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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
commit | 6bf0a5cb5034a7e684dcc3500e841785237ce2dd (patch) | |
tree | a68f146d7fa01f0134297619fbe7e33db084e0aa /dom/media/platforms/agnostic/OpusDecoder.cpp | |
parent | Initial commit. (diff) | |
download | thunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.tar.xz thunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.zip |
Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'dom/media/platforms/agnostic/OpusDecoder.cpp')
-rw-r--r-- | dom/media/platforms/agnostic/OpusDecoder.cpp | 380 |
1 files changed, 380 insertions, 0 deletions
diff --git a/dom/media/platforms/agnostic/OpusDecoder.cpp b/dom/media/platforms/agnostic/OpusDecoder.cpp new file mode 100644 index 0000000000..715fa848dc --- /dev/null +++ b/dom/media/platforms/agnostic/OpusDecoder.cpp @@ -0,0 +1,380 @@ +/* -*- 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 <inttypes.h> // For PRId64 + +#include "OpusParser.h" +#include "TimeUnits.h" +#include "VideoUtils.h" +#include "VorbisDecoder.h" // For VorbisLayout +#include "VorbisUtils.h" +#include "mozilla/EndianUtils.h" +#include "mozilla/PodOperations.h" +#include "mozilla/SyncRunnable.h" +#include "opus/opus.h" +extern "C" { +#include "opus/opus_multistream.h" +} + +#define OPUS_DEBUG(arg, ...) \ + DDMOZ_LOG(sPDMLog, mozilla::LogLevel::Debug, "::%s: " arg, __func__, \ + ##__VA_ARGS__) + +namespace mozilla { + +OpusDataDecoder::OpusDataDecoder(const CreateDecoderParams& aParams) + : mInfo(aParams.AudioConfig()), + mOpusDecoder(nullptr), + mSkip(0), + mDecodedHeader(false), + mPaddingDiscarded(false), + mFrames(0), + mChannelMap(AudioConfig::ChannelLayout::UNKNOWN_MAP), + mDefaultPlaybackDeviceMono(aParams.mOptions.contains( + CreateDecoderParams::Option::DefaultPlaybackDeviceMono)) {} + +OpusDataDecoder::~OpusDataDecoder() { + if (mOpusDecoder) { + opus_multistream_decoder_destroy(mOpusDecoder); + mOpusDecoder = nullptr; + } +} + +RefPtr<ShutdownPromise> OpusDataDecoder::Shutdown() { + // mThread may not be set if Init hasn't been called first. + MOZ_ASSERT(!mThread || mThread->IsOnCurrentThread()); + return ShutdownPromise::CreateAndResolve(true, __func__); +} + +RefPtr<MediaDataDecoder::InitPromise> OpusDataDecoder::Init() { + mThread = GetCurrentSerialEventTarget(); + if (!mInfo.mCodecSpecificConfig.is<OpusCodecSpecificData>()) { + MOZ_ASSERT_UNREACHABLE(); + OPUS_DEBUG("Opus decoder got non-opus codec specific data"); + return InitPromise::CreateAndReject( + MediaResult( + NS_ERROR_DOM_MEDIA_FATAL_ERR, + RESULT_DETAIL("Opus decoder got non-opus codec specific data!")), + __func__); + } + const OpusCodecSpecificData opusCodecSpecificData = + mInfo.mCodecSpecificConfig.as<OpusCodecSpecificData>(); + RefPtr<MediaByteBuffer> opusHeaderBlob = + opusCodecSpecificData.mHeadersBinaryBlob; + size_t length = opusHeaderBlob->Length(); + uint8_t* p = opusHeaderBlob->Elements(); + if (NS_FAILED(DecodeHeader(p, length))) { + OPUS_DEBUG("Error decoding header!"); + return InitPromise::CreateAndReject( + MediaResult(NS_ERROR_DOM_MEDIA_FATAL_ERR, + RESULT_DETAIL("Error decoding header!")), + __func__); + } + + MOZ_ASSERT(mMappingTable.Length() >= uint32_t(mOpusParser->mChannels)); + int r; + mOpusDecoder = opus_multistream_decoder_create( + mOpusParser->mRate, mOpusParser->mChannels, mOpusParser->mStreams, + mOpusParser->mCoupledStreams, mMappingTable.Elements(), &r); + + if (!mOpusDecoder) { + OPUS_DEBUG("Error creating decoder!"); + return InitPromise::CreateAndReject( + MediaResult(NS_ERROR_DOM_MEDIA_FATAL_ERR, + RESULT_DETAIL("Error creating decoder!")), + __func__); + } + + // Opus has a special feature for stereo coding where it represent wide + // stereo channels by 180-degree out of phase. This improves quality, but + // needs to be disabled when the output is downmixed to mono. Playback number + // of channels are set in AudioSink, using the same method + // `DecideAudioPlaybackChannels()`, and triggers downmix if needed. + if (mDefaultPlaybackDeviceMono || DecideAudioPlaybackChannels(mInfo) == 