summaryrefslogtreecommitdiffstats
path: root/dom/media/platforms/agnostic/OpusDecoder.cpp
diff options
context:
space:
mode:
authorMatt A. Tobin <mattatobin@localhost.localdomain>2018-02-02 04:16:08 -0500
committerMatt A. Tobin <mattatobin@localhost.localdomain>2018-02-02 04:16:08 -0500
commit5f8de423f190bbb79a62f804151bc24824fa32d8 (patch)
tree10027f336435511475e392454359edea8e25895d /dom/media/platforms/agnostic/OpusDecoder.cpp
parent49ee0794b5d912db1f95dce6eb52d781dc210db5 (diff)
downloadUXP-5f8de423f190bbb79a62f804151bc24824fa32d8.tar
UXP-5f8de423f190bbb79a62f804151bc24824fa32d8.tar.gz
UXP-5f8de423f190bbb79a62f804151bc24824fa32d8.tar.lz
UXP-5f8de423f190bbb79a62f804151bc24824fa32d8.tar.xz
UXP-5f8de423f190bbb79a62f804151bc24824fa32d8.zip
Add m-esr52 at 52.6.0
Diffstat (limited to 'dom/media/platforms/agnostic/OpusDecoder.cpp')
-rw-r--r--dom/media/platforms/agnostic/OpusDecoder.cpp356
1 files changed, 356 insertions, 0 deletions
diff --git a/dom/media/platforms/agnostic/OpusDecoder.cpp b/dom/media/platforms/agnostic/OpusDecoder.cpp
new file mode 100644
index 000000000..9163ed058
--- /dev/null
+++ b/dom/media/platforms/agnostic/OpusDecoder.cpp
@@ -0,0 +1,356 @@
+/* -*- 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 <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)
+ , mCallback(aParams.mCallback)
+ , mOpusDecoder(nullptr)
+ , mSkip(0)
+ , mDecodedHeader(false)
+ , mPaddingDiscarded(false)
+ , mFrames(0)
+ , mIsFlushing(false)
+{
+}
+
+OpusDataDecoder::~OpusDataDecoder()
+{
+ if (mOpusDecoder) {
+ opus_multistream_decoder_destroy(mOpusDecoder);
+ mOpusDecoder = nullptr;
+ }
+}
+
+void
+OpusDataDecoder::Shutdown()
+{
+}
+
+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;
+}
+
+void
+OpusDataDecoder::Input(MediaRawData* aSample)
+{
+ mTaskQueue->Dispatch(NewRunnableMethod<RefPtr<MediaRawData>>(
+ this, &OpusDataDecoder::ProcessDecode, aSample));
+}
+
+void
+OpusDataDecoder::ProcessDecode(MediaRawData* aSample)
+{
+ if (mIsFlushing) {
+ return;
+ }
+
+ MediaResult rv = DoDecode(aSample);
+ if (NS_FAILED(rv)) {
+ mCallback->Error(rv);
+ return;
+ }
+ mCallback->InputExhausted();
+}
+
+MediaResult
+OpusDataDecoder::DoDecode(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 MediaResult(
+ NS_ERROR_DOM_MEDIA_FATAL_ERR,
+ RESULT_DETAIL("Discard padding on interstitial packet"));
+ }
+
+ if (!mLastFrameTime || mLastFrameTime.ref() != aSample->mTime) {
+ // We are starting a new block.
+ mFrames = 0;
+ mLastFrameTime = Some(aSample->mTime);
+ }
+
+ // 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=%u",
+ frames_number, uint32_t(aSample->Size()));
+ return MediaResult(NS_ERROR_DOM_MEDIA_DECODE_ERR,
+ RESULT_DETAIL("Invalid packet header: r=%d length=%u",
+ frames_number, uint32_t(aSample->Size())));
+ }
+
+ 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 MediaResult(NS_ERROR_DOM_MEDIA_DECODE_ERR,
+ RESULT_DETAIL("Frames count overflow"));
+ }
+
+ int frames = totalFrames.value();
+ if (frames < 120 || frames > 5760) {
+ OPUS_DEBUG("Invalid packet frames: %d", frames);
+ return MediaResult(NS_ERROR_DOM_MEDIA_DECODE_ERR,
+ RESULT_DETAIL("Invalid packet frames:%d", frames));
+ }
+
+ AlignedAudioBuffer buffer(frames * channels);
+ if (!buffer) {
+ return MediaResult(NS_ERROR_OUT_OF_MEMORY, __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 MediaResult(NS_ERROR_DOM_MEDIA_DECODE_ERR,
+ RESULT_DETAIL("Opus decoding error:%d", ret));
+ }
+ NS_ASSERTION(ret == frames, "Opus decoded too few audio samples");
+ CheckedInt64 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 + FramesToUsecs(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 MediaResult(NS_ERROR_DOM_MEDIA_FATAL_ERR,
+ RESULT_DETAIL("Discard padding larger than packet"));
+ }
+
+ 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
+
+ CheckedInt64 duration = FramesToUsecs(frames, mOpusParser->mRate);
+ if (!duration.isValid()) {
+ return MediaResult(
+ NS_ERROR_DOM_MEDIA_OVERFLOW_ERR,
+ RESULT_DETAIL("Overflow converting WebM audio duration"));
+ }
+ CheckedInt64 time =
+ startTime - FramesToUsecs(mOpusParser->mPreSkip, mOpusParser->mRate) +
+ FramesToUsecs(mFrames, mOpusParser->mRate);
+ if (!time.isValid()) {
+ return MediaResult(
+ NS_ERROR_DOM_MEDIA_OVERFLOW_ERR,
+ RESULT_DETAIL("Overflow shifting tstamp by codec delay"));
+ };
+
+ mCallback->Output(new AudioData(aSample->mOffset,
+ time.value(),
+ duration.value(),
+ frames,
+ Move(buffer),
+ mOpusParser->mChannels,
+ mOpusParser->mRate));
+ mFrames += frames;
+ return NS_OK;
+}
+
+void
+OpusDataDecoder::ProcessDrain()
+{
+ mCallback->DrainComplete();
+}
+
+void
+OpusDataDecoder::Drain()
+{
+ mTaskQueue->Dispatch(NewRunnableMethod(this, &OpusDataDecoder::ProcessDrain));
+}
+
+void
+OpusDataDecoder::Flush()
+{
+ if (!mOpusDecoder) {
+ return;
+ }
+ mIsFlushing = true;
+ RefPtr<OpusDataDecoder> self = this;
+ nsCOMPtr<nsIRunnable> runnable = NS_NewRunnableFunction([self] () {
+ MOZ_ASSERT(self->mOpusDecoder);
+ // Reset the decoder.
+ opus_multistream_decoder_ctl(self->mOpusDecoder, OPUS_RESET_STATE);
+ self->mSkip = self->mOpusParser->mPreSkip;
+ self->mPaddingDiscarded = false;
+ self->mLastFrameTime.reset();
+ });
+ SyncRunnable::DispatchToThread(mTaskQueue, runnable);
+ mIsFlushing = false;
+}
+
+/* static */
+bool
+OpusDataDecoder::IsOpus(const nsACString& aMimeType)
+{
+ return aMimeType.EqualsLiteral("audio/opus");
+}
+
+} // namespace mozilla
+#undef OPUS_DEBUG