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

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C++
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/* -*- 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 "OpusDecoder.h"
#include "OpusParser.h"
#include "TimeUnits.h"
#include "VorbisUtils.h"
#include "VorbisDecoder.h" // For VorbisLayout
#include "mozilla/EndianUtils.h"
#include "mozilla/PodOperations.h"
#include "mozilla/SyncRunnable.h"
#include "mozilla/SizePrintfMacros.h"
#include <inttypes.h> // For PRId64
#include "opus/opus.h"
extern "C" {
#include "opus/opus_multistream.h"
}
#define OPUS_DEBUG(arg, ...) MOZ_LOG(sPDMLog, mozilla::LogLevel::Debug, \
("OpusDataDecoder(%p)::%s: " arg, this, __func__, ##__VA_ARGS__))
namespace mozilla {
OpusDataDecoder::OpusDataDecoder(const CreateDecoderParams& aParams)
: mInfo(aParams.AudioConfig())
, mTaskQueue(aParams.mTaskQueue)
, mOpusDecoder(nullptr)
, mSkip(0)
, mDecodedHeader(false)
, mPaddingDiscarded(false)
, mFrames(0)
{
}
OpusDataDecoder::~OpusDataDecoder()
{
if (mOpusDecoder) {
opus_multistream_decoder_destroy(mOpusDecoder);
mOpusDecoder = nullptr;
}
}
RefPtr<ShutdownPromise>
OpusDataDecoder::Shutdown()
{
RefPtr<OpusDataDecoder> self = this;
return InvokeAsync(mTaskQueue, __func__, [self]() {
return ShutdownPromise::CreateAndResolve(true, __func__);
});
}
void
OpusDataDecoder::AppendCodecDelay(MediaByteBuffer* config, uint64_t codecDelayUS)
{
uint8_t buffer[sizeof(uint64_t)];
BigEndian::writeUint64(buffer, codecDelayUS);
config->AppendElements(buffer, sizeof(uint64_t));
}
RefPtr<MediaDataDecoder::InitPromise>
OpusDataDecoder::Init()
{
size_t length = mInfo.mCodecSpecificConfig->Length();
uint8_t *p = mInfo.mCodecSpecificConfig->Elements();
if (length < sizeof(uint64_t)) {
OPUS_DEBUG("CodecSpecificConfig too short to read codecDelay!");
return InitPromise::CreateAndReject(NS_ERROR_DOM_MEDIA_FATAL_ERR, __func__);
}
int64_t codecDelay = BigEndian::readUint64(p);
length -= sizeof(uint64_t);
p += sizeof(uint64_t);
if (NS_FAILED(DecodeHeader(p, length))) {
OPUS_DEBUG("Error decoding header!");
return InitPromise::CreateAndReject(NS_ERROR_DOM_MEDIA_FATAL_ERR, __func__);
}
int r;
mOpusDecoder = opus_multistream_decoder_create(mOpusParser->mRate,
mOpusParser->mChannels,
mOpusParser->mStreams,
mOpusParser->mCoupledStreams,
mMappingTable,
&r);
mSkip = mOpusParser->mPreSkip;
mPaddingDiscarded = false;
if (codecDelay != FramesToUsecs(mOpusParser->mPreSkip,
mOpusParser->mRate).value()) {
NS_WARNING("Invalid Opus header: CodecDelay and pre-skip do not match!");
return InitPromise::CreateAndReject(NS_ERROR_DOM_MEDIA_FATAL_ERR, __func__);
}
if (mInfo.mRate != (uint32_t)mOpusParser->mRate) {
NS_WARNING("Invalid Opus header: container and codec rate do not match!");
}
if (mInfo.mChannels != (uint32_t)mOpusParser->mChannels) {
NS_WARNING("Invalid Opus header: container and codec channels do not match!");
}
return r == OPUS_OK ? InitPromise::CreateAndResolve(TrackInfo::kAudioTrack, __func__)
: InitPromise::CreateAndReject(NS_ERROR_DOM_MEDIA_FATAL_ERR, __func__);
}
nsresult
OpusDataDecoder::DecodeHeader(const unsigned char* aData, size_t aLength)
{
MOZ_ASSERT(!mOpusParser);
MOZ_ASSERT(!mOpusDecoder);
MOZ_ASSERT(!mDecodedHeader);
mDecodedHeader = true;
mOpusParser = new OpusParser;
if (!mOpusParser->DecodeHeader(const_cast<unsigned char*>(aData), aLength)) {
return NS_ERROR_FAILURE;
}
int channels = mOpusParser->mChannels;
AudioConfig::ChannelLayout layout(channels);
if (!layout.IsValid()) {
OPUS_DEBUG("Invalid channel mapping. Source is %d channels", channels);
return NS_ERROR_FAILURE;
}
AudioConfig::ChannelLayout vorbisLayout(
channels, VorbisDataDecoder::VorbisLayout(channels));
AudioConfig::ChannelLayout smpteLayout(channels);
static_assert(sizeof(mOpusParser->mMappingTable) / sizeof(mOpusParser->mMappingTable[0]) >= MAX_AUDIO_CHANNELS,
"Invalid size set");
uint8_t map[sizeof(mOpusParser->mMappingTable) / sizeof(mOpusParser->mMappingTable[0])];
if (vorbisLayout.MappingTable(smpteLayout, map)) {
for (int i = 0; i < channels; i++) {
mMappingTable[i] = mOpusParser->mMappingTable[map[i]];
}
} else {
// Should never get here as vorbis layout is always convertible to SMPTE
// default layout.
