |
| return STRATEGY_DTMF; |
| default: |
| ALOGE("unknown stream type"); |
| case AudioSystem::SYSTEM: |
| // NOTE: SYSTEM stream uses MEDIA strategy because muting music and switching outputs |
| // while key clicks are played produces a poor result |
| case AudioSystem::TTS: |
| case AudioSystem::MUSIC: |
| #ifdef QCOM_FM_ENABLED |
| case AudioSystem::FM: |
| #endif |
| return STRATEGY_MEDIA; |
| case AudioSystem::ENFORCED_AUDIBLE: |
| return STRATEGY_ENFORCED_AUDIBLE; |
| } |
| } |
| audio_devices_t AudioPolicyManager::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]; |
| } |
| |
| switch (strategy) { |
| |
| case STRATEGY_SONIFICATION_RESPECTFUL: |
| if (isInCall()) { |
| device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/); |
| } else if (isStreamActive(AudioSystem::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*/); |
| } |
| |
| 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: |
| // for phone strategy, we first consider the forced use and then the available devices by order |
| // of priority |
| switch (mForceUse[AudioSystem::FOR_COMMUNICATION]) { |
| case AudioSystem::FORCE_BT_SCO: |
| if (!isInCall() || strategy != STRATEGY_DTMF) { |
| device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT; |
| if (device) break; |
| } |
| device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET; |
| if (device) break; |
| device = mAvailableOutputDevices & 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()) |
| { |
| if ((mForceUse[AudioSystem::FOR_MEDIA] != AudioSystem::FORCE_NO_BT_A2DP) && |
| !mA2dpSuspended) { |
| device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP; |
| if (device) break; |
| device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES; |
| if (device) break; |
| } |
| } |
| device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_WIRED_HEADPHONE; |
| if (device) break; |
| device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_WIRED_HEADSET; |
| if (device) break; |
| #ifdef QCOM_ANC_HEADSET_ENABLED |
| device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_ANC_HEADPHONE; |
| if (device) break; |
| device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_ANC_HEADSET; |
| if (device) break; |
| #endif |
| if (mPhoneState != AudioSystem::MODE_IN_CALL) { |
| device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_ACCESSORY; |
| if (device) break; |
| device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_DEVICE; |
| if (device) break; |
| device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET; |
| if (device) break; |
| device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_AUX_DIGITAL; |
| if (device) break; |
| } |
| device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET; |
| if (device) break; |
| device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_EARPIECE; |
| if (device) break; |
| device = mDefaultOutputDevice; |
| if (device == AUDIO_DEVICE_NONE) { |
| ALOGE("getDeviceForStrategy() no device found for STRATEGY_PHONE"); |
| } |
| break; |
| |
| case AudioSystem::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[AudioSystem::FOR_MEDIA] != AudioSystem::FORCE_NO_BT_A2DP) && |
| !mA2dpSuspended) { |
| device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER; |
| if (device) break; |
| } |
| if (mPhoneState != AudioSystem::MODE_IN_CALL) { |
| device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_ACCESSORY; |
| if (device) break; |
| device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_DEVICE; |
| if (device) break; |
| device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET; |
