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Commit abc9af07 authored by Jitendra Singh Naruka's avatar Jitendra Singh Naruka Committed by Steve Kondik
Browse files

media, audioflinger: check for divide by zero possibilities and err 

When output stream is not available to audioflinger due to any reason
, sampleRate and frameCount have zero values when trying to create
new Audiotrack. This might result in divide by 0 situation.

Change-Id: Ic13cb51facb8497e68ab596abb027b44f496b907
CRs-Fixed: 478480
parent a3602971

media/libmedia/AudioTrack.cpp

+22 −8
Original line number Diff line number Diff line
@@ -69,25 +69,30 @@ status_t AudioTrack::getMinFrameCount( 
    //          audio_format_t format

    //          audio_channel_mask_t channelMask

    //          audio_output_flags_t flags

    int afSampleRate;

    int afSampleRate = 0;

    if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {

        return NO_INIT;

    }

    int afFrameCount;

    int afFrameCount = 0;

    if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) {

        return NO_INIT;

    }

    uint32_t afLatency;

    uint32_t afLatency = 0;

    if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) {

        return NO_INIT;

    }



    if(!afSampleRate || !afFrameCount) {

        ALOGW("samplerate or framecount 0");

        return NO_INIT;

    }

    // Ensure that buffer depth covers at least audio hardware latency

    uint32_t minBufCount = afLatency / ((1000 * afFrameCount) / afSampleRate);

    if (minBufCount < 2) minBufCount = 2;



    *frameCount = (sampleRate == 0) ? afFrameCount * minBufCount :

                afFrameCount * minBufCount * sampleRate / afSampleRate;



    ALOGV("getMinFrameCount=%d: afFrameCount=%d, minBufCount=%d, afSampleRate=%d, afLatency=%d",

            *frameCount, afFrameCount, minBufCount, afSampleRate, afLatency);

    return NO_ERROR;
@@ -988,20 +993,26 @@ status_t AudioTrack::createTrack_l( 
    } else if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) {



        // FIXME move these calculations and associated checks to server

        int afSampleRate;

        int afSampleRate = 0;

        if (AudioSystem::getSamplingRate(output, streamType, &afSampleRate) != NO_ERROR) {

            return NO_INIT;

        }

        int afFrameCount;

        int afFrameCount = 0;

        if (AudioSystem::getFrameCount(output, streamType, &afFrameCount) != NO_ERROR) {

            return NO_INIT;

        }



        if(!afSampleRate && !afFrameCount) {

            ALOGW("samplerate or framecount zero");

            return NO_INIT;

        }



        // Ensure that buffer depth covers at least audio hardware latency

        uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate);

        if (minBufCount < 2) minBufCount = 2;



        int minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate;

        uint32_t minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate;



        ALOGV("minFrameCount: %d, afFrameCount=%d, minBufCount=%d, sampleRate=%d, afSampleRate=%d"

                ", afLatency=%d",

                minFrameCount, afFrameCount, minBufCount, sampleRate, afSampleRate, afLatency);
@@ -1096,7 +1107,10 @@ status_t AudioTrack::createTrack_l( 
    mCblk->waitTimeMs = 0;

    mRemainingFrames = mNotificationFramesAct;

    // FIXME don't believe this lie

    if(sampleRate)

        mLatency = afLatency + (1000*mCblk->frameCount) / sampleRate;

    else

        mLatency = afLatency;

    // If IAudioTrack is re-created, don't let the requested frameCount

    // decrease.  This can confuse clients that cache frameCount().

    if (mCblk->frameCount > mFrameCount) {

services/audioflinger/AudioFlinger.cpp

+17 −4
Original line number Diff line number Diff line
@@ -2176,7 +2176,9 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrac 
        // This is probably too conservative, but legacy application code may depend on it.

        // If you change this calculation, also review the start threshold which is related.

        uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream);

        uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate);

        uint32_t minBufCount = 0;

        if(mSampleRate)

            minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate);

        if (minBufCount < 2) {

            minBufCount = 2;

        }
@@ -2279,7 +2281,10 @@ uint32_t AudioFlinger::MixerThread::correctLatency(uint32_t latency) const 
{

    if (mFastMixer != NULL) {

        MonoPipe *pipe = (MonoPipe *)mPipeSink.get();

        if(mSampleRate)

            latency += (pipe->getAvgFrames() * 1000) / mSampleRate;

        else

            ALOGW("SampleRate is 0");

    }

    return latency;

}
@@ -3521,7 +3526,12 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac 
                minFrames = mNormalFrameCount;

            } else {

                // +1 for rounding and +1 for additional sample needed for interpolation

                if(mSampleRate)

                    minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1;

                else {

                    minFrames = 2;

                    ALOGW("SampleRate is 0");

                }

                // add frames already consumed but not yet released by the resampler

                // because cblk->framesReady() will include these frames

                minFrames += mAudioMixer->getUnreleasedFrames(track->name());
@@ -4467,7 +4477,10 @@ void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 
{

    Mutex::Autolock _l(mLock);

    // FIXME explain this formula

    int frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate();

    int sampleRate = thread->sampleRate();

    int frameCount = 0;

    if (sampleRate)

        frameCount = (3 * mNormalFrameCount * mSampleRate) / sampleRate;

    OutputTrack *outputTrack = new OutputTrack(thread,

                                            this,

                                            mSampleRate,