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dom/media: add S16 support code to audio decoders

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roytam1 2023-08-26 07:37:26 +08:00
commit 28f55eb5c4
2 changed files with 49 additions and 0 deletions
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45
dom/media/platforms/ffmpeg/FFmpegAudioDecoder.cpp
View file

@ -46,7 +46,11 @@ FFmpegAudioDecoder<LIBAV_VER>::InitCodecContext()
// FFmpeg takes this as a suggestion for what format to use for audio samples.
// LibAV 0.8 produces rubbish float interleaved samples, request 16 bits audio.
mCodecContext->request_sample_fmt =
#ifdef MOZ_SAMPLE_TYPE_FLOAT32
(mLib->mVersion == 53) ? AV_SAMPLE_FMT_S16 : AV_SAMPLE_FMT_FLT;
#else
AV_SAMPLE_FMT_S16;
#endif
}
static AlignedAudioBuffer
@ -60,11 +64,23 @@ CopyAndPackAudio(AVFrame* aFrame, uint32_t aNumChannels, uint32_t aNumAFrames)
}
if (aFrame->format == AV_SAMPLE_FMT_FLT) {
#ifdef MOZ_SAMPLE_TYPE_FLOAT32
// Audio data already packed. No need to do anything other than copy it
// into a buffer we own.
memcpy(audio.get(), aFrame->data[0],
aNumChannels * aNumAFrames * sizeof(AudioDataValue));
#else
// Audio data already packed. Need to convert from 32 bits Float to S16
AudioDataValue* tmp = audio.get();
float* data = reinterpret_cast<float**>(aFrame->data)[0];
for (uint32_t frame = 0; frame < aNumAFrames; frame++) {
for (uint32_t channel = 0; channel < aNumChannels; channel++) {
*tmp++ = FloatToAudioSample<int16_t>(*data++);
}
}
#endif
} else if (aFrame->format == AV_SAMPLE_FMT_FLTP) {
#ifdef MOZ_SAMPLE_TYPE_FLOAT32
// Planar audio data. Pack it into something we can understand.
AudioDataValue* tmp = audio.get();
AudioDataValue** data = reinterpret_cast<AudioDataValue**>(aFrame->data);
@ -73,7 +89,19 @@ CopyAndPackAudio(AVFrame* aFrame, uint32_t aNumChannels, uint32_t aNumAFrames)
*tmp++ = data[channel][frame];
}
}
#else
// Planar audio data. Convert it from 32 bits Float to S16
// and pack it into something we can understand.
AudioDataValue* tmp = audio.get();
float** data = reinterpret_cast<float**>(aFrame->data);
for (uint32_t frame = 0; frame < aNumAFrames; frame++) {
for (uint32_t channel = 0; channel < aNumChannels; channel++) {
*tmp++ = FloatToAudioSample<int16_t>(data[channel][frame]);
}
}
#endif
} else if (aFrame->format == AV_SAMPLE_FMT_S16) {
#ifdef MOZ_SAMPLE_TYPE_FLOAT32
// Audio data already packed. Need to convert from S16 to 32 bits Float
AudioDataValue* tmp = audio.get();
int16_t* data = reinterpret_cast<int16_t**>(aFrame->data)[0];
@ -82,7 +110,14 @@ CopyAndPackAudio(AVFrame* aFrame, uint32_t aNumChannels, uint32_t aNumAFrames)
*tmp++ = AudioSampleToFloat(*data++);
}
}
#else
// Audio data already packed. No need to do anything other than copy it
// into a buffer we own.
memcpy(audio.get(), aFrame->data[0],
aNumChannels * aNumAFrames * sizeof(AudioDataValue));
#endif
} else if (aFrame->format == AV_SAMPLE_FMT_S16P) {
#ifdef MOZ_SAMPLE_TYPE_FLOAT32
// Planar audio data. Convert it from S16 to 32 bits float
// and pack it into something we can understand.
AudioDataValue* tmp = audio.get();
@ -111,6 +146,16 @@ CopyAndPackAudio(AVFrame* aFrame, uint32_t aNumChannels, uint32_t aNumAFrames)
*tmp++ = AudioSampleToFloat(data[channel][frame]);
}
}
#else
// Planar audio data. Pack it into something we can understand.
AudioDataValue* tmp = audio.get();
AudioDataValue** data = reinterpret_cast<AudioDataValue**>(aFrame->data);
for (uint32_t frame = 0; frame < aNumAFrames; frame++) {
for (uint32_t channel = 0; channel < aNumChannels; channel++) {
*tmp++ = data[channel][frame];
}
}
#endif
}
return audio;

4
dom/media/platforms/wmf/WMFAudioMFTManager.cpp
View file

@ -312,7 +312,11 @@ WMFAudioMFTManager::Output(int64_t aStreamOffset,
int16_t* pcm = (int16_t*)data;
for (int32_t i = 0; i < numSamples; ++i) {
#ifdef MOZ_SAMPLE_TYPE_FLOAT32
audioData[i] = AudioSampleToFloat(pcm[i]);
#else
audioData[i] = pcm[i];
#endif
}
buffer->Unlock();