PodCopy(mMappingTable, mOpusParser->mMappingTable, MAX_AUDIO_CHANNELS);
}
return NS_OK;
}
RefPtr<MediaDataDecoder::DecodePromise>
OpusDataDecoder::Decode(MediaRawData* aSample)
{
return InvokeAsync<MediaRawData*>(mTaskQueue, this, __func__,
&OpusDataDecoder::ProcessDecode, aSample);
}
RefPtr<MediaDataDecoder::DecodePromise>
OpusDataDecoder::ProcessDecode(MediaRawData* aSample)
{
uint32_t channels = mOpusParser->mChannels;
if (mPaddingDiscarded) {
// Discard padding should be used only on the final packet, so
// decoding after a padding discard is invalid.
OPUS_DEBUG("Opus error, discard padding on interstitial packet");
return DecodePromise::CreateAndReject(
MediaResult(NS_ERROR_DOM_MEDIA_FATAL_ERR,
RESULT_DETAIL("Discard padding on interstitial packet")),
__func__);
}
if (!mLastFrameTime ||
mLastFrameTime.ref() != aSample->mTime.ToMicroseconds()) {
// We are starting a new block.
mFrames = 0;
mLastFrameTime = Some(aSample->mTime.ToMicroseconds());
}
// Maximum value is 63*2880, so there's no chance of overflow.
int frames_number =
opus_packet_get_nb_frames(aSample->Data(), aSample->Size());
if (frames_number <= 0) {
OPUS_DEBUG("Invalid packet header: r=%d length=%" PRIuSIZE, frames_number,
aSample->Size());
return DecodePromise::CreateAndReject(
MediaResult(NS_ERROR_DOM_MEDIA_DECODE_ERR,
RESULT_DETAIL("Invalid packet header: r=%d length=%u",
frames_number, uint32_t(aSample->Size()))),
__func__);
}
int samples = opus_packet_get_samples_per_frame(
aSample->Data(), opus_int32(mOpusParser->mRate));
// A valid Opus packet must be between 2.5 and 120 ms long (48kHz).
CheckedInt32 totalFrames =
CheckedInt32(frames_number) * CheckedInt32(samples);
if (!totalFrames.isValid()) {
return DecodePromise::CreateAndReject(
MediaResult(NS_ERROR_DOM_MEDIA_DECODE_ERR,
RESULT_DETAIL("Frames count overflow")),
__func__);
}
int frames = totalFrames.value();
if (frames < 120 || frames > 5760) {
OPUS_DEBUG("Invalid packet frames: %d", frames);
return DecodePromise::CreateAndReject(
MediaResult(NS_ERROR_DOM_MEDIA_DECODE_ERR,
RESULT_DETAIL("Invalid packet frames:%d", frames)),
__func__);
}
AlignedAudioBuffer buffer(frames * channels);
if (!buffer) {
return DecodePromise::CreateAndReject(
MediaResult(NS_ERROR_OUT_OF_MEMORY, __func__), __func__);
}
// Decode to the appropriate sample type.
