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Commit 34f30b43 authored by Ramjee Singh's avatar Ramjee Singh
Browse files

policy-hal: enable custom audio policy. 

This change enable custom audio policy and also includes
the fixes that went in to audio policy manager

audio policy: add support for external audio sources
Change-Id: If5805d58a4356b2f681f1aabf54375f62b55b98a

Add setMasterMono and getMasterMono
Change-Id: I32dc1fcecf285967a61bd508af3bb299595db57d

AudioPolicy: AudioOutputDescriptor set/get for patch handle
Change-Id: I5bffecf7a157ec042d749064208bc6c279008542

AudioPolicyManager: force encoded surround setting
Change-Id: I0f7a486bf1aa7a27f15cf220e2c75ef23f8f8536

audio: several fixes in audio routing callbacks
Change-Id: Ieb0d9f92f563a40552eb31bc0499c8ac65f78ce4

audio policy: move volume table to XML file
Change-Id: I78c91bc3378b6c47202abe7b5c1c1e011ff62eed

AudioPolicy: capture sample rate 0 is route-specific default
Change-Id: I7d193640572c08ab5e5f1ecfc8ad6d31635967dd

audio policy: remove deserialization from policy objects
Change-Id: I1ce798848f4657b37e47446c9fbdc63f7ed0390e

DO NOT MERGE - audio policy: disable concurrent capture
Change-Id: I9085918f1f7b12e28430958cceb768bb29c12bdb

audio policy: select output with best sample format match
Change-Id: I31981940be5d1c06d713d52187cfb0a7bf355773

audio policy: audio sessions on input descriptors
Change-Id: I915a65989a7fd0d3cbe2fcf5a0aee2ea0df5f4f5

Call SoundTrigger::setCaptureState in AudioPoilcyManager::closeAllInputs
Change-Id: I613434ecacce3ef03a657008fd94c314948ba2b8

Change-Id: Iee8ef2647691fe1167b95f19357f17c28f54bf0e
parent efd1c719

policy_hal/AudioPolicyManager.cpp

+107 −61
Original line number Diff line number Diff line
@@ -116,7 +116,7 @@ status_t AudioPolicyManagerCustom::setDeviceConnectionStateInt(audio_devices_t d 
                                                         const char *device_address,

                                                         const char *device_name)

{

    ALOGV("setDeviceConnectionStateInt() device: 0x%X, state %d, address %s name %s",

    ALOGD("setDeviceConnectionStateInt() device: 0x%X, state %d, address %s name %s",

            device, state, device_address, device_name);



    // connect/disconnect only 1 device at a time
@@ -314,6 +314,10 @@ status_t AudioPolicyManagerCustom::setDeviceConnectionStateInt(audio_devices_t d 
            }

        }



        if (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) {

            cleanUpForDevice(devDesc);

        }



        mpClientInterface->onAudioPortListUpdate();

        return NO_ERROR;

    }  // end if is output device
@@ -390,6 +394,10 @@ status_t AudioPolicyManagerCustom::setDeviceConnectionStateInt(audio_devices_t d 
            updateCallRouting(newDevice);

        }



        if (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) {

            cleanUpForDevice(devDesc);

        }



        mpClientInterface->onAudioPortListUpdate();

        return NO_ERROR;

    } // end if is input device
@@ -409,6 +417,11 @@ bool AudioPolicyManagerCustom::isOffloadSupported(const audio_offload_info_t& of 
     offloadInfo.format,

     offloadInfo.stream_type, offloadInfo.bit_rate, offloadInfo.duration_us,

     offloadInfo.has_video);



     if (mMasterMono) {

        return false; // no offloading if mono is set.