| if (device) break; |
| device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_AUX_DIGITAL; |
| if (device) break; |
| } |
| device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET; |
| if (device) break; |
| device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_SPEAKER; |
| if (device) break; |
| device = mDefaultOutputDevice; |
| if (device == AUDIO_DEVICE_NONE) { |
| ALOGE("getDeviceForStrategy() no device found for STRATEGY_PHONE, FORCE_SPEAKER"); |
| } |
| break; |
| } |
| #ifdef QCOM_FM_ENABLED |
| if (mAvailableOutputDevices & AUDIO_DEVICE_OUT_FM) { |
| if (mForceUse[AudioSystem::FOR_MEDIA] == AudioSystem::FORCE_SPEAKER) { |
| device &= ~(AUDIO_DEVICE_OUT_WIRED_HEADSET); |
| device &= ~(AUDIO_DEVICE_OUT_WIRED_HEADPHONE); |
| device &= ~(AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET); |
| 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[AudioSystem::FOR_SYSTEM] == AudioSystem::FORCE_SYSTEM_ENFORCED)) { |
| device = mAvailableOutputDevices & 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 (mForceUse[AudioSystem::FOR_MEDIA] != AudioSystem::FORCE_SPEAKER) { |
| if (strategy != STRATEGY_SONIFICATION) { |
| #ifdef QCOM_PROXY_DEVICE_ENABLED |
| // no sonification on WFD sink |
| device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_PROXY; |
| #else |
| // no sonification on remote submix (e.g. WFD) |
| device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_REMOTE_SUBMIX; |
| #endif |
| } |
| if ((device2 == AUDIO_DEVICE_NONE) && |
| (mForceUse[AudioSystem::FOR_MEDIA] != AudioSystem::FORCE_NO_BT_A2DP) && |
| !mA2dpSuspended) { |
| if (device2 == AUDIO_DEVICE_NONE) { |
| device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP; |
| } |
| if (device2 == AUDIO_DEVICE_NONE) { |
| device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES; |
| } |
| if (device2 == AUDIO_DEVICE_NONE) { |
| device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER; |
| } |
| } |
| if (device2 == AUDIO_DEVICE_NONE) { |
| device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_WIRED_HEADPHONE; |
| } |
| if (device2 == AUDIO_DEVICE_NONE) { |
| device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_WIRED_HEADSET; |
| } |
| #ifdef QCOM_ANC_HEADSET_ENABLED |
| if (device2 == AUDIO_DEVICE_NONE) { |
| device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_ANC_HEADPHONE; |
| } |
| if (device2 == AUDIO_DEVICE_NONE) { |
| device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_ANC_HEADSET; |
| } |
| #endif |
| if (device2 == AUDIO_DEVICE_NONE) { |
| device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_ACCESSORY; |
| } |
| if (device2 == AUDIO_DEVICE_NONE) { |
| device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_DEVICE; |
| } |
| if (device2 == AUDIO_DEVICE_NONE) { |
| device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET; |
| } |
| if ((device2 == AUDIO_DEVICE_NONE) && (strategy != STRATEGY_SONIFICATION)) { |
| // no sonification on aux digital (e.g. HDMI) |
| device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_AUX_DIGITAL; |
| } |
| if (device2 == AUDIO_DEVICE_NONE) { |
| device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET; |
| } |
| #ifdef QCOM_FM_ENABLED |
| if (device2 == AUDIO_DEVICE_NONE) { |
| device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_FM_TX; |
| } |
| #endif |
| if (device2 == AUDIO_DEVICE_NONE) { |
| device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_SPEAKER; |
| } |
| |
| // device is DEVICE_OUT_SPEAKER if we come from case STRATEGY_SONIFICATION or |
| // STRATEGY_ENFORCED_AUDIBLE, AUDIO_DEVICE_NONE otherwise |
| device |= device2; |
