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Bug 1098637 - Enforce FFmpeg's audio decode API. r=edwin

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This commit is contained in:
Jean-Yves Avenard 2014-11-15 10:19:55 +11:00
parent 5c9077604a
commit d104b16262
1 changed files with 43 additions and 24 deletions
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67
dom/media/fmp4/ffmpeg/FFmpegAudioDecoder.cpp
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@ -92,7 +92,6 @@ FFmpegAudioDecoder<LIBAV_VER>::DecodePacket(MP4Sample* aSample)
aSample->Pad(FF_INPUT_BUFFER_PADDING_SIZE);
packet.data = aSample->data;
packet.size = aSample->size;
packet.pos = aSample->byte_offset;
if (!PrepareFrame()) {
NS_WARNING("FFmpeg audio decoder failed to allocate frame.");
@ -100,31 +99,50 @@ FFmpegAudioDecoder<LIBAV_VER>::DecodePacket(MP4Sample* aSample)
return;
}
int decoded;
int bytesConsumed =
avcodec_decode_audio4(mCodecContext, mFrame, &decoded, &packet);
int64_t samplePosition = aSample->byte_offset;
Microseconds pts = aSample->composition_timestamp;
if (bytesConsumed < 0 || !decoded) {
NS_WARNING("FFmpeg audio decoder error.");
mCallback->Error();
return;
while (packet.size > 0) {
int decoded;
int bytesConsumed =
avcodec_decode_audio4(mCodecContext, mFrame, &decoded, &packet);
if (bytesConsumed < 0) {
NS_WARNING("FFmpeg audio decoder error.");
mCallback->Error();
return;
}
if (decoded) {
uint32_t numChannels = mCodecContext->channels;
uint32_t samplingRate = mCodecContext->sample_rate;
nsAutoArrayPtr<AudioDataValue> audio(
CopyAndPackAudio(mFrame, numChannels, mFrame->nb_samples));
CheckedInt<Microseconds> duration =
FramesToUsecs(mFrame->nb_samples, samplingRate);
if (!duration.isValid()) {
NS_WARNING("Invalid count of accumulated audio samples");
mCallback->Error();
return;
}
AudioData* data = new AudioData(samplePosition,
pts,
duration.value(),
mFrame->nb_samples,
audio.forget(),
numChannels,
samplingRate);
mCallback->Output(data);
pts += duration.value();
}
packet.data += bytesConsumed;
packet.size -= bytesConsumed;
samplePosition += bytesConsumed;
}
NS_ASSERTION(bytesConsumed == (int)aSample->size,
"Only one audio packet should be received at a time.");
uint32_t numChannels = mCodecContext->channels;
uint32_t samplingRate = mCodecContext->sample_rate;
nsAutoArrayPtr<AudioDataValue> audio(
CopyAndPackAudio(mFrame, numChannels, mFrame->nb_samples));
nsAutoPtr<AudioData> data(
new AudioData(packet.pos, aSample->composition_timestamp, aSample->duration,
mFrame->nb_samples, audio.forget(), numChannels, samplingRate));
mCallback->Output(data.forget());
if (mTaskQueue->IsEmpty()) {
mCallback->InputExhausted();
}
@ -142,8 +160,9 @@ FFmpegAudioDecoder<LIBAV_VER>::Input(MP4Sample* aSample)
nsresult
FFmpegAudioDecoder<LIBAV_VER>::Drain()
{
mTaskQueue->AwaitIdle();
mCallback->DrainComplete();
return NS_OK;
return Flush();
}
AVCodecID