     }



#ifdef VOICE_CONCURRENCY

    char concpropValue[PROPERTY_VALUE_MAX];

    if (property_get("voice.playback.conc.disabled", concpropValue, NULL)) {
@@ -575,12 +588,12 @@ audio_devices_t AudioPolicyManagerCustom::getNewOutputDevice(const sp<AudioOutpu 
{

    audio_devices_t device = AUDIO_DEVICE_NONE;



    ssize_t index = mAudioPatches.indexOfKey(outputDesc->mPatchHandle);

    ssize_t index = mAudioPatches.indexOfKey(outputDesc->getPatchHandle());

    if (index >= 0) {

        sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);

        if (patchDesc->mUid != mUidCached) {

            ALOGV("getNewOutputDevice() device %08x forced by patch %d",

                  outputDesc->device(), outputDesc->mPatchHandle);

                  outputDesc->device(), outputDesc->getPatchHandle());

            return outputDesc->device();

        }

    }
@@ -638,7 +651,7 @@ audio_devices_t AudioPolicyManagerCustom::getNewOutputDevice(const sp<AudioOutpu 


void AudioPolicyManagerCustom::setPhoneState(audio_mode_t state)

{

    ALOGV("setPhoneState() state %d", state);

    ALOGD("setPhoneState() state %d", state);

    // store previous phone state for management of sonification strategy below

    audio_devices_t newDevice = AUDIO_DEVICE_NONE;

    int oldState = mEngine->getPhoneState();
@@ -714,25 +727,29 @@ void AudioPolicyManagerCustom::setPhoneState(audio_mode_t state) 
            audio_io_handle_t activeInput = mInputs.getActiveInput();

            if (activeInput != 0) {

               sp<AudioInputDescriptor> activeDesc = mInputs.valueFor(activeInput);

               switch(activeDesc->mInputSource) {

               switch(activeDesc->inputSource()) {

                   case AUDIO_SOURCE_VOICE_UPLINK:

                   case AUDIO_SOURCE_VOICE_DOWNLINK:

                   case AUDIO_SOURCE_VOICE_CALL:

                       ALOGD("voice_conc:FOUND active input during call active: %d",activeDesc->mInputSource);

                       ALOGD("voice_conc:FOUND active input during call active: %d",activeDesc->inputSource());

                   break;



                   case  AUDIO_SOURCE_VOICE_COMMUNICATION:

                        if(prop_voip_enabled) {

                            ALOGD("voice_conc:CLOSING VoIP input source on call setup :%d ",activeDesc->mInputSource);

                            stopInput(activeInput, activeDesc->mSessions.itemAt(0));

                            releaseInput(activeInput, activeDesc->mSessions.itemAt(0));

                            ALOGD("voice_conc:CLOSING VoIP input source on call setup :%d ",activeDesc->inputSource());

                            AudioSessionCollection activeSessions = activeDesc->getActiveAudioSessions();

                            audio_session_t activeSession = activeSessions.keyAt(0);

                            stopInput(activeInput, activeSession);

                            releaseInput(activeInput, activeSession);

                        }

                   break;



                   default:

                       ALOGD("voice_conc:CLOSING input on call setup  for inputSource: %d",activeDesc->mInputSource);

                       stopInput(activeInput, activeDesc->mSessions.itemAt(0));

                       releaseInput(activeInput, activeDesc->mSessions.itemAt(0));

                       ALOGD("voice_conc:CLOSING input on call setup  for inputSource: %d",activeDesc->inputSource());

                       AudioSessionCollection activeSessions = activeDesc->getActiveAudioSessions();

                       audio_session_t activeSession = activeSessions.keyAt(0);

                       stopInput(activeInput, activeSession);

                       releaseInput(activeInput, activeSession);

                   break;

               }

           }
@@ -740,10 +757,12 @@ void AudioPolicyManagerCustom::setPhoneState(audio_mode_t state) 
            audio_io_handle_t activeInput = mInputs.getActiveInput();

            if (activeInput != 0) {

                sp<AudioInputDescriptor> activeDesc = mInputs.valueFor(activeInput);

                if (AUDIO_SOURCE_VOICE_COMMUNICATION == activeDesc->mInputSource) {

                    ALOGD("voice_conc:CLOSING VoIP on call setup : %d",activeDesc->mInputSource);

                    stopInput(activeInput, activeDesc->mSessions.itemAt(0));

                    releaseInput(activeInput, activeDesc->mSessions.itemAt(0));

                if (AUDIO_SOURCE_VOICE_COMMUNICATION == activeDesc->inputSource()) {

                    ALOGD("voice_conc:CLOSING VoIP on call setup : %d",activeDesc->inputSource());

                    AudioSessionCollection activeSessions = activeDesc->getActiveAudioSessions();

                    audio_session_t activeSession = activeSessions.keyAt(0);

                    stopInput(activeInput, activeSession);

                    releaseInput(activeInput, activeSession);