| if (!device) { |
| device = mDefaultOutputDevice; |
| } |
| if (device == AUDIO_DEVICE_NONE) { |
| ALOGE("getDeviceForStrategy() no device found for STRATEGY_MEDIA"); |
| } |
| |
| } else { |
| //AudioSystem::FORCE_SPEAKER for STRATEGY_MEDIA |
| device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_SPEAKER; |
| } |
| |
| if (isInCall()) { |
| // when in call, get the device for Phone strategy |
| device = getDeviceForStrategy(STRATEGY_PHONE, false /*fromCache*/); |
| } |
| |
| } break; |
| |
| default: |
| ALOGW("getDeviceForStrategy() unknown strategy: %d", strategy); |
| break; |
| } |
| |
| ALOGVV("getDeviceForStrategy() strategy %d, device %x", strategy, device); |
| return device; |
| } |
| |
| uint32_t AudioPolicyManager::setOutputDevice(audio_io_handle_t output, |
| audio_devices_t device, |
| bool force, |
| int delayMs) |
| |
| { |
| ALOGV("setOutputDevice() output %d device %04x delayMs %d", output, device, delayMs); |
| AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output); |
| AudioParameter param; |
| uint32_t muteWaitMs = 0; |
| |
| if (outputDesc->isDuplicated()) { |
| muteWaitMs = setOutputDevice(outputDesc->mOutput1->mId, device, force, delayMs); |
| muteWaitMs += setOutputDevice(outputDesc->mOutput2->mId, device, force, delayMs); |
| return muteWaitMs; |
| } |
| // filter devices according to output selected |
| device = (audio_devices_t)(device & outputDesc->mProfile->mSupportedDevices); |
| |
| audio_devices_t prevDevice = outputDesc->mDevice; |
| |
| ALOGV("setOutputDevice() prevDevice %04x", prevDevice); |
| |
| if (device != AUDIO_DEVICE_NONE) { |
| outputDesc->mDevice = device; |
| } |
| muteWaitMs = checkDeviceMuteStrategies(outputDesc, prevDevice, delayMs); |
| |
| // Do not change the routing if: |
| // - the requested device is AUDIO_DEVICE_NONE |
| // - the requested device is the same as current device and force is not specified. |
| // Doing this check here allows the caller to call setOutputDevice() without conditions |
| if ((device == AUDIO_DEVICE_NONE) || ((device == prevDevice) && !force)) { |
| ALOGV("setOutputDevice() setting same device %04x or null device for output %d", device, output); |
| return muteWaitMs; |
| } |
| if (device == prevDevice) { |
| ALOGV("setOutputDevice() Call routing with same device with zero delay "); |
| delayMs = 0; |
| } |
| ALOGV("setOutputDevice() changing device:%x",device); |
| // do the routing |
| param.addInt(String8(AudioParameter::keyRouting), (int)device); |
| mpClientInterface->setParameters(output, param.toString(), delayMs); |
| |
| // update stream volumes according to new device |
| applyStreamVolumes(output, device, delayMs); |
| |
| return muteWaitMs; |
| } |
| |
| audio_devices_t AudioPolicyManager::getDeviceForInputSource(int inputSource) |
| { |
| uint32_t device = AUDIO_DEVICE_NONE; |
| |
| switch(inputSource) { |
| case AUDIO_SOURCE_DEFAULT: |
| case AUDIO_SOURCE_MIC: |
| case AUDIO_SOURCE_VOICE_RECOGNITION: |
| if (mForceUse[AudioSystem::FOR_RECORD] == AudioSystem::FORCE_BT_SCO && |
| mAvailableInputDevices & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) { |
| device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET; |
| } else if (mAvailableInputDevices & AUDIO_DEVICE_IN_WIRED_HEADSET) { |
| device = AUDIO_DEVICE_IN_WIRED_HEADSET; |
| #ifdef QCOM_ANC_HEADSET_ENABLED |
| } else if (mAvailableInputDevices & AUDIO_DEVICE_IN_ANC_HEADSET) { |
| device = AUDIO_DEVICE_IN_ANC_HEADSET; |
| #endif |
| } else if (mAvailableInputDevices & AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET) { |
