gecko-dev/dom/media/platforms/agnostic/VorbisDecoder.cpp

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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 "VorbisDecoder.h"
#include "VorbisUtils.h"
#include "XiphExtradata.h"
#include "mozilla/PodOperations.h"
#include "mozilla/SyncRunnable.h"
#undef LOG
#define LOG(type, msg) MOZ_LOG(sPDMLog, type, msg)
namespace mozilla {
ogg_packet InitVorbisPacket(const unsigned char* aData, size_t aLength,
bool aBOS, bool aEOS,
int64_t aGranulepos, int64_t aPacketNo)
{
ogg_packet packet;
packet.packet = const_cast<unsigned char*>(aData);
packet.bytes = aLength;
packet.b_o_s = aBOS;
packet.e_o_s = aEOS;
packet.granulepos = aGranulepos;
packet.packetno = aPacketNo;
return packet;
}
VorbisDataDecoder::VorbisDataDecoder(const CreateDecoderParams& aParams)
: mInfo(aParams.AudioConfig())
, mTaskQueue(aParams.mTaskQueue)
, mPacketCount(0)
, mFrames(0)
{
// Zero these member vars to avoid crashes in Vorbis clear functions when
// destructor is called before |Init|.
PodZero(&mVorbisBlock);
PodZero(&mVorbisDsp);
PodZero(&mVorbisInfo);
PodZero(&mVorbisComment);
}
VorbisDataDecoder::~VorbisDataDecoder()
{
vorbis_block_clear(&mVorbisBlock);
vorbis_dsp_clear(&mVorbisDsp);
vorbis_info_clear(&mVorbisInfo);
vorbis_comment_clear(&mVorbisComment);
}
RefPtr<ShutdownPromise>
VorbisDataDecoder::Shutdown()
{
RefPtr<VorbisDataDecoder> self = this;
return InvokeAsync(mTaskQueue, __func__, [self]() {
return ShutdownPromise::CreateAndResolve(true, __func__);
});
}
RefPtr<MediaDataDecoder::InitPromise>
VorbisDataDecoder::Init()
{
vorbis_info_init(&mVorbisInfo);
vorbis_comment_init(&mVorbisComment);
PodZero(&mVorbisDsp);
PodZero(&mVorbisBlock);
AutoTArray<unsigned char*,4> headers;
AutoTArray<size_t,4> headerLens;
if (!XiphExtradataToHeaders(headers, headerLens,
mInfo.mCodecSpecificConfig->Elements(),
mInfo.mCodecSpecificConfig->Length())) {
return InitPromise::CreateAndReject(NS_ERROR_DOM_MEDIA_FATAL_ERR, __func__);
}
for (size_t i = 0; i < headers.Length(); i++) {
if (NS_FAILED(DecodeHeader(headers[i], headerLens[i]))) {
return InitPromise::CreateAndReject(NS_ERROR_DOM_MEDIA_FATAL_ERR, __func__);
}
}
MOZ_ASSERT(mPacketCount == 3);
int r = vorbis_synthesis_init(&mVorbisDsp, &mVorbisInfo);
if (r) {
return InitPromise::CreateAndReject(NS_ERROR_DOM_MEDIA_FATAL_ERR, __func__);
}
r = vorbis_block_init(&mVorbisDsp, &mVorbisBlock);
if (r) {
return InitPromise::CreateAndReject(NS_ERROR_DOM_MEDIA_FATAL_ERR, __func__);
}
if (mInfo.mRate != (uint32_t)mVorbisDsp.vi->rate) {
LOG(LogLevel::Warning,
("Invalid Vorbis header: container and codec rate do not match!"));
}
if (mInfo.mChannels != (uint32_t)mVorbisDsp.vi->channels) {
LOG(LogLevel::Warning,
("Invalid Vorbis header: container and codec channels do not match!"));
}
AudioConfig::ChannelLayout layout(mVorbisDsp.vi->channels);
if (!layout.IsValid()) {
return InitPromise::CreateAndReject(NS_ERROR_DOM_MEDIA_FATAL_ERR, __func__);
}
return InitPromise::CreateAndResolve(TrackInfo::kAudioTrack, __func__);
}
nsresult
VorbisDataDecoder::DecodeHeader(const unsigned char* aData, size_t aLength)
{
bool bos = mPacketCount == 0;
ogg_packet pkt = InitVorbisPacket(aData, aLength, bos, false, 0, mPacketCount++);
MOZ_ASSERT(mPacketCount <= 3);
int r = vorbis_synthesis_headerin(&mVorbisInfo,
&mVorbisComment,
&pkt);
return r == 0 ? NS_OK : NS_ERROR_FAILURE;
}
RefPtr<MediaDataDecoder::DecodePromise>
VorbisDataDecoder::Decode(MediaRawData* aSample)
{
return InvokeAsync<MediaRawData*>(mTaskQueue, this, __func__,
&VorbisDataDecoder::ProcessDecode, aSample);
}
RefPtr<MediaDataDecoder::DecodePromise>
VorbisDataDecoder::ProcessDecode(MediaRawData* aSample)
{
MOZ_ASSERT(mTaskQueue->IsCurrentThreadIn());
const unsigned char* aData = aSample->Data();
size_t aLength = aSample->Size();
int64_t aOffset = aSample->mOffset;
int64_t aTstampUsecs = aSample->mTime;
int64_t aTotalFrames = 0;
MOZ_ASSERT(mPacketCount >= 3);
if (!mLastFrameTime || mLastFrameTime.ref() != aSample->mTime) {
// We are starting a new block.