                }

            }

        }
@@ -776,23 +795,23 @@ void AudioPolicyManagerCustom::setPhoneState(audio_mode_t state) 
            }



            if (AUDIO_OUTPUT_FLAG_FAST == mFallBackflag) {

                if (((!outputDesc->isDuplicated() &&outputDesc->mProfile->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY))

                if (((!outputDesc->isDuplicated() &&outputDesc->mProfile->getFlags() & AUDIO_OUTPUT_FLAG_PRIMARY))

                            && prop_playback_enabled) {

                    ALOGD("voice_conc:calling suspendOutput on call mode for primary output");

                    mpClientInterface->suspendOutput(mOutputs.keyAt(i));

                } //Close compress all sessions

                else if ((outputDesc->mProfile->mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)

                else if ((outputDesc->mProfile->getFlags() & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)

                                &&  prop_playback_enabled) {

                    ALOGD("voice_conc:calling closeOutput on call mode for COMPRESS output");

                    closeOutput(mOutputs.keyAt(i));

                }

                else if ((outputDesc->mProfile->mFlags & AUDIO_OUTPUT_FLAG_VOIP_RX)

                else if ((outputDesc->mProfile->getFlags() & AUDIO_OUTPUT_FLAG_VOIP_RX)

                                && prop_voip_enabled) {

                    ALOGD("voice_conc:calling closeOutput on call mode for DIRECT  output");

                    closeOutput(mOutputs.keyAt(i));

                }

            } else if (AUDIO_OUTPUT_FLAG_DEEP_BUFFER == mFallBackflag) {

                if ((outputDesc->mProfile->mFlags & AUDIO_OUTPUT_FLAG_DIRECT)

                if ((outputDesc->mProfile->getFlags() & AUDIO_OUTPUT_FLAG_DIRECT)

                                &&  prop_playback_enabled) {

                    ALOGD("voice_conc:calling closeOutput on call mode for COMPRESS output");

                    closeOutput(mOutputs.keyAt(i));
@@ -813,7 +832,7 @@ void AudioPolicyManagerCustom::setPhoneState(audio_mode_t state) 
                   ALOGD("voice_conc:ouput desc / profile is NULL");

                   continue;

                }

                if (!outputDesc->isDuplicated() && outputDesc->mProfile->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY) {

                if (!outputDesc->isDuplicated() && outputDesc->mProfile->getFlags() & AUDIO_OUTPUT_FLAG_PRIMARY) {

                    ALOGD("voice_conc:calling restoreOutput after call mode for primary output");

                    mpClientInterface->restoreOutput(mOutputs.keyAt(i));

                }
@@ -862,7 +881,7 @@ void AudioPolicyManagerCustom::setPhoneState(audio_mode_t state) 
                   continue;

                }



                if (outputDesc->mProfile->mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {

                if (outputDesc->mProfile->getFlags() & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {

                    ALOGD("calling closeOutput on call mode for COMPRESS output");

                    closeOutput(mOutputs.keyAt(i));

                }
@@ -1043,7 +1062,7 @@ status_t AudioPolicyManagerCustom::startSource(sp<AudioOutputDescriptor> outputD 
    *delayMs = 0;

    if (stream == AUDIO_STREAM_TTS) {

        ALOGV("\t found BEACON stream");

        if (mOutputs.isAnyOutputActive(AUDIO_STREAM_TTS /*streamToIgnore*/)) {

        if (!mTtsOutputAvailable && mOutputs.isAnyOutputActive(AUDIO_STREAM_TTS /*streamToIgnore*/)) {

            return INVALID_OPERATION;

        } else {

            beaconMuteLatency = handleEventForBeacon(STARTING_BEACON);
@@ -1053,6 +1072,9 @@ status_t AudioPolicyManagerCustom::startSource(sp<AudioOutputDescriptor> outputD 
        beaconMuteLatency = handleEventForBeacon(STARTING_OUTPUT);

    }



    // check active before incrementing usage count

    bool force = !outputDesc->isActive();



    // 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()
@@ -1068,7 +1090,6 @@ status_t AudioPolicyManagerCustom::startSource(sp<AudioOutputDescriptor> outputD 
                            (strategy == STRATEGY_SONIFICATION_RESPECTFUL) ||

