From 6bf0a5cb5034a7e684dcc3500e841785237ce2dd Mon Sep 17 00:00:00 2001 From: Daniel Baumann Date: Sun, 7 Apr 2024 19:32:43 +0200 Subject: Adding upstream version 1:115.7.0. Signed-off-by: Daniel Baumann --- dom/media/mp3/MP3Demuxer.cpp | 890 +++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 890 insertions(+) create mode 100644 dom/media/mp3/MP3Demuxer.cpp (limited to 'dom/media/mp3/MP3Demuxer.cpp') diff --git a/dom/media/mp3/MP3Demuxer.cpp b/dom/media/mp3/MP3Demuxer.cpp new file mode 100644 index 0000000000..25d878b3be --- /dev/null +++ b/dom/media/mp3/MP3Demuxer.cpp @@ -0,0 +1,890 @@ +/* -*- 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 "MP3Demuxer.h" + +#include +#include +#include + +#include "ByteWriter.h" +#include "TimeUnits.h" +#include "VideoUtils.h" +#include "mozilla/Assertions.h" + +extern mozilla::LazyLogModule gMediaDemuxerLog; +#define MP3LOG(msg, ...) \ + DDMOZ_LOG(gMediaDemuxerLog, LogLevel::Debug, msg, ##__VA_ARGS__) +#define MP3LOGV(msg, ...) \ + DDMOZ_LOG(gMediaDemuxerLog, LogLevel::Verbose, msg, ##__VA_ARGS__) + +using mozilla::media::TimeInterval; +using mozilla::media::TimeIntervals; +using mozilla::media::TimeUnit; + +namespace mozilla { + +// MP3Demuxer + +MP3Demuxer::MP3Demuxer(MediaResource* aSource) : mSource(aSource) { + DDLINKCHILD("source", aSource); +} + +bool MP3Demuxer::InitInternal() { + if (!mTrackDemuxer) { + mTrackDemuxer = new MP3TrackDemuxer(mSource); + DDLINKCHILD("track demuxer", mTrackDemuxer.get()); + } + return mTrackDemuxer->Init(); +} + +RefPtr MP3Demuxer::Init() { + if (!InitInternal()) { + MP3LOG("MP3Demuxer::Init() failure: waiting for data"); + + return InitPromise::CreateAndReject(NS_ERROR_DOM_MEDIA_METADATA_ERR, + __func__); + } + + MP3LOG("MP3Demuxer::Init() successful"); + return InitPromise::CreateAndResolve(NS_OK, __func__); +} + +uint32_t MP3Demuxer::GetNumberTracks(TrackInfo::TrackType aType) const { + return aType == TrackInfo::kAudioTrack ? 1u : 0u; +} + +already_AddRefed MP3Demuxer::GetTrackDemuxer( + TrackInfo::TrackType aType, uint32_t aTrackNumber) { + if (!mTrackDemuxer) { + return nullptr; + } + return RefPtr(mTrackDemuxer).forget(); +} + +bool MP3Demuxer::IsSeekable() const { return true; } + +void MP3Demuxer::NotifyDataArrived() { + // TODO: bug 1169485. + NS_WARNING("Unimplemented function NotifyDataArrived"); + MP3LOGV("NotifyDataArrived()"); +} + +void MP3Demuxer::NotifyDataRemoved() { + // TODO: bug 1169485. + NS_WARNING("Unimplemented function NotifyDataRemoved"); + MP3LOGV("NotifyDataRemoved()"); +} + +// MP3TrackDemuxer + +MP3TrackDemuxer::MP3TrackDemuxer(MediaResource* aSource) + : mSource(aSource), + mFrameLock(false), + mOffset(0), + mFirstFrameOffset(0), + mNumParsedFrames(0), + mFrameIndex(0), + mTotalFrameLen(0), + mSamplesPerFrame(0), + mSamplesPerSecond(0), + mChannels(0) { + DDLINKCHILD("source", aSource); + Reset(); +} + +bool MP3TrackDemuxer::Init() { + Reset(); + FastSeek(TimeUnit()); + // Read the first frame to fetch sample rate and other meta data. + RefPtr frame(GetNextFrame(FindFirstFrame())); + + MP3LOG("Init