Files
palemoon27/dom/media/platforms/ffmpeg/FFmpegAudioDecoder.cpp
T
trav90 95a8f149db Have specialized AVCodecContext initialization
A common initialization procedure is too restrictive, and we end up setting up audio configuration for video decoding and vice-versa.
2018-07-25 06:22:41 +08:00

201 lines
6.0 KiB
C++

/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
/* vim:set ts=2 sw=2 sts=2 et cindent: */
/* 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 "MediaTaskQueue.h"
#include "FFmpegRuntimeLinker.h"
#include "FFmpegAudioDecoder.h"
#define MAX_CHANNELS 16
namespace mozilla
{
FFmpegAudioDecoder<LIBAV_VER>::FFmpegAudioDecoder(
FlushableMediaTaskQueue* aTaskQueue, MediaDataDecoderCallback* aCallback,
const AudioInfo& aConfig)
: FFmpegDataDecoder(aTaskQueue, GetCodecId(aConfig.mMimeType))
, mCallback(aCallback)
{
MOZ_COUNT_CTOR(FFmpegAudioDecoder);
// Use a new MediaByteBuffer as the object will be modified during initialization.
mExtraData = new MediaByteBuffer;
mExtraData->AppendElements(*aConfig.mCodecSpecificConfig);
}
nsresult
FFmpegAudioDecoder<LIBAV_VER>::Init()
{
nsresult rv = FFmpegDataDecoder::Init();
NS_ENSURE_SUCCESS(rv, rv);
return NS_OK;
}
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.
uint32_t major, minor;
FFmpegRuntimeLinker::GetVersion(major, minor);
// LibAV 0.8 produces rubbish float interleaved samples, request 16 bits audio.
mCodecContext->request_sample_fmt =
(major == 53) ? AV_SAMPLE_FMT_S16 : AV_SAMPLE_FMT_FLT;
}
static AudioDataValue*
CopyAndPackAudio(AVFrame* aFrame, uint32_t aNumChannels, uint32_t aNumAFrames)
{
MOZ_ASSERT(aNumChannels <= MAX_CHANNELS);
nsAutoArrayPtr<AudioDataValue> audio(
new AudioDataValue[aNumChannels * aNumAFrames]);
if (aFrame->format == AV_SAMPLE_FMT_FLT) {
// Audio data already packed. No need to do anything other than copy it
// into a buffer we own.
memcpy(audio, aFrame->data[0],
aNumChannels * aNumAFrames * sizeof(AudioDataValue));
} else if (aFrame->format == AV_SAMPLE_FMT_FLTP) {
// Planar audio data. Pack it into something we can understand.
AudioDataValue* tmp = audio;
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];
}
}
} else if (aFrame->format == AV_SAMPLE_FMT_S16) {
// Audio data already packed. Need to convert from S16 to 32 bits Float
AudioDataValue* tmp = audio;
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 if (aFrame->format == AV_SAMPLE_FMT_S16P) {
// Planar audio data. Convert it from S16 to 32 bits float
// and pack it into something we can understand.
AudioDataValue* tmp = audio;
int16_t** data = reinterpret_cast<int16_t**>(aFrame->data);
for (uint32_t frame = 0; frame < aNumAFrames; frame++) {
for (uint32_t channel = 0; channel < aNumChannels; channel++) {
*tmp++ = AudioSampleToFloat(data[channel][frame]);
}
}
}
return audio.forget();
}
void
FFmpegAudioDecoder<LIBAV_VER>::DecodePacket(MediaRawData* aSample)
{
AVPacket packet;
av_init_packet(&packet);
packet.data = const_cast<uint8_t*>(aSample->Data());
packet.size = aSample->Size();
if (!PrepareFrame()) {
NS_WARNING("FFmpeg audio decoder failed to allocate frame.");
mCallback->Error();
return;
}
int64_t samplePosition = aSample->mOffset;
Microseconds pts = aSample->mTime;
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;
}
nsRefPtr<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;
}
if (mTaskQueue->IsEmpty()) {
mCallback->InputExhausted();
}
}
nsresult
FFmpegAudioDecoder<LIBAV_VER>::Input(MediaRawData* aSample)
{
mTaskQueue->Dispatch(NS_NewRunnableMethodWithArg<nsRefPtr<MediaRawData> >(
this, &FFmpegAudioDecoder::DecodePacket, nsRefPtr<MediaRawData>(aSample)));
return NS_OK;
}
nsresult
FFmpegAudioDecoder<LIBAV_VER>::Drain()
{
mTaskQueue->AwaitIdle();
mCallback->DrainComplete();
return Flush();
}
AVCodecID
FFmpegAudioDecoder<LIBAV_VER>::GetCodecId(const nsACString& aMimeType)
{
if (aMimeType.EqualsLiteral("audio/mpeg")) {
return AV_CODEC_ID_MP3;
}
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