                            (beaconMuteLatency > 0);

        uint32_t waitMs = beaconMuteLatency;

        bool force = false;

        for (size_t i = 0; i < mOutputs.size(); i++) {

            sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i);

            if (desc != outputDesc) {
@@ -1089,7 +1110,8 @@ status_t AudioPolicyManagerCustom::startSource(sp<AudioOutputDescriptor> outputD 
                }

            }

        }

        uint32_t muteWaitMs = setOutputDevice(outputDesc, device, force);

        uint32_t muteWaitMs;

        muteWaitMs = setOutputDevice(outputDesc, device, force);



        // handle special case for sonification while in call

        if (isInCall()) {
@@ -1098,7 +1120,7 @@ status_t AudioPolicyManagerCustom::startSource(sp<AudioOutputDescriptor> outputD 


        // apply volume rules for current stream and device if necessary

        checkAndSetVolume(stream,

                          mStreams.valueFor(stream).getVolumeIndex(device),

                          mVolumeCurves->getVolumeIndex(stream, device),

                          outputDesc,

                          device);
@@ -1226,7 +1248,7 @@ status_t AudioPolicyManagerCustom::checkAndSetVolume(audio_stream_type_t stream, 
        float voiceVolume;

        // Force voice volume to max for bluetooth SCO as volume is managed by the headset

        if (stream == AUDIO_STREAM_VOICE_CALL) {

            voiceVolume = (float)index/(float)mStreams.valueFor(stream).getVolumeIndexMax();

            voiceVolume = (float)index/(float)mVolumeCurves->getVolumeIndexMax(stream);

        } else {

            voiceVolume = 1.0;

        }
@@ -1314,7 +1336,6 @@ audio_io_handle_t AudioPolicyManagerCustom::getOutputForDevice( 
        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
@@ -1558,7 +1579,7 @@ audio_io_handle_t AudioPolicyManagerCustom::getOutputForDevice( 
    // skip direct output selection if the request can obviously be attached to a mixed output

    // and not explicitly requested

    if (((flags & (AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_DIRECT_PCM)) == 0) &&

            audio_is_linear_pcm(format) && samplingRate <= MAX_MIXER_SAMPLING_RATE &&

            audio_is_linear_pcm(format) && samplingRate <= SAMPLE_RATE_HZ_MAX &&

            audio_channel_count_from_out_mask(channelMask) <= 2) {

        goto non_direct_output;

    }
@@ -1571,7 +1592,7 @@ audio_io_handle_t AudioPolicyManagerCustom::getOutputForDevice( 
    // in the background.



    if (((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) == 0) ||

            !mEffects.isNonOffloadableEffectEnabled()) {

            !(mEffects.isNonOffloadableEffectEnabled() || mMasterMono)) {

        profile = getProfileForDirectOutput(device,

                                           samplingRate,

                                           format,
@@ -1582,7 +1603,7 @@ audio_io_handle_t AudioPolicyManagerCustom::getOutputForDevice( 
    if (profile != 0) {



        if (!(flags & AUDIO_OUTPUT_FLAG_DIRECT_PCM) &&

             (profile->mFlags & AUDIO_OUTPUT_FLAG_DIRECT_PCM)) {

             (profile->getFlags() & AUDIO_OUTPUT_FLAG_DIRECT_PCM)) {

            ALOGI("got Direct_PCM without requesting ... reject ");

            profile = NULL;

            goto non_direct_output;
@@ -1596,7 +1617,7 @@ audio_io_handle_t AudioPolicyManagerCustom::getOutputForDevice( 
                outputDesc = desc;

                // reuse direct output if currently open and configured with same parameters

                if ((samplingRate == outputDesc->mSamplingRate) &&

                        (format == outputDesc->mFormat) &&

                        audio_formats_match(format, outputDesc->mFormat) &&

                        (channelMask == outputDesc->mChannelMask)) {

                    outputDesc->mDirectOpenCount++;

                    ALOGV("getOutput() reusing direct output %d", mOutputs.keyAt(i));
@@ -1612,7 +1633,7 @@ audio_io_handle_t AudioPolicyManagerCustom::getOutputForDevice( 
        // if the selected profile is offloaded and no offload info was specified,