#ifdef MOZ_SAMPLE_TYPE_FLOAT32
int ret = opus_multistream_decode_float(mOpusDecoder,
aSample->Data(), aSample->Size(),
buffer.get(), frames, false);
#else
int ret = opus_multistream_decode(mOpusDecoder,
aSample->Data(), aSample->Size(),
buffer.get(), frames, false);
#endif
if (ret < 0) {
return DecodePromise::CreateAndReject(
MediaResult(NS_ERROR_DOM_MEDIA_DECODE_ERR,
RESULT_DETAIL("Opus decoding error:%d", ret)),
__func__);
}
NS_ASSERTION(ret == frames, "Opus decoded too few audio samples");
auto startTime = aSample->mTime;
// Trim the initial frames while the decoder is settling.
if (mSkip > 0) {
int32_t skipFrames = std::min<int32_t>(mSkip, frames);
int32_t keepFrames = frames - skipFrames;
OPUS_DEBUG(
"Opus decoder skipping %d of %d frames", skipFrames, frames);
PodMove(buffer.get(),
buffer.get() + skipFrames * channels,
keepFrames * channels);
startTime = startTime + FramesToTimeUnit(skipFrames, mOpusParser->mRate);
frames = keepFrames;
mSkip -= skipFrames;
}
if (aSample->mDiscardPadding > 0) {
OPUS_DEBUG("Opus decoder discarding %u of %d frames",
aSample->mDiscardPadding, frames);
// Padding discard is only supposed to happen on the final packet.
// Record the discard so we can return an error if another packet is
// decoded.
if (aSample->mDiscardPadding > uint32_t(frames)) {
// Discarding more than the entire packet is invalid.
OPUS_DEBUG("Opus error, discard padding larger than packet");
return DecodePromise::CreateAndReject(
MediaResult(NS_ERROR_DOM_MEDIA_FATAL_ERR,
RESULT_DETAIL("Discard padding larger than packet")),
__func__);
}
mPaddingDiscarded = true;
frames = frames - aSample->mDiscardPadding;
}
// Apply the header gain if one was specified.
#ifdef MOZ_SAMPLE_TYPE_FLOAT32
if (mOpusParser->mGain != 1.0f) {
float gain = mOpusParser->mGain;
uint32_t samples = frames * channels;
for (uint32_t i = 0; i < samples; i++) {
buffer[i] *= gain;
}
}
#else
if (mOpusParser->mGain_Q16 != 65536) {
int64_t gain_Q16 = mOpusParser->mGain_Q16;
uint32_t samples = frames * channels;
for (uint32_t i = 0; i < samples; i++) {
int32_t val = static_cast<int32_t>((gain_Q16*buffer[i] + 32768)>>16);
buffer[i] = static_cast<AudioDataValue>(MOZ_CLIP_TO_15(val));
}
}
#endif
auto duration = FramesToTimeUnit(frames, mOpusParser->mRate);
if (!duration.IsValid()) {
return DecodePromise::CreateAndReject(
MediaResult(NS_ERROR_DOM_MEDIA_OVERFLOW_ERR,
RESULT_DETAIL("Overflow converting WebM audio duration")),
__func__);
}
auto time = startTime -
FramesToTimeUnit(mOpusParser->mPreSkip, mOpusParser->mRate) +
FramesToTimeUnit(mFrames, mOpusParser->mRate);
if (!time.IsValid()) {
return DecodePromise::CreateAndReject(
MediaResult(NS_ERROR_DOM_MEDIA_OVERFLOW_ERR,
RESULT_DETAIL("Overflow shifting tstamp by codec delay")),
__func__);
};
mFrames += frames;
return DecodePromise::CreateAndResolve(
DecodedData{ new AudioData(aSample->mOffset, time, duration,
frames, Move(buffer), mOpusParser->mChannels,
mOpusParser->mRate) },
__func__);
}
RefPtr<MediaDataDecoder::DecodePromise>
OpusDataDecoder::Drain()
{
RefPtr<OpusDataDecoder> self = this;
// InvokeAsync dispatches a task that will be run after any pending decode
// completes. As such, once the drain task run, there's nothing more to do.
return InvokeAsync(mTaskQueue, __func__, [] {
return DecodePromise::CreateAndResolve(DecodedData(), __func__);
});
}
RefPtr<MediaDataDecoder::FlushPromise>
OpusDataDecoder::Flush()
{
if (!mOpusDecoder) {
return FlushPromise::CreateAndResolve(true, __func__);
}
RefPtr<OpusDataDecoder> self = this;
return InvokeAsync(mTaskQueue, __func__, [self, this]() {
MOZ_ASSERT(mOpusDecoder);
// Reset the decoder.
opus_multistream_decoder_ctl(mOpusDecoder, OPUS_RESET_STATE);
mSkip = mOpusParser->mPreSkip;
mPaddingDiscarded = false;
mLastFrameTime.reset();
return FlushPromise::CreateAndResolve(true, __func__);
});
}
/* static */
bool
OpusDataDecoder::IsOpus(const nsACString& aMimeType)
{
return aMimeType.EqualsLiteral("audio/opus");
}
} // namespace mozilla
#undef OPUS_DEBUG
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