mFrames = 0;
mLastFrameTime = Some(aSample->mTime);
}
ogg_packet pkt = InitVorbisPacket(aData, aLength, false, aSample->mEOS,
aSample->mTimecode, mPacketCount++);
int err = vorbis_synthesis(&mVorbisBlock, &pkt);
if (err) {
return DecodePromise::CreateAndReject(
MediaResult(NS_ERROR_DOM_MEDIA_DECODE_ERR,
RESULT_DETAIL("vorbis_synthesis:%d", err)),
__func__);
}
err = vorbis_synthesis_blockin(&mVorbisDsp, &mVorbisBlock);
if (err) {
return DecodePromise::CreateAndReject(
MediaResult(NS_ERROR_DOM_MEDIA_DECODE_ERR,
RESULT_DETAIL("vorbis_synthesis_blockin:%d", err)),
__func__);
}
VorbisPCMValue** pcm = 0;
int32_t frames = vorbis_synthesis_pcmout(&mVorbisDsp, &pcm);
if (frames == 0) {
return DecodePromise::CreateAndResolve(DecodedData(), __func__);
}
DecodedData results;
while (frames > 0) {
uint32_t channels = mVorbisDsp.vi->channels;
uint32_t rate = mVorbisDsp.vi->rate;
AlignedAudioBuffer buffer(frames*channels);
if (!buffer) {
return DecodePromise::CreateAndReject(
MediaResult(NS_ERROR_OUT_OF_MEMORY, __func__), __func__);
}
for (uint32_t j = 0; j < channels; ++j) {
VorbisPCMValue* channel = pcm[j];
for (uint32_t i = 0; i < uint32_t(frames); ++i) {
buffer[i*channels + j] = MOZ_CONVERT_VORBIS_SAMPLE(channel[i]);
}
}
CheckedInt64 duration = FramesToUsecs(frames, rate);
if (!duration.isValid()) {
return DecodePromise::CreateAndReject(
MediaResult(NS_ERROR_DOM_MEDIA_OVERFLOW_ERR,
RESULT_DETAIL("Overflow converting audio duration")),
__func__);
}
CheckedInt64 total_duration = FramesToUsecs(mFrames, rate);
if (!total_duration.isValid()) {
return DecodePromise::CreateAndReject(
MediaResult(NS_ERROR_DOM_MEDIA_OVERFLOW_ERR,
RESULT_DETAIL("Overflow converting audio total_duration")),
__func__);
}
CheckedInt64 time = total_duration + aTstampUsecs;
if (!time.isValid()) {
return DecodePromise::CreateAndReject(
MediaResult(
NS_ERROR_DOM_MEDIA_OVERFLOW_ERR,
RESULT_DETAIL("Overflow adding total_duration and aTstampUsecs")),
__func__);
};
if (!mAudioConverter) {
AudioConfig in(AudioConfig::ChannelLayout(channels, VorbisLayout(channels)),
rate);
AudioConfig out(channels, rate);
if (!in.IsValid() || !out.IsValid()) {
return DecodePromise::CreateAndReject(
MediaResult(NS_ERROR_DOM_MEDIA_FATAL_ERR,
RESULT_DETAIL("Invalid channel layout:%u", channels)),
__func__);
}
mAudioConverter = MakeUnique<AudioConverter>(in, out);
}
MOZ_ASSERT(mAudioConverter->CanWorkInPlace());
AudioSampleBuffer data(Move(buffer));
data = mAudioConverter->Process(Move(data));
aTotalFrames += frames;
results.AppendElement(new AudioData(aOffset, time.value(), duration.value(),
frames, data.Forget(), channels, rate));
mFrames += frames;
err = vorbis_synthesis_read(&mVorbisDsp, frames);
if (err) {
return DecodePromise::CreateAndReject(
MediaResult(NS_ERROR_DOM_MEDIA_DECODE_ERR,
RESULT_DETAIL("vorbis_synthesis_read:%d", err)),
__func__);
}
frames = vorbis_synthesis_pcmout(&mVorbisDsp, &pcm);
}
return DecodePromise::CreateAndResolve(Move(results), __func__);
}
RefPtr<MediaDataDecoder::DecodePromise>
VorbisDataDecoder::Drain()
{
return InvokeAsync(mTaskQueue, __func__, [] {
return DecodePromise::CreateAndResolve(DecodedData(), __func__);
});
}
RefPtr<MediaDataDecoder::FlushPromise>
VorbisDataDecoder::Flush()
{
RefPtr<VorbisDataDecoder> self = this;
return InvokeAsync(mTaskQueue, __func__, [self, this]() {
// Ignore failed results from vorbis_synthesis_restart. They
// aren't fatal and it fails when ResetDecode is called at a
// time when no vorbis data has been read.
