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Commit 57326621 authored by Eric Laurent's avatar Eric Laurent
Browse files

Fix issue 1743700: AudioTrack: setPlaybackRate can not set the playback rate... 

Fix issue 1743700: AudioTrack: setPlaybackRate can not set the playback rate to twice of the ouputSR

Store sample rate on 32 bits instead of 16 bits in audio_track_cblk_t.
Removed sampleRate() methods from AudioTrack and AudioRecord: replaced by getSampleRate().
AudioTrack::setSampleRate() no returns a status.
parent 76aa2837

include/media/AudioRecord.h

+1 −3
Original line number Diff line number Diff line
@@ -197,7 +197,6 @@ public: 


   /* getters, see constructor */



            uint32_t    sampleRate() const;

            int         format() const;

            int         channelCount() const;

            uint32_t    frameCount() const;
@@ -217,7 +216,7 @@ public: 
            status_t    stop();

            bool        stopped() const;



    /* get sample rate for this track

    /* get sample rate for this record track

     */

            uint32_t    getSampleRate();
@@ -323,7 +322,6 @@ private: 
    sp<ClientRecordThread>  mClientRecordThread;

    Mutex                   mRecordThreadLock;



    uint32_t                mSampleRate;

    uint32_t                mFrameCount;



    audio_track_cblk_t*     mCblk;

include/media/AudioTrack.h

+1 −2
Original line number Diff line number Diff line
@@ -201,7 +201,6 @@ public: 
    /* getters, see constructor */



            int         streamType() const;

            uint32_t    sampleRate() const;

            int         format() const;

            int         channelCount() const;

            uint32_t    frameCount() const;
@@ -246,7 +245,7 @@ public: 


    /* set sample rate for this track, mostly used for games' sound effects

     */

            void        setSampleRate(int sampleRate);

            status_t    setSampleRate(int sampleRate);

            uint32_t    getSampleRate();



    /* Enables looping and sets the start and end points of looping.

include/private/media/AudioTrackShared.h

+3 −4
Original line number Diff line number Diff line
@@ -26,7 +26,6 @@ namespace android { 


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



#define MAX_SAMPLE_RATE     65535

#define THREAD_PRIORITY_AUDIO_CLIENT (ANDROID_PRIORITY_AUDIO)

// Maximum cumulated timeout milliseconds before restarting audioflinger thread

#define MAX_STARTUP_TIMEOUT_MS  3000    // Longer timeout period at startup to cope with A2DP init time
@@ -55,9 +54,9 @@ struct audio_track_cblk_t 
                    uint16_t    volume[2];

                    uint32_t    volumeLR;

                };

                uint16_t    sampleRate;

                uint16_t    channels;

                int16_t     flowControlFlag; // underrun (out) or overrrun (in) indication

                uint32_t    sampleRate;

                uint8_t     channels;

                uint8_t     flowControlFlag; // underrun (out) or overrrun (in) indication

                uint8_t     out;        // out equals 1 for AudioTrack and 0 for AudioRecord

                uint8_t     forceReady; 

                uint16_t    bufferTimeoutMs; // Maximum cumulated timeout before restarting audioflinger

media/libmedia/AudioRecord.cpp

+6 −7
Original line number Diff line number Diff line
@@ -185,7 +185,6 @@ status_t AudioRecord::set( 
    mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer());

    mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);

    mCblk->out = 0;

    mSampleRate = sampleRate;

    mFormat = format;

    // Update buffer size in case it has been limited by AudioFlinger during track creation

    mFrameCount = mCblk->frameCount;
@@ -196,7 +195,7 @@ status_t AudioRecord::set( 
    mRemainingFrames = notificationFrames;

    mUserData = user;

    // TODO: add audio hardware input latency here

    mLatency = (1000*mFrameCount) / mSampleRate;

    mLatency = (1000*mFrameCount) / sampleRate;

    mMarkerPosition = 0;

    mMarkerReached = false;

    mNewPosition = 0;
@@ -218,11 +217,6 @@ uint32_t AudioRecord::latency() const 
    return mLatency;

}



uint32_t AudioRecord::sampleRate() const

{

    return mSampleRate;

}



int AudioRecord::format() const

{

    return mFormat;
@@ -321,6 +315,11 @@ bool AudioRecord::stopped() const 
    return !mActive;

}



uint32_t AudioRecord::getSampleRate()

{

    return mCblk->sampleRate;

}



status_t AudioRecord::setMarkerPosition(uint32_t marker)

{

    if (mCbf == 0) return INVALID_OPERATION;

media/libmedia/AudioTrack.cpp

+8 −15
Original line number Diff line number Diff line
@@ -243,7 +243,6 @@ status_t AudioTrack::set( 
    mCblk->volume[0] = mCblk->volume[1] = 0x1000;

    mVolume[LEFT] = 1.0f;

    mVolume[RIGHT] = 1.0f;

    mSampleRate = sampleRate;

    mStreamType = streamType;

    mFormat = format;

    mChannelCount = channelCount;
@@ -254,7 +253,7 @@ status_t AudioTrack::set( 
    mNotificationFrames = notificationFrames;

    mRemainingFrames = notificationFrames;

    mUserData = user;

    mLatency = afLatency + (1000*mFrameCount) / mSampleRate;

    mLatency = afLatency + (1000*mFrameCount) / sampleRate;

    mLoopCount = 0;

    mMarkerPosition = 0;

    mMarkerReached = false;
@@ -281,11 +280,6 @@ int AudioTrack::streamType() const 
    return mStreamType;

}



uint32_t AudioTrack::sampleRate() const

{

    return mSampleRate;

}



int AudioTrack::format() const

{

    return mFormat;
@@ -438,24 +432,23 @@ void AudioTrack::getVolume(float* left, float* right) 
    *right = mVolume[RIGHT];

}



void AudioTrack::setSampleRate(int rate)

status_t AudioTrack::setSampleRate(int rate)

{

    int afSamplingRate;



    if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) {

        return;

        return NO_INIT;

    }

    // Resampler implementation limits input sampling rate to 2 x output sampling rate.

    if (rate <= 0) rate = 1;

    if (rate > afSamplingRate*2) rate = afSamplingRate*2;

    if (rate > MAX_SAMPLE_RATE) rate = MAX_SAMPLE_RATE;

    if (rate <= 0 || rate > afSamplingRate*2 ) return BAD_VALUE;



    mCblk->sampleRate = (uint16_t)rate;

    mCblk->sampleRate = rate;

    return NO_ERROR;

}



uint32_t AudioTrack::getSampleRate()

{

    return uint32_t(mCblk->sampleRate);

    return mCblk->sampleRate;

}



status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount)
@@ -866,7 +859,7 @@ status_t AudioTrack::dump(int fd, const Vector<String16>& args) const 
    result.append(buffer);

    snprintf(buffer, 255, "  format(%d), channel count(%d), frame count(%d)\n", mFormat, mChannelCount, mFrameCount);

    result.append(buffer);

    snprintf(buffer, 255, "  sample rate(%d), status(%d), muted(%d)\n", mSampleRate, mStatus, mMuted);

    snprintf(buffer, 255, "  sample rate(%d), status(%d), muted(%d)\n", (mCblk == 0) ? 0 : mCblk->sampleRate, mStatus, mMuted);

    result.append(buffer);

    snprintf(buffer, 255, "  active(%d), latency (%d)\n", mActive, mLatency);

    result.append(buffer);