mirror of
https://github.com/roytam1/UXP.git
synced 2026-05-26 14:39:05 +00:00
278 lines
9.5 KiB
C++
278 lines
9.5 KiB
C++
/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
|
|
/* This Source Code Form is subject to the terms of the Mozilla Public
|
|
* License, v. 2.0. If a copy of the MPL was not distributed with this
|
|
* file, You can obtain one at http://mozilla.org/MPL/2.0/. */
|
|
|
|
#include "mozilla/TaskQueue.h"
|
|
|
|
#include "FFmpegAudioDecoder.h"
|
|
#include "TimeUnits.h"
|
|
|
|
#define MAX_CHANNELS 16
|
|
|
|
namespace mozilla
|
|
{
|
|
|
|
FFmpegAudioDecoder<LIBAV_VER>::FFmpegAudioDecoder(FFmpegLibWrapper* aLib,
|
|
TaskQueue* aTaskQueue, MediaDataDecoderCallback* aCallback,
|
|
const AudioInfo& aConfig)
|
|
: FFmpegDataDecoder(aLib, aTaskQueue, aCallback, GetCodecId(aConfig.mMimeType))
|
|
{
|
|
MOZ_COUNT_CTOR(FFmpegAudioDecoder);
|
|
// Use a new MediaByteBuffer as the object will be modified during initialization.
|
|
if (aConfig.mCodecSpecificConfig && aConfig.mCodecSpecificConfig->Length()) {
|
|
mExtraData = new MediaByteBuffer;
|
|
mExtraData->AppendElements(*aConfig.mCodecSpecificConfig);
|
|
}
|
|
}
|
|
|
|
RefPtr<MediaDataDecoder::InitPromise>
|
|
FFmpegAudioDecoder<LIBAV_VER>::Init()
|
|
{
|
|
nsresult rv = InitDecoder();
|
|
|
|
return rv == NS_OK ? InitPromise::CreateAndResolve(TrackInfo::kAudioTrack, __func__)
|
|
: InitPromise::CreateAndReject(NS_ERROR_DOM_MEDIA_FATAL_ERR, __func__);
|
|
}
|
|
|
|
void
|
|
FFmpegAudioDecoder<LIBAV_VER>::InitCodecContext()
|
|
{
|
|
MOZ_ASSERT(mCodecContext);
|
|
// We do not want to set this value to 0 as FFmpeg by default will
|
|
// use the number of cores, which with our mozlibavutil get_cpu_count
|
|
// isn't implemented.
|
|
mCodecContext->thread_count = 1;
|
|
// 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
|
|
CopyAndPackAudio(AVFrame* aFrame, uint32_t aNumChannels, uint32_t aNumAFrames)
|
|
{
|
|
MOZ_ASSERT(aNumChannels <= MAX_CHANNELS);
|
|
|
|
AlignedAudioBuffer audio(aNumChannels * aNumAFrames);
|
|
if (!audio) {
|
|
return audio;
|
|
}
|
|
|
|
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->extended_data);
|
|
for (uint32_t frame = 0; frame < aNumAFrames; frame++) {
|
|
for (uint32_t channel = 0; channel < aNumChannels; channel++) {
|
|
*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];
|
|
for (uint32_t frame = 0; frame < aNumAFrames; frame++) {
|
|
for (uint32_t channel = 0; channel < aNumChannels; channel++) {
|
|
*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();
|
|
int16_t** data = reinterpret_cast<int16_t**>(aFrame->extended_data);
|
|
for (uint32_t frame = 0; frame < aNumAFrames; frame++) {
|
|
for (uint32_t channel = 0; channel < aNumChannels; channel++) {
|
|
*tmp++ = AudioSampleToFloat(data[channel][frame]);
|
|
}
|
|
}
|
|
} else if (aFrame->format == AV_SAMPLE_FMT_S32) {
|
|
// Audio data already packed. Need to convert from S16 to 32 bits Float
|
|
AudioDataValue* tmp = audio.get();
|
|
int32_t* data = reinterpret_cast<int32_t**>(aFrame->data)[0];
|
|
for (uint32_t frame = 0; frame < aNumAFrames; frame++) {
|
|
for (uint32_t channel = 0; channel < aNumChannels; channel++) {
|
|
*tmp++ = AudioSampleToFloat(*data++);
|
|
}
|
|
}
|
|
} else if (aFrame->format == AV_SAMPLE_FMT_S32P) {
|
|
// Planar audio data. Convert it from S32 to 32 bits float
|
|
// and pack it into something we can understand.