vorbis_synthesis_restart(&mVorbisDsp);
mLastFrameTime.reset();
return FlushPromise::CreateAndResolve(true, __func__);
});
}
/* static */
bool
VorbisDataDecoder::IsVorbis(const nsACString& aMimeType)
{
return aMimeType.EqualsLiteral("audio/vorbis");
}
/* static */ const AudioConfig::Channel*
VorbisDataDecoder::VorbisLayout(uint32_t aChannels)
{
// From https://www.xiph.org/vorbis/doc/Vorbis_I_spec.html
// Section 4.3.9.
typedef AudioConfig::Channel Channel;
switch (aChannels) {
case 1: // the stream is monophonic
{
static const Channel config[] = { AudioConfig::CHANNEL_MONO };
return config;
}
case 2: // the stream is stereo. channel order: left, right
{
static const Channel config[] = { AudioConfig::CHANNEL_LEFT, AudioConfig::CHANNEL_RIGHT };
return config;
}
case 3: // the stream is a 1d-surround encoding. channel order: left, center, right
{
static const Channel config[] = { AudioConfig::CHANNEL_LEFT, AudioConfig::CHANNEL_CENTER, AudioConfig::CHANNEL_RIGHT };
return config;
}
case 4: // the stream is quadraphonic surround. channel order: front left, front right, rear left, rear right
{
static const Channel config[] = { AudioConfig::CHANNEL_LEFT, AudioConfig::CHANNEL_RIGHT, AudioConfig::CHANNEL_LS, AudioConfig::CHANNEL_RS };
return config;
}
case 5: // the stream is five-channel surround. channel order: front left, center, front right, rear left, rear right
{
static const Channel config[] = { AudioConfig::CHANNEL_LEFT, AudioConfig::CHANNEL_CENTER, AudioConfig::CHANNEL_RIGHT, AudioConfig::CHANNEL_LS, AudioConfig::CHANNEL_RS };
return config;
}
case 6: // the stream is 5.1 surround. channel order: front left, center, front right, rear left, rear right, LFE
{
static const Channel config[] = { AudioConfig::CHANNEL_LEFT, AudioConfig::CHANNEL_CENTER, AudioConfig::CHANNEL_RIGHT, AudioConfig::CHANNEL_LS, AudioConfig::CHANNEL_RS, AudioConfig::CHANNEL_LFE };
return config;
}
case 7: // surround. channel order: front left, center, front right, side left, side right, rear center, LFE
{
static const Channel config[] = { AudioConfig::CHANNEL_LEFT, AudioConfig::CHANNEL_CENTER, AudioConfig::CHANNEL_RIGHT, AudioConfig::CHANNEL_LS, AudioConfig::CHANNEL_RS, AudioConfig::CHANNEL_RCENTER, AudioConfig::CHANNEL_LFE };
return config;
}
case 8: // the stream is 7.1 surround. channel order: front left, center, front right, side left, side right, rear left, rear right, LFE
{
static const Channel config[] = { AudioConfig::CHANNEL_LEFT, AudioConfig::CHANNEL_CENTER, AudioConfig::CHANNEL_RIGHT, AudioConfig::CHANNEL_LS, AudioConfig::CHANNEL_RS, AudioConfig::CHANNEL_RLS, AudioConfig::CHANNEL_RRS, AudioConfig::CHANNEL_LFE };
return config;
}
default:
return nullptr;
}
}
} // namespace mozilla
#undef LOG