| device = AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET; |
| } else if (mAvailableInputDevices & AUDIO_DEVICE_IN_BUILTIN_MIC) { |
| device = AUDIO_DEVICE_IN_BUILTIN_MIC; |
| } |
| break; |
| case AUDIO_SOURCE_VOICE_COMMUNICATION: |
| device = AUDIO_DEVICE_IN_COMMUNICATION; |
| break; |
| case AUDIO_SOURCE_CAMCORDER: |
| if (mAvailableInputDevices & AUDIO_DEVICE_IN_BACK_MIC) { |
| device = AUDIO_DEVICE_IN_BACK_MIC; |
| } else if (mAvailableInputDevices & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) { |
| device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET; |
| } else if (mAvailableInputDevices & AUDIO_DEVICE_IN_WIRED_HEADSET) { |
| device = AUDIO_DEVICE_IN_WIRED_HEADSET; |
| } else if (mAvailableInputDevices & AUDIO_DEVICE_IN_AUX_DIGITAL) { |
| device = AUDIO_DEVICE_IN_AUX_DIGITAL; |
| } else if (mAvailableInputDevices & AUDIO_DEVICE_IN_BUILTIN_MIC) { |
| device = AUDIO_DEVICE_IN_BUILTIN_MIC; |
| } |
| break; |
| case AUDIO_SOURCE_VOICE_UPLINK: |
| case AUDIO_SOURCE_VOICE_DOWNLINK: |
| case AUDIO_SOURCE_VOICE_CALL: |
| if (mAvailableInputDevices & AUDIO_DEVICE_IN_VOICE_CALL) { |
| device = AUDIO_DEVICE_IN_VOICE_CALL; |
| } |
| break; |
| case AUDIO_SOURCE_REMOTE_SUBMIX: |
| if (mAvailableInputDevices & AUDIO_DEVICE_IN_REMOTE_SUBMIX) { |
| device = AUDIO_DEVICE_IN_REMOTE_SUBMIX; |
| } |
| break; |
| #ifdef QCOM_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; |
| } |
| |
| |
| |
| |
| audio_devices_t AudioPolicyManager::getDeviceForVolume(audio_devices_t device) |
| { |
| if (device == AUDIO_DEVICE_NONE) { |
| // this happens when forcing a route update and no track is active on an output. |
| // In this case the returned category is not important. |
| device = AUDIO_DEVICE_OUT_SPEAKER; |
| } else if (AudioSystem::popCount(device) > 1) { |
| // Multiple device selection is either: |
| // - speaker + one other device: give priority to speaker in this case. |
| // - one A2DP device + another device: happens with duplicated output. In this case |
| // retain the device on the A2DP output as the other must not correspond to an active |
| // selection if not the speaker. |
| if (device & AUDIO_DEVICE_OUT_SPEAKER) { |
| device = AUDIO_DEVICE_OUT_SPEAKER; |
| } else if ((device & AUDIO_DEVICE_OUT_WIRED_HEADSET) != 0) { |
| device = AUDIO_DEVICE_OUT_WIRED_HEADSET; |
| } else if ((device & AUDIO_DEVICE_OUT_WIRED_HEADPHONE) != 0) { |
| device = AUDIO_DEVICE_OUT_WIRED_HEADPHONE; |
| } else { |
| device = (audio_devices_t)(device & AUDIO_DEVICE_OUT_ALL_A2DP); |
| } |
| } |
| |
| ALOGW_IF(AudioSystem::popCount(device) != 1, |
| "getDeviceForVolume() invalid device combination: %08x", |
| device); |
| |
| return device; |
| } |
| |
| AudioPolicyManager::device_category AudioPolicyManager::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_ANC_HEADSET: |
| case AUDIO_DEVICE_OUT_ANC_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 QCOM_FM_ENABLED |
| case AUDIO_DEVICE_OUT_FM: |
| #endif |
| return DEVICE_CATEGORY_HEADSET; |
| case AUDIO_DEVICE_OUT_SPEAKER: |
| case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT: |
| case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER: |
| case AUDIO_DEVICE_OUT_AUX_DIGITAL: |
| case AUDIO_DEVICE_OUT_USB_ACCESSORY: |
| case AUDIO_DEVICE_OUT_USB_DEVICE: |
| case AUDIO_DEVICE_OUT_REMOTE_SUBMIX: |
| case AUDIO_DEVICE_OUT_PROXY: |
| default: |
| return DEVICE_CATEGORY_SPEAKER; |
| } |
| } |
| |
| bool AudioPolicyManager::isDirectOutput(audio_io_handle_t output) { |
| for (size_t i = 0; i < mOutputs.size(); i++) { |
| audio_io_handle_t curOutput = mOutputs.keyAt(i); |
