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audio: check for divide by zero possibilities
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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: If7fbcb6eb4cfcdba6d6bb3b4f758e46e4a57e37b
(cherry picked from commit d1576c1648a78354535fc9cd7130412f300864e5)
(cherry picked from commit 15b951bfcb2180f728669ab23ef0d6791fd790c2)
(cherry picked from commit c980dbea39f984a3c28e2694174e87b311e77308)
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Pavan Chikkala authored and Linux Build Service Account committed Oct 31, 2013
1 parent 3f5fc47 commit 79bb230
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Showing 2 changed files with 41 additions and 13 deletions.
27 changes: 19 additions & 8 deletions media/libmedia/AudioTrack.cpp
View file
Original file line number Diff line number Diff line change
Expand Up @@ -54,19 +54,22 @@ status_t AudioTrack::getMinFrameCount(
// audio_format_t format
// audio_channel_mask_t channelMask
// audio_output_flags_t flags
uint32_t afSampleRate;
uint32_t afSampleRate = 0;
if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
return NO_INIT;
}
size_t afFrameCount;
size_t 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) {
Expand Down Expand Up @@ -883,21 +886,21 @@ status_t AudioTrack::createTrack_l(

// Not all of these values are needed under all conditions, but it is easier to get them all

uint32_t afLatency;
uint32_t afLatency = 0;
status = AudioSystem::getLatency(output, streamType, &afLatency);
if (status != NO_ERROR) {
ALOGE("getLatency(%d) failed status %d", output, status);
return NO_INIT;
}

size_t afFrameCount;
size_t afFrameCount = 0;
status = AudioSystem::getFrameCount(output, streamType, &afFrameCount);
if (status != NO_ERROR) {
ALOGE("getFrameCount(output=%d, streamType=%d) status %d", output, streamType, status);
return NO_INIT;
}

uint32_t afSampleRate;
uint32_t afSampleRate = 0;
status = AudioSystem::getSamplingRate(output, streamType, &afSampleRate);
if (status != NO_ERROR) {
ALOGE("getSamplingRate(output=%d, streamType=%d) status %d", output, streamType, status);
Expand Down Expand Up @@ -966,6 +969,10 @@ status_t AudioTrack::createTrack_l(
// FIXME move these calculations and associated checks to server

// Ensure that buffer depth covers at least audio hardware latency
if(!afSampleRate && !afFrameCount) {
ALOGW("samplerate or framecount zero");
return NO_INIT;
}
uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate);
ALOGV("afFrameCount=%d, minBufCount=%d, afSampleRate=%u, afLatency=%d",
afFrameCount, minBufCount, afSampleRate, afLatency);
Expand Down Expand Up @@ -1102,7 +1109,11 @@ status_t AudioTrack::createTrack_l(

mAudioTrack->attachAuxEffect(mAuxEffectId);
// FIXME don't believe this lie
mLatency = afLatency + (1000*frameCount) / sampleRate;
if(sampleRate){
mLatency = afLatency + (1000*frameCount) / sampleRate;
}else{
mLatency = afLatency;
}
mFrameCount = frameCount;
// If IAudioTrack is re-created, don't let the requested frameCount
// decrease. This can confuse clients that cache frameCount().
Expand Down
27 changes: 22 additions & 5 deletions services/audioflinger/Threads.cpp
View file
Original file line number Diff line number Diff line change
Expand Up @@ -1248,7 +1248,10 @@ 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;
}
Expand Down Expand Up @@ -2617,7 +2620,10 @@ uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const
{
if (mFastMixer != NULL) {
MonoPipe *pipe = (MonoPipe *)mPipeSink.get();
latency += (pipe->getAvgFrames() * 1000) / mSampleRate;
if(mSampleRate)
latency += (pipe->getAvgFrames() * 1000) / mSampleRate;
else
ALOGW("SampleRate is 0");
}
return latency;
}
Expand Down Expand Up @@ -3005,7 +3011,12 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac
desiredFrames = mNormalFrameCount;
} else {
// +1 for rounding and +1 for additional sample needed for interpolation
desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1;
if (mSampleRate) {
desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1;
} else {
desiredFrames = 2;
ALOGW("SampleRate is 0");
}
// add frames already consumed but not yet released by the resampler
// because cblk->framesReady() will include these frames
desiredFrames += mAudioMixer->getUnreleasedFrames(track->name());
Expand Down Expand Up @@ -4237,7 +4248,10 @@ void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread)
{
Mutex::Autolock _l(mLock);
// FIXME explain this formula
size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate();
int sampleRate = thread->sampleRate();
size_t frameCount = 0;
if (sampleRate)
frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate();
OutputTrack *outputTrack = new OutputTrack(thread,
this,
mSampleRate,
Expand Down Expand Up @@ -4664,7 +4678,10 @@ sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createR
// If you change this calculation, also review the start threshold which is related.
uint32_t latencyMs = 50; // FIXME mInput->stream->get_latency(mInput->stream);
size_t mNormalFrameCount = 2048; // FIXME
uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate);
uint32_t minBufCount = 0;
if (mSampleRate) {
minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate);
}
if (minBufCount < 2) {
minBufCount = 2;
}
Expand Down

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