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Commit 4bb21c49 authored by Eric Laurent's avatar Eric Laurent
Browse files

Fix issue 3479042. 

The problem is that when an AudioRecord using the resampler is restarted,
the resampler state is not reset (as there is no reset function in the resampler).
The consequence is that the first time the record thread loop runs, it calls the resampler
which consumes the remaining data in the input buffer and when this buffer is released
the input index is incremented over the limit.

The fix consists in implementing a reset function in the resampler.

A similar problem was also present for playback but unoticed because the track buffer is always
drained by the mixer when a track stops. The only problem for playback was that the initial
phase fraction was wrong when restarting a track after stop (it was correct after a pause).

Change-Id: Ifc2585d685f4402d29f4afc63f6efd1d69265de3
parent dd1ddc3b

services/audioflinger/AudioFlinger.cpp

+5 −1
Original line number Diff line number Diff line
@@ -1629,6 +1629,7 @@ uint32_t AudioFlinger::MixerThread::prepareTracks_l(const SortedVector< wp<Track 
                    track->mState = TrackBase::ACTIVE;

                    param = AudioMixer::RAMP_VOLUME;

                }

                mAudioMixer->setParameter(AudioMixer::RESAMPLE, AudioMixer::RESET, NULL);

            } else if (cblk->server != 0) {

                // If the track is stopped before the first frame was mixed,

                // do not apply ramp
@@ -3855,9 +3856,12 @@ status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrac 
            mActiveTrack.clear();

            return status;

        }

        mActiveTrack->mState = TrackBase::RESUMING;

        mRsmpInIndex = mFrameCount;

        mBytesRead = 0;

        if (mResampler != NULL) {

            mResampler->reset();

        }

        mActiveTrack->mState = TrackBase::RESUMING;

        // signal thread to start

        LOGV("Signal record thread");

        mWaitWorkCV.signal();

services/audioflinger/AudioMixer.cpp

+13 −0
Original line number Diff line number Diff line
@@ -220,6 +220,12 @@ status_t AudioMixer::setParameter(int target, int name, void *value) 
                return NO_ERROR;

            }

        }

        if (name == RESET) {

            track_t& track = mState.tracks[ mActiveTrack ];

            track.resetResampler();

            invalidateState(1<<mActiveTrack);

            return NO_ERROR;

        }

        break;

    case RAMP_VOLUME:

    case VOLUME:
@@ -289,6 +295,13 @@ bool AudioMixer::track_t::doesResample() const 
    return resampler != 0;

}



void AudioMixer::track_t::resetResampler()

{

    if (resampler != 0) {

        resampler->reset();

    }

}



inline

void AudioMixer::track_t::adjustVolumeRamp(bool aux)

{

services/audioflinger/AudioMixer.h

+2 −0
Original line number Diff line number Diff line
@@ -67,6 +67,7 @@ public: 
        AUX_BUFFER      = 0x4003,

        // for TARGET RESAMPLE

        SAMPLE_RATE     = 0x4100,

        RESET           = 0x4101,

        // for TARGET VOLUME (8 channels max)

        VOLUME0         = 0x4200,

        VOLUME1         = 0x4201,
@@ -163,6 +164,7 @@ private: 


        bool        setResampler(uint32_t sampleRate, uint32_t devSampleRate);

        bool        doesResample() const;

        void        resetResampler();

        void        adjustVolumeRamp(bool aux);

    };

services/audioflinger/AudioResampler.cpp

+6 −0
Original line number Diff line number Diff line
@@ -148,6 +148,12 @@ void AudioResampler::setVolume(int16_t left, int16_t right) { 
    mVolume[1] = right;

}



void AudioResampler::reset() {

    mInputIndex = 0;

    mPhaseFraction = 0;

    mBuffer.frameCount = 0;

}



// ----------------------------------------------------------------------------



void AudioResamplerOrder1::resample(int32_t* out, size_t outFrameCount,

services/audioflinger/AudioResampler.h

+2 −0
Original line number Diff line number Diff line
@@ -53,6 +53,8 @@ public: 
    virtual void resample(int32_t* out, size_t outFrameCount,

            AudioBufferProvider* provider) = 0;



    virtual void reset();



protected:

    // number of bits for phase fraction - 30 bits allows nearly 2x downsampling

    static const int kNumPhaseBits = 30;