| AudioOutputDescriptor *desc = mOutputs.valueAt(i); |
| if ((curOutput == output) && (desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { |
| return true; |
| } |
| } |
| return false; |
| } |
| |
| status_t AudioPolicyManager::checkAndSetVolume(int 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 == AudioSystem::VOICE_CALL && mForceUse[AudioSystem::FOR_COMMUNICATION] == AudioSystem::FORCE_BT_SCO) || |
| (stream == AudioSystem::BLUETOOTH_SCO && mForceUse[AudioSystem::FOR_COMMUNICATION] != AudioSystem::FORCE_BT_SCO)) { |
| ALOGV("checkAndSetVolume() cannot set stream %d volume with force use = %d for comm", |
| stream, mForceUse[AudioSystem::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] || |
| #ifdef QCOM_FM_ENABLED |
| (stream == AudioSystem::FM) || |
| #endif |
| 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 == AudioSystem::BLUETOOTH_SCO) { |
| mpClientInterface->setStreamVolume(AudioSystem::VOICE_CALL, volume, output, delayMs); |
| #ifdef QCOM_FM_ENABLED |
| } else if (stream == AudioSystem::FM) { |
| float fmVolume = -1.0; |
| fmVolume = computeVolume(stream, index, output, device); |
| if (fmVolume >= 0) { |
| if(output == mPrimaryOutput) |
| mpClientInterface->setFmVolume(fmVolume, delayMs); |
| else if(mHasA2dp && output == getA2dpOutput()) |
| mpClientInterface->setStreamVolume((AudioSystem::stream_type)stream, volume, output, delayMs); |
| } |
| return NO_ERROR; |
| #endif |
| } |
| mpClientInterface->setStreamVolume((AudioSystem::stream_type)stream, volume, output, delayMs); |
| } |
| |
| if (stream == AudioSystem::VOICE_CALL || |
| stream == AudioSystem::BLUETOOTH_SCO) { |
| float voiceVolume; |
| // Force voice volume to max for bluetooth SCO as volume is managed by the headset |
| if (stream == AudioSystem::VOICE_CALL) { |
| voiceVolume = (float)index/(float)mStreams[stream].mIndexMax; |
| } else { |
| voiceVolume = 1.0; |
| } |
| |
| voiceVolume = (float)index/(float)mStreams[stream].mIndexMax; |
| |
| // Force voice volume to max when Vgs is set for bluetooth SCO as volume is managed by the headset |
| if (stream == AudioSystem::BLUETOOTH_SCO) { |
| String8 key ("bt_headset_vgs"); |
| mpClientInterface->getParameters(output,key); |
| AudioParameter result(mpClientInterface->getParameters(0,key)); |
| int value; |
| if (result.getInt(String8("isVGS"),value) == NO_ERROR) { |
| ALOGV("Use BT-SCO Voice Volume"); |
| voiceVolume = 1.0; |
| } |
| } |
| |
| if (voiceVolume != mLastVoiceVolume && (output == mPrimaryOutput || |
| isDirectOutput(output))) { |
| mpClientInterface->setVoiceVolume(voiceVolume, delayMs); |
| mLastVoiceVolume = voiceVolume; |
| } |
| } |
| |
| return NO_ERROR; |
| } |
| |
| |
| void AudioPolicyManager::checkA2dpSuspend() |
| { |
| |
| // suspend A2DP output if: |
| // (NOT already suspended) && |
| // ((SCO device is connected && |
| // (forced usage for communication || for record is SCO))) || |
| // (phone state is ringing || in call) |
| // |
| // restore A2DP output if: |
| // (Already suspended) && |
| // ((SCO device is NOT connected || |
| // (forced usage NOT for communication && NOT for record is SCO))) && |
| // (phone state is NOT ringing && NOT in call) |
| // |
| if (mA2dpSuspended) { |
| if (((mScoDeviceAddress == "") || |
| ((mForceUse[AudioSystem::FOR_COMMUNICATION] != AudioSystem::FORCE_BT_SCO) && |
| (mForceUse[AudioSystem::FOR_RECORD] != AudioSystem::FORCE_BT_SCO))) || |
| ((mPhoneState != AudioSystem::MODE_IN_CALL) && |
| (mPhoneState != AudioSystem::MODE_RINGTONE))) { |
| |
| mA2dpSuspended = false; |
| } |
| } else { |
| if (((mScoDeviceAddress != "") && |
