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Commit 7ebec885 authored by Eric Laurent's avatar Eric Laurent Committed by Automerger Merge Worker
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Merge "AudioFlinger: allow shared record audio history for fast capture" into sc-dev am: 5f526b58 

Original change: https://googleplex-android-review.googlesource.com/c/platform/frameworks/av/+/14468784

Change-Id: Id5c522e7bed9b3188970afff2f033d13afcc8e2a
parents 323fcf87 5f526b58

services/audioflinger/Threads.cpp

+30 −22
Original line number Diff line number Diff line
@@ -7305,8 +7305,21 @@ reacquire_wakelock: 
                    //    the only active track

                    // 2) invalidate this track: this will cause the client to reconnect and possibly

                    //    be invalidated again until unsilenced

                    bool invalidate = false;

                    if (activeTrack->isSilenced()) {

                        if (size > 1) {

                            invalidate = true;

                        } else {

                            silenceFastCapture = true;

                        }

                    }

                    // Invalidate fast tracks if access to audio history is required as this is not

                    // possible with fast tracks. Once the fast track has been invalidated, no new

                    // fast track will be created until mMaxSharedAudioHistoryMs is cleared.

                    if (mMaxSharedAudioHistoryMs != 0) {

                        invalidate = true;

                    }

                    if (invalidate) {

                        activeTrack->invalidate();

                        ALOG_ASSERT(fastTrackToRemove == 0);

                        fastTrackToRemove = activeTrack;
@@ -7314,9 +7327,6 @@ reacquire_wakelock: 
                        mActiveTracks.remove(activeTrack);

                        size--;

                        continue;

                        } else {

                            silenceFastCapture = true;

                        }

                    }

                    fastTrack = activeTrack;

                }
@@ -7837,12 +7847,6 @@ sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRe 
            lStatus = PERMISSION_DENIED;

            goto Exit;

        }

        //TODO: b/185972521 allow resampling buffer resizing on fast mixers by pausing

        // the fast mixer thread while resizing the buffer in the normal thread

        if (hasFastCapture()) {

            lStatus = BAD_VALUE;

            goto Exit;

        }

        if (maxSharedAudioHistoryMs < 0

                || maxSharedAudioHistoryMs > AudioFlinger::kMaxSharedAudioHistoryMs) {

            lStatus = BAD_VALUE;
@@ -7854,8 +7858,9 @@ sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRe 
    }

    sampleRate = *pSampleRate;



    // special case for FAST flag considered OK if fast capture is present

    if (hasFastCapture()) {

    // special case for FAST flag considered OK if fast capture is present and access to

    // audio history is not required

    if (hasFastCapture() && mMaxSharedAudioHistoryMs == 0) {

        inputFlags = (audio_input_flags_t)(inputFlags | AUDIO_INPUT_FLAG_FAST);

    }
@@ -7867,8 +7872,9 @@ sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRe 
        *flags = (audio_input_flags_t)(*flags & inputFlags);

    }



    // client expresses a preference for FAST, but we get the final say

    if (*flags & AUDIO_INPUT_FLAG_FAST) {

    // client expresses a preference for FAST and no access to audio history,

    // but we get the final say

    if (*flags & AUDIO_INPUT_FLAG_FAST && maxSharedAudioHistoryMs == 0) {

      if (

            // we formerly checked for a callback handler (non-0 tid),

            // but that is no longer required for TRANSFER_OBTAIN mode
@@ -7988,7 +7994,6 @@ sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRe 
        if (maxSharedAudioHistoryMs != 0) {

            sendResizeBufferConfigEvent_l(maxSharedAudioHistoryMs);

        }



    }



    lStatus = NO_ERROR;
@@ -8219,9 +8224,6 @@ status_t AudioFlinger::RecordThread::shareAudioHistory( 
status_t AudioFlinger::RecordThread::shareAudioHistory_l(

        const std::string& sharedAudioPackageName, audio_session_t sharedSessionId,

        int64_t sharedAudioStartMs) {

    if (hasFastCapture()) {

        return BAD_VALUE;

    }

    if ((hasAudioSession_l(sharedSessionId) & ThreadBase::TRACK_SESSION) == 0) {

        return BAD_VALUE;

    }
@@ -8464,6 +8466,7 @@ status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 
    // FIXME if client not keeping up, discard

    LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames));

    // 'filled' may be non-contiguous, so return only the first contiguous chunk



    front &= recordThread->mRsmpInFramesP2 - 1;

    size_t part1 = recordThread->mRsmpInFramesP2 - front;

    if (part1 > (size_t) filled) {
@@ -8678,7 +8681,7 @@ void AudioFlinger::RecordThread::readInputParameters_l() 


    // mRsmpInFrames must be 0 before calling resizeInputBuffer_l for the first time

    mRsmpInFrames = 0;

    resizeInputBuffer_l();

    resizeInputBuffer_l(0 /*maxSharedAudioHistoryMs*/);



    // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints.

    // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks?
@@ -8919,6 +8922,10 @@ void AudioFlinger::RecordThread::resizeInputBuffer_l(int32_t maxSharedAudioHisto 
    int32_t previousRear = mRsmpInRear;

    mRsmpInRear = 0;



    ALOG_ASSERT(maxSharedAudioHistoryMs >= 0

            && maxSharedAudioHistoryMs <= AudioFlinger::kMaxSharedAudioHistoryMs,

            "resizeInputBuffer_l() called with invalid max shared history %d",

            maxSharedAudioHistoryMs);

    if (maxSharedAudioHistoryMs != 0) {

        // resizeInputBuffer_l should never be called with a non zero shared history if the

        // buffer was not already allocated
@@ -8931,6 +8938,7 @@ void AudioFlinger::RecordThread::resizeInputBuffer_l(int32_t maxSharedAudioHisto 
        }

        mRsmpInFrames = rsmpInFrames;

    }

    mMaxSharedAudioHistoryMs = maxSharedAudioHistoryMs;

    // Note: mRsmpInFrames is 0 when called with maxSharedAudioHistoryMs equals to 0 so it is always

    // initialized

    if (mRsmpInFrames < minRsmpInFrames) {

services/audioflinger/Threads.h

+3 −2
Original line number Diff line number Diff line
@@ -338,7 +338,7 @@ public: 
    virtual     void        updateOutDevices(const DeviceDescriptorBaseVector& outDevices);

    virtual     void        toAudioPortConfig(struct audio_port_config *config) = 0;



    virtual     void        resizeInputBuffer_l(int32_t maxSharedAudioHistoryMs = 0);

    virtual     void        resizeInputBuffer_l(int32_t maxSharedAudioHistoryMs);
@@ -1717,7 +1717,7 @@ public: 
                                           audio_patch_handle_t *handle);

    virtual status_t    releaseAudioPatch_l(const audio_patch_handle_t handle);

            void        updateOutDevices(const DeviceDescriptorBaseVector& outDevices) override;

            void        resizeInputBuffer_l(int32_t maxSharedAudioHistoryMs = 0) override;

            void        resizeInputBuffer_l(int32_t maxSharedAudioHistoryMs) override;



            void        addPatchTrack(const sp<PatchRecord>& record);

            void        deletePatchTrack(const sp<PatchRecord>& record);
@@ -1862,6 +1862,7 @@ private: 


            DeviceDescriptorBaseVector          mOutDevices;



            int32_t                             mMaxSharedAudioHistoryMs = 0;

            std::string                         mSharedAudioPackageName = {};

            int32_t                             mSharedAudioStartFrames = -1;

            audio_session_t                     mSharedAudioSessionId = AUDIO_SESSION_NONE;