        // create a default one

        audio_offload_info_t defaultOffloadInfo = AUDIO_INFO_INITIALIZER;

        if ((profile->mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) && !offloadInfo) {

        if ((profile->getFlags() & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) && !offloadInfo) {

            flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD);

            defaultOffloadInfo.sample_rate = samplingRate;

            defaultOffloadInfo.channel_mask = channelMask;
@@ -1647,7 +1668,7 @@ audio_io_handle_t AudioPolicyManagerCustom::getOutputForDevice( 
        // 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) ||

            (format != AUDIO_FORMAT_DEFAULT && !audio_formats_match(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,
@@ -1657,7 +1678,7 @@ audio_io_handle_t AudioPolicyManagerCustom::getOutputForDevice( 
                mpClientInterface->closeOutput(output);

            }

            // fall back to mixer output if possible when the direct output could not be open

            if (audio_is_linear_pcm(format) && samplingRate <= MAX_MIXER_SAMPLING_RATE) {

            if (audio_is_linear_pcm(format) && samplingRate <= SAMPLE_RATE_HZ_MAX) {

                goto non_direct_output;

            }

            return AUDIO_IO_HANDLE_NONE;
@@ -1735,7 +1756,8 @@ status_t AudioPolicyManagerCustom::getInputForAttr(const audio_attributes_t *att 
                                             audio_port_handle_t selectedDeviceId,

                                             input_type_t *inputType)

{

    audio_source_t inputSource = attr->source;

    audio_source_t inputSource;

    inputSource = attr->source;

#ifdef VOICE_CONCURRENCY



    char propValue[PROPERTY_VALUE_MAX];
@@ -1819,8 +1841,8 @@ status_t AudioPolicyManagerCustom::startInput(audio_io_handle_t input, 
    }

    sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(index);



    index = inputDesc->mSessions.indexOf(session);

    if (index < 0) {

    sp<AudioSession> audioSession = inputDesc->getAudioSession(session);

    if (audioSession == 0) {

        ALOGW("startInput() unknown session %d on input %d", session, input);

        return BAD_VALUE;

    }
@@ -1835,15 +1857,31 @@ status_t AudioPolicyManagerCustom::startInput(audio_io_handle_t 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) {

            if ((activeDesc->inputSource() == AUDIO_SOURCE_HOTWORD) &&

                    !activeDesc->hasPreemptedSession(session)) {

                ALOGW("startInput(%d) preempting low-priority input %d", input, activeInput);

                stopInput(activeInput, activeDesc->mSessions.itemAt(0));

                releaseInput(activeInput, activeDesc->mSessions.itemAt(0));

                //FIXME: consider all active sessions

                AudioSessionCollection activeSessions = activeDesc->getActiveAudioSessions();

                audio_session_t activeSession = activeSessions.keyAt(0);

                SortedVector<audio_session_t> sessions =

                                           activeDesc->getPreemptedSessions();

                sessions.add(activeSession);

                inputDesc->setPreemptedSessions(sessions);

                stopInput(activeInput, activeSession);

                releaseInput(activeInput, activeSession);

            } else {

                ALOGE("startInput(%d) failed: other input %d already started", input, activeInput);

                return INVALID_OPERATION;

            }

        }

        // Do not allow capture if an active voice call is using a software patch and

        // the call TX source device is on the same HW module.

        // FIXME: would be better to refine to only inputs whose profile connects to the

        // call TX device but this information is not in the audio patch

        if (mCallTxPatch != 0 &&

            inputDesc->getModuleHandle() == mCallTxPatch->mPatch.sources[0].ext.device.hw_module) {

            return INVALID_OPERATION;

        }

    }



    // Routing?
@@ -1882,7 +1920,7 @@ status_t AudioPolicyManagerCustom::startInput(audio_io_handle_t input, 
               ALOGD("ouput desc / profile is NULL");

               continue;

            }

            if (outputDesc->mProfile->mFlags

            if (outputDesc->mProfile->getFlags()

                            & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {

                // close compress  sessions

                ALOGD("calling closeOutput on record conc for COMPRESS output");
@@ -1892,7 +1930,7 @@ status_t AudioPolicyManagerCustom::startInput(audio_io_handle_t input, 
    }