|
|
AudioDataValue* tmp = audio.get();
|
|
int32_t** data = reinterpret_cast<int32_t**>(aFrame->extended_data);
|
|
for (uint32_t frame = 0; frame < aNumAFrames; frame++) {
|
|
for (uint32_t channel = 0; channel < aNumChannels; channel++) {
|
|
*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;
|
|
}
|
|
|
|
MediaResult
|
|
FFmpegAudioDecoder<LIBAV_VER>::DoDecode(MediaRawData* aSample)
|
|
{
|
|
AVPacket packet;
|
|
mLib->av_init_packet(&packet);
|
|
|
|
packet.data = const_cast<uint8_t*>(aSample->Data());
|
|
packet.size = aSample->Size();
|
|
|
|
if (!PrepareFrame()) {
|
|
return MediaResult(
|
|
NS_ERROR_OUT_OF_MEMORY,
|
|
RESULT_DETAIL("FFmpeg audio decoder failed to allocate frame"));
|
|
}
|
|
|
|
int64_t samplePosition = aSample->mOffset;
|
|
media::TimeUnit pts = media::TimeUnit::FromMicroseconds(aSample->mTime);
|
|
|
|
while (packet.size > 0) {
|
|
int decoded;
|
|
int bytesConsumed =
|
|
mLib->avcodec_decode_audio4(mCodecContext, mFrame, &decoded, &packet);
|
|
|
|
if (bytesConsumed < 0) {
|
|
NS_WARNING("FFmpeg audio decoder error.");
|
|
return MediaResult(NS_ERROR_DOM_MEDIA_DECODE_ERR,
|
|
RESULT_DETAIL("FFmpeg audio error:%d", bytesConsumed));
|
|
}
|
|
|
|
if (decoded) {
|
|
if (mFrame->format != AV_SAMPLE_FMT_FLT &&
|
|
mFrame->format != AV_SAMPLE_FMT_FLTP &&
|
|
mFrame->format != AV_SAMPLE_FMT_S16 &&
|
|
mFrame->format != AV_SAMPLE_FMT_S16P &&
|
|
mFrame->format != AV_SAMPLE_FMT_S32 &&
|
|
mFrame->format != AV_SAMPLE_FMT_S32P) {
|
|
return MediaResult(
|
|
NS_ERROR_DOM_MEDIA_DECODE_ERR,
|
|
RESULT_DETAIL(
|
|
"FFmpeg audio decoder outputs unsupported audio format"));
|
|
}
|
|
uint32_t numChannels = mCodecContext->channels;
|
|
AudioConfig::ChannelLayout layout(numChannels);
|
|
if (!layout.IsValid()) {
|
|
return MediaResult(
|
|
NS_ERROR_DOM_MEDIA_FATAL_ERR,
|
|
RESULT_DETAIL("Unsupported channel layout:%u", numChannels));
|
|
}
|
|
|
|
uint32_t samplingRate = mCodecContext->sample_rate;
|
|
|
|
AlignedAudioBuffer audio =
|
|
CopyAndPackAudio(mFrame, numChannels, mFrame->nb_samples);
|
|
if (!audio) {
|
|
return MediaResult(NS_ERROR_OUT_OF_MEMORY, __func__);
|
|
}
|
|
|
|
media::TimeUnit duration =
|
|
FramesToTimeUnit(mFrame->nb_samples, samplingRate);
|
|
if (!duration.IsValid()) {
|
|
return MediaResult(
|
|
NS_ERROR_DOM_MEDIA_OVERFLOW_ERR,
|
|
RESULT_DETAIL("Invalid sample duration"));
|
|
}
|
|
|
|
RefPtr<AudioData> data = new AudioData(samplePosition,
|
|
pts.ToMicroseconds(),
|
|
duration.ToMicroseconds(),
|
|
mFrame->nb_samples,
|
|
Move(audio),
|
|
numChannels,
|
|
samplingRate);
|
|
mCallback->Output(data);
|
|
pts += duration;
|
|
if (!pts.IsValid()) {
|
|
return MediaResult(
|
|
NS_ERROR_DOM_MEDIA_OVERFLOW_ERR,
|
|
RESULT_DETAIL("Invalid count of accumulated audio samples"));
|
|
}
|
|
}
|
|
packet.data += bytesConsumed;
|
|
packet.size -= bytesConsumed;
|
|
samplePosition += bytesConsumed;
|
|
}
|
|
return NS_OK;
|
|
}
|
|
|
|
void
|
|
FFmpegAudioDecoder<LIBAV_VER>::ProcessDrain()
|
|
{
|
|
ProcessFlush();
|
|
mCallback->DrainComplete();
|
|
}
|
|
|
|
AVCodecID
|
|
FFmpegAudioDecoder<LIBAV_VER>::GetCodecId(const nsACString& aMimeType)
|
|
{
|
|
if (aMimeType.EqualsLiteral("audio/mpeg")) {
|
|
return AV_CODEC_ID_MP3;
|
|
} else if (aMimeType.EqualsLiteral("audio/flac")) {
|
|
return AV_CODEC_ID_FLAC;
|
|
} else if (aMimeType.EqualsLiteral("audio/mp4a-latm")) {
|
|
return AV_CODEC_ID_AAC;
|
|
}
|
|
|
|
return AV_CODEC_ID_NONE;
|
|
}
|
|
|
|
FFmpegAudioDecoder<LIBAV_VER>::~FFmpegAudioDecoder()
|
|
{
|
|
MOZ_COUNT_DTOR(FFmpegAudioDecoder);
|
|
}
|
|
|
|
} // namespace mozilla
|