1) { + opus_multistream_decoder_ctl(mOpusDecoder, + OPUS_SET_PHASE_INVERSION_DISABLED(1)); + } + + mSkip = mOpusParser->mPreSkip; + mPaddingDiscarded = false; + + if (opusCodecSpecificData.mContainerCodecDelayMicroSeconds != + FramesToUsecs(mOpusParser->mPreSkip, mOpusParser->mRate).value()) { + NS_WARNING( + "Invalid Opus header: container CodecDelay and Opus pre-skip do not " + "match!"); + } + OPUS_DEBUG("Opus preskip in extradata: %" PRId64 "us", + opusCodecSpecificData.mContainerCodecDelayMicroSeconds); + + 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( + MediaResult( + NS_ERROR_DOM_MEDIA_FATAL_ERR, + RESULT_DETAIL( + "could not create opus multistream decoder!")), + __func__); +} + +nsresult OpusDataDecoder::DecodeHeader(const unsigned char* aData, + size_t aLength) { + MOZ_ASSERT(!mOpusParser); + MOZ_ASSERT(!mOpusDecoder); + MOZ_ASSERT(!mDecodedHeader); + mDecodedHeader = true; + + mOpusParser = MakeUnique<OpusParser>(); + if (!mOpusParser->DecodeHeader(const_cast<unsigned char*>(aData), aLength)) { + return NS_ERROR_FAILURE; + } + int channels = mOpusParser->mChannels; + + mMappingTable.SetLength(channels); + AudioConfig::ChannelLayout vorbisLayout( + channels, VorbisDataDecoder::VorbisLayout(channels)); + if (vorbisLayout.IsValid()) { + mChannelMap = vorbisLayout.Map(); + + AudioConfig::ChannelLayout smpteLayout( + AudioConfig::ChannelLayout::SMPTEDefault(vorbisLayout)); + + AutoTArray<uint8_t, 8> map; + map.SetLength(channels); + if (mOpusParser->mChannelMapping == 1 && + vorbisLayout.MappingTable(smpteLayout, &map)) { + for (int i = 0; i < channels; i++) { + mMappingTable[i] = mOpusParser->mMappingTable[map[i]]; + } + } else { + // Use Opus set channel mapping and return channels as-is. + PodCopy(mMappingTable.Elements(), mOpusParser->mMappingTable, channels); + } + } else { + // Create a dummy mapping table so that channel ordering stay the same + // during decoding. + for (int i = 0; i < channels; i++) { + mMappingTable[i] = i; + } + } + + return NS_OK; +} + +RefPtr<MediaDataDecoder::DecodePromise> OpusDataDecoder::Decode( + MediaRawData* aSample) { + MOZ_ASSERT(mThread->IsOnCurrentThread()); + PROCESS_DECODE_LOG(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=%zu", 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; + + OPUS_DEBUG("Decoding frames: [%lf, %lf]", aSample->mTime.ToSeconds(), + aSample->GetEndTime().ToSeconds()); + + // 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 trimming %d of %d frames", skipFrames, frames); + PodMove(buffer.get(), buffer.get() + skipFrames * channels, + keepFrames * channels); + startTime = startTime + media::TimeUnit(skipFrames, mOpusParser->mRate); + frames = keepFrames; + mSkip -= skipFrames; + aSample->mTime += media::TimeUnit(skipFrames, 48000); + aSample->mDuration -= media::TimeUnit(skipFrames, 48000); + OPUS_DEBUG("Adjusted frame after trimming pre-roll: [%lf, %lf]", + aSample->mTime.ToSeconds(), aSample->GetEndTime().ToSeconds()); + } + + 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 = media::TimeUnit(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 - + media::TimeUnit(mOpusParser->mPreSkip, mOpusParser->mRate) + + media::TimeUnit(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; + mTotalFrames += frames; + + OPUS_DEBUG("Total frames so far: %" PRId64, mTotalFrames); + + if (!frames) { + return DecodePromise::CreateAndResolve(DecodedData(), __func__); + } + + // Trim extra allocated frames. + buffer.SetLength(frames * channels); + + return DecodePromise::CreateAndResolve( + DecodedData{new AudioData(aSample->mOffset, time, std::move(buffer), + mOpusParser->mChannels, mOpusParser->mRate, + mChannelMap)}, + __func__); +} + +RefPtr<MediaDataDecoder::DecodePromise> OpusDataDecoder::Drain() { + MOZ_ASSERT(mThread->IsOnCurrentThread()); + return DecodePromise::CreateAndResolve(DecodedData(), __func__); +} + +RefPtr<MediaDataDecoder::FlushPromise> OpusDataDecoder::Flush() { + MOZ_ASSERT(mThread->IsOnCurrentThread()); + if (!mOpusDecoder) { + return FlushPromise::CreateAndResolve(true, __func__); + } + + 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 |