#endif



    if (inputDesc->mRefCount == 0 || mInputRoutes.hasRouteChanged(session)) {

    if (!inputDesc->isActive() || mInputRoutes.hasRouteChanged(session)) {

        // if input maps to a dynamic policy with an activity listener, notify of state change

        if ((inputDesc->mPolicyMix != NULL)

                && ((inputDesc->mPolicyMix->mCbFlags & AudioMix::kCbFlagNotifyActivity) != 0)) {
@@ -1923,9 +1961,9 @@ status_t AudioPolicyManagerCustom::startInput(audio_io_handle_t input, 
        }

    }



    ALOGV("AudioPolicyManager::startInput() input source = %d", inputDesc->mInputSource);

    ALOGV("AudioPolicyManager::startInput() input source = %d", audioSession->inputSource());



    inputDesc->mRefCount++;

    audioSession->changeActiveCount(1);

#ifdef RECORD_PLAY_CONCURRENCY

    mIsInputRequestOnProgress = false;

#endif
@@ -1982,30 +2020,38 @@ AudioPolicyManagerCustom::AudioPolicyManagerCustom(AudioPolicyClientInterface *c 
        prop_ssr_enabled = atoi(ssr_enabled) || !strncmp("true", ssr_enabled, 4);

    }



    //TODO: Check the new logic to parse policy conf and update the below code

    //      Need this when SSR encoding is enabled



    if (property_get("ro.qc.sdk.audio.ssr", ssr_enabled, NULL)) {

        prop_ssr_enabled = atoi(ssr_enabled) || !strncmp("true", ssr_enabled, 4);

    }



    for (size_t i = 0; i < mHwModules.size(); i++) {

        ALOGV("Hw module %d", i);

        ALOGV("Hw module %zu", i);

        for (size_t j = 0; j < mHwModules[i]->mInputProfiles.size(); j++) {

            const sp<IOProfile> inProfile = mHwModules[i]->mInputProfiles[j];

            ALOGV("Input profile ", j);

            for (size_t k = 0; k  < inProfile->mChannelMasks.size(); k++) {

                audio_channel_mask_t channelMask =

                    inProfile->mChannelMasks.itemAt(k);

                ALOGV("Channel Mask %x size %d", channelMask,

                    inProfile->mChannelMasks.size());

            AudioProfileVector profiles = inProfile->getAudioProfiles();

            for (size_t k = 0; k < profiles.size(); k++){

                ChannelsVector channels = profiles[k]->getChannels();

                for (size_t x = 0; x < channels.size(); x++) {

                    audio_channel_mask_t channelMask = channels[x];

                    ALOGV("Channel Mask %x size %zu", channelMask,

                         channels.size());

                    if (AUDIO_CHANNEL_IN_5POINT1 == channelMask) {

                        if (!prop_ssr_enabled) {

                            ALOGI("removing AUDIO_CHANNEL_IN_5POINT1 from"

                                " input profile as SSR(surround sound record)"

                                " is not supported on this chipset variant");

                        inProfile->mChannelMasks.removeItemsAt(k, 1);

                        ALOGV("Channel Mask size now %d",

                            inProfile->mChannelMasks.size());

                            channels.removeItemsAt(x, 1);

                            ALOGV("Channel Mask size now %zu",

                                channels.size());

                        }

                    }

                }

            }

        }

    }



#ifdef RECORD_PLAY_CONCURRENCY

    mIsInputRequestOnProgress = false;

#endif

policy_hal/AudioPolicyManager.h

+3 −1
Original line number Diff line number Diff line 
/*

 * Copyright (c) 2013-2015, The Linux Foundation. All rights reserved.

 * Copyright (c) 2013-2016, The Linux Foundation. All rights reserved.

 * Not a contribution.

 *

 * Copyright (C) 2009 The Android Open Source Project
@@ -140,7 +140,9 @@ private: 
                const audio_offload_info_t *offloadInfo);

        // Used for voip + voice concurrency usecase

        int mPrevPhoneState;

#ifdef VOICE_CONCURRENCY

        int mvoice_call_state;

#endif

#ifdef RECORD_PLAY_CONCURRENCY

        // Used for record + playback concurrency

        bool mIsInputRequestOnProgress;