| ((mForceUse[AudioSystem::FOR_COMMUNICATION] == AudioSystem::FORCE_BT_SCO) || |
| (mForceUse[AudioSystem::FOR_RECORD] == AudioSystem::FORCE_BT_SCO))) && |
| ((mPhoneState == AudioSystem::MODE_IN_CALL) || |
| (mPhoneState == AudioSystem::MODE_RINGTONE))) { |
| |
| mA2dpSuspended = true; |
| } |
| } |
| } |
| |
| audio_io_handle_t AudioPolicyManager::getA2dpOutput() |
| { |
| return 0; |
| } |
| |
| //private function, no changes from AudioPolicyManagerBase |
| void AudioPolicyManager::handleNotificationRoutingForStream(AudioSystem::stream_type stream) { |
| switch(stream) { |
| case AudioSystem::MUSIC: |
| checkOutputForStrategy(STRATEGY_SONIFICATION_RESPECTFUL); |
| updateDevicesAndOutputs(); |
| break; |
| default: |
| break; |
| } |
| } |
| |
| |
| AudioPolicyManagerBase::IOProfile *AudioPolicyManager::getProfileForDirectOutput( |
| audio_devices_t device, |
| uint32_t samplingRate, |
| uint32_t format, |
| uint32_t channelMask, |
| audio_output_flags_t flags) |
| { |
| if( !((flags & AUDIO_OUTPUT_FLAG_LPA) || |
| (flags & AUDIO_OUTPUT_FLAG_TUNNEL)|| |
| (flags & AUDIO_OUTPUT_FLAG_VOIP_RX)) ) |
| flags = AUDIO_OUTPUT_FLAG_DIRECT; |
| |
| for (size_t i = 0; i < mHwModules.size(); i++) { |
| if (mHwModules[i]->mHandle == 0) { |
| continue; |
| } |
| for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++) { |
| AudioPolicyManagerBase::IOProfile *profile = mHwModules[i]->mOutputProfiles[j]; |
| if (isCompatibleProfile(profile, device, samplingRate, format, |
| channelMask, |
| flags)) { |
| if (mAvailableOutputDevices & profile->mSupportedDevices) { |
| return mHwModules[i]->mOutputProfiles[j]; |
| } |
| } |
| } |
| } |
| return 0; |
| } |
| |
| bool AudioPolicyManager::isCompatibleProfile(AudioPolicyManagerBase::IOProfile *profile, |
| audio_devices_t device, |
| uint32_t samplingRate, |
| uint32_t format, |
| uint32_t channelMask, |
| audio_output_flags_t flags) |
| { |
| if ((profile->mSupportedDevices & device) != device) { |
| return false; |
| } |
| if (profile->mFlags != flags) { |
| return false; |
| } |
| if (samplingRate != 0) { |
| size_t i; |
| for (i = 0; i < profile->mSamplingRates.size(); i++) |
| { |
| if (profile->mSamplingRates[i] == samplingRate) { |
| break; |
| } |
| } |
| if (i == profile->mSamplingRates.size()) { |
| return false; |
| } |
| } |
| if (format != 0) { |
| size_t i; |
| for (i = 0; i < profile->mFormats.size(); i++) |
| { |
| if (profile->mFormats[i] == format) { |
| break; |
| } |
| } |
| if (i == profile->mFormats.size()) { |
| return false; |
| } |
| } |
| if (channelMask != 0) { |
| size_t i; |
| for (i = 0; i < profile->mChannelMasks.size(); i++) |
| { |
| if (profile->mChannelMasks[i] == channelMask) { |
| break; |
| } |
| } |
| if (i == profile->mChannelMasks.size()) { |
| return false; |
| } |
| } |
| ALOGD(" profile found: device %x, flags %x, samplingrate %d,\ |
| format %x, channelMask %d", |
| device, flags, samplingRate, format, channelMask); |
| return true; |
| } |
| |
| bool AudioPolicyManager::platform_is_Fusion3() |
| { |
| char platform[128], baseband[128]; |
| property_get("ro.board.platform", platform, ""); |
| property_get("ro.baseband", baseband, ""); |
| if (!strcmp("msm8960", platform) && !strcmp("mdm", baseband)) |
| return true; |
| else |
| return false; |
| } |
| |
| extern "C" AudioPolicyInterface* createAudioPolicyManager(AudioPolicyClientInterface *clientInterface) |
| { |
| return new AudioPolicyManager(clientInterface); |
| } |
| |
| extern "C" void destroyAudioPolicyManager(AudioPolicyInterface *interface) |
| { |
| delete interface; |
| } |
| |
| |
| }; // namespace android |