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Bug 1264199: P8. Handle potential resampling errors. r=kinetik

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The speex resampler can never return an error in its current state. But just in case and to handle any future changes to the speex resampler.
Also ensure that we can never access a stale speex resampler.

MozReview-Commit-ID: CEME0AA9ghK

--HG--
extra : rebase_source : 3dc40aa0833a85646be6d1d1a3b927fa0b5eb038
This commit is contained in:
Jean-Yves Avenard 2016-04-19 14:53:18 +10:00
parent 71b5b956c8
commit 2fb52f2e0d
1 changed files with 18 additions and 10 deletions
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28
dom/media/AudioConverter.cpp
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@ -105,10 +105,7 @@ AudioConverter::ReOrderInterleavedChannels(void* aOut, const void* aIn,
{
MOZ_DIAGNOSTIC_ASSERT(mIn.Channels() == mOut.Channels());
if (mOut.Layout() == mIn.Layout()) {
return;
}
if (mOut.Channels() == 1) {
if (mOut.Channels() == 1 || mOut.Layout() == mIn.Layout()) {
// If channel count is 1, planar and non-planar formats are the same and
// there's nothing to reorder.
if (aOut != aIn) {
@ -253,19 +250,28 @@ AudioConverter::ResampleAudio(void* aOut, const void* aIn, size_t aFrames)
uint32_t outframes = ResampleRecipientFrames(aFrames);
uint32_t inframes = aFrames;
int error;
if (mOut.Format() == AudioConfig::FORMAT_FLT) {
const float* in = reinterpret_cast<const float*>(aIn);
float* out = reinterpret_cast<float*>(aOut);
speex_resampler_process_interleaved_float(mResampler, in, &inframes,
out, &outframes);
error =
speex_resampler_process_interleaved_float(mResampler, in, &inframes,
out, &outframes);
} else if (mOut.Format() == AudioConfig::FORMAT_S16) {
const int16_t* in = reinterpret_cast<const int16_t*>(aIn);
int16_t* out = reinterpret_cast<int16_t*>(aOut);
speex_resampler_process_interleaved_int(mResampler, in, &inframes,
out, &outframes);
error =
speex_resampler_process_interleaved_int(mResampler, in, &inframes,
out, &outframes);
} else {
MOZ_DIAGNOSTIC_ASSERT(false, "Unsupported data type");
}
MOZ_ASSERT(error == RESAMPLER_ERR_SUCCESS);
if (error != RESAMPLER_ERR_SUCCESS) {
speex_resampler_destroy(mResampler);
mResampler = nullptr;
return 0;
}
MOZ_ASSERT(inframes == aFrames, "Some frames will be dropped");
return outframes;
}
@ -298,8 +304,7 @@ AudioConverter::DrainResampler(void* aOut)
return 0;
}
int frames = speex_resampler_get_input_latency(mResampler);
AlignedByteBuffer buffer(FramesOutToSamples(frames) *
AudioConfig::SampleSize(mOut.Format()));
AlignedByteBuffer buffer(FramesOutToBytes(frames));
if (!buffer) {
// OOM
return 0;
@ -358,6 +363,9 @@ AudioConverter::ResampleRecipientFrames(size_t aFrames) const
if (!aFrames && mIn.Rate() != mOut.Rate()) {
// The resampler will be drained, account for frames currently buffered
// in the resampler.
if (!mResampler) {
return 0;
}
return speex_resampler_get_output_latency(mResampler);
} else {
return (uint64_t)aFrames * mOut.Rate() / mIn.Rate() + 1;