StreamLength()=%" PRId64 " first-frame-found=%d", StreamLength(), + !!frame); + + if (!frame) { + return false; + } + + // Rewind back to the stream begin to avoid dropping the first frame. + FastSeek(TimeUnit()); + + if (!mInfo) { + mInfo = MakeUnique(); + } + + mInfo->mRate = mSamplesPerSecond; + mInfo->mChannels = mChannels; + mInfo->mBitDepth = 16; + mInfo->mMimeType = "audio/mpeg"; + mInfo->mDuration = Duration().valueOr(TimeUnit::FromInfinity()); + Mp3CodecSpecificData mp3CodecData{}; + if (mEncoderDelay) { + mp3CodecData.mEncoderDelayFrames = mEncoderDelay; + mp3CodecData.mEncoderPaddingFrames = mEncoderPadding; + } + mInfo->mCodecSpecificConfig = + AudioCodecSpecificVariant{std::move(mp3CodecData)}; + + MP3LOG("Init mInfo={mRate=%d mChannels=%d mBitDepth=%d mDuration=%s (%lfs)}", + mInfo->mRate, mInfo->mChannels, mInfo->mBitDepth, + mInfo->mDuration.ToString().get(), mInfo->mDuration.ToSeconds()); + + return mSamplesPerSecond && mChannels; +} + +media::TimeUnit MP3TrackDemuxer::SeekPosition() const { + TimeUnit pos = Duration(mFrameIndex); + auto duration = Duration(); + if (duration) { + pos = std::min(*duration, pos); + } + return pos; +} + +const FrameParser::Frame& MP3TrackDemuxer::LastFrame() const { + return mParser.PrevFrame(); +} + +RefPtr MP3TrackDemuxer::DemuxSample() { + return GetNextFrame(FindNextFrame()); +} + +const ID3Parser::ID3Header& MP3TrackDemuxer::ID3Header() const { + return mParser.ID3Header(); +} + +const FrameParser::VBRHeader& MP3TrackDemuxer::VBRInfo() const { + return mParser.VBRInfo(); +} + +UniquePtr MP3TrackDemuxer::GetInfo() const { return mInfo->Clone(); } + +RefPtr MP3TrackDemuxer::Seek( + const TimeUnit& aTime) { + mRemainingEncoderPadding = AssertedCast(mEncoderPadding); + // Efficiently seek to the position. + FastSeek(aTime); + // Correct seek position by scanning the next frames. + const TimeUnit seekTime = ScanUntil(aTime); + + return SeekPromise::CreateAndResolve(seekTime, __func__); +} + +TimeUnit MP3TrackDemuxer::FastSeek(const TimeUnit& aTime) { + MP3LOG("FastSeek(%" PRId64 ") avgFrameLen=%f mNumParsedFrames=%" PRIu64 + " mFrameIndex=%" PRId64 " mOffset=%" PRIu64, + aTime.ToMicroseconds(), AverageFrameLength(), mNumParsedFrames, + mFrameIndex, mOffset); + + const auto& vbr = mParser.VBRInfo(); + if (aTime.IsZero()) { + // Quick seek to the beginning of the stream. + mFrameIndex = 0; + } else if (vbr.IsTOCPresent() && Duration() && + *Duration() != TimeUnit::Zero()) { + // Use TOC for more precise seeking. + mFrameIndex = FrameIndexFromOffset(vbr.Offset(aTime, Duration().value())); + } else if (AverageFrameLength() > 0) { + mFrameIndex = FrameIndexFromTime(aTime); + } + + mOffset = OffsetFromFrameIndex(mFrameIndex); + + if (mOffset > mFirstFrameOffset && StreamLength() > 0) { + mOffset = std::min(StreamLength() - 1, mOffset); + } + + mParser.EndFrameSession(); + + MP3LOG("FastSeek End TOC=%d avgFrameLen=%f mNumParsedFrames=%" PRIu64 + " mFrameIndex=%" PRId64 " mFirstFrameOffset=%" PRId64 + " mOffset=%" PRIu64 " SL=%" PRId64 " NumBytes=%u", + vbr.IsTOCPresent(), AverageFrameLength(), mNumParsedFrames, + mFrameIndex, mFirstFrameOffset, mOffset, StreamLength(), + vbr.NumBytes().valueOr(0)); + + return Duration(mFrameIndex); +} + +TimeUnit MP3TrackDemuxer::ScanUntil(const TimeUnit& aTime) { + MP3LOG("ScanUntil(%" PRId64 ") avgFrameLen=%f mNumParsedFrames=%" PRIu64 + " mFrameIndex=%" PRId64 " mOffset=%" PRIu64, + aTime.ToMicroseconds(), AverageFrameLength(), mNumParsedFrames, + mFrameIndex, mOffset); + + if (aTime.IsZero()) { + return FastSeek(aTime); + } + + if (Duration(mFrameIndex) > aTime) { + // We've seeked past the target time, rewind back a little to correct it. + const int64_t rewind = aTime.ToMicroseconds() / 100; + FastSeek(aTime - TimeUnit::FromMicroseconds(rewind)); + } + + if (Duration(mFrameIndex + 1) > aTime) { + return SeekPosition(); + } + + MediaByteRange nextRange = FindNextFrame(); + while (SkipNextFrame(nextRange) && Duration(mFrameIndex + 1) < aTime) { + nextRange = FindNextFrame(); + MP3LOGV("ScanUntil* avgFrameLen=%f mNumParsedFrames=%" PRIu64 + " mFrameIndex=%" PRId64 " mOffset=%" PRIu64 " Duration=%" PRId64, + AverageFrameLength(), mNumParsedFrames, mFrameIndex, mOffset, + Duration(mFrameIndex + 1).ToMicroseconds()); + } + + MP3LOG("ScanUntil End avgFrameLen=%f mNumParsedFrames=%" PRIu64 + " mFrameIndex=%" PRId64 " mOffset=%" PRIu64, + AverageFrameLength(), mNumParsedFrames, mFrameIndex, mOffset); + + return SeekPosition(); +} + +RefPtr MP3TrackDemuxer::GetSamples( + int32_t aNumSamples) { + MP3LOGV("GetSamples(%d) Begin mOffset=%" PRIu64 " mNumParsedFrames=%" PRIu64 + " mFrameIndex=%" PRId64 " mTotalFrameLen=%" PRIu64 + " mSamplesPerFrame=%d mSamplesPerSecond=%d mChannels=%d", + aNumSamples, mOffset, mNumParsedFrames, mFrameIndex, mTotalFrameLen, + mSamplesPerFrame, mSamplesPerSecond, mChannels); + + if (!aNumSamples) { + return SamplesPromise::CreateAndReject(NS_ERROR_DOM_MEDIA_DEMUXER_ERR, + __func__); + } + + RefPtr frames = new SamplesHolder(); + + while (aNumSamples--) { + RefPtr frame(GetNextFrame(FindNextFrame())); + if (!frame) { + break; + } + if (!frame->HasValidTime()) { + return SamplesPromise::CreateAndReject(NS_ERROR_DOM_MEDIA_DEMUXER_ERR, + __func__); + } + frames->AppendSample(frame); + } + + MP3LOGV("GetSamples() End mSamples.Size()=%zu aNumSamples=%d mOffset=%" PRIu64 + " mNumParsedFrames=%" PRIu64 " mFrameIndex=%" PRId64 + " mTotalFrameLen=%" PRIu64 + " mSamplesPerFrame=%d mSamplesPerSecond=%d " + "mChannels=%d", + frames->GetSamples().Length(), aNumSamples, mOffset, mNumParsedFrames, + mFrameIndex, mTotalFrameLen, mSamplesPerFrame, mSamplesPerSecond, + mChannels); + + if (frames->GetSamples().IsEmpty()) { + return SamplesPromise::CreateAndReject(NS_ERROR_DOM_MEDIA_END_OF_STREAM, + __func__); + } + return SamplesPromise::CreateAndResolve(frames, __func__); +} + +void MP3TrackDemuxer::Reset() { + MP3LOG("Reset()"); + + FastSeek(TimeUnit()); + mParser.Reset(); +} + +RefPtr +MP3TrackDemuxer::SkipToNextRandomAccessPoint(const TimeUnit& aTimeThreshold) { + // Will not be called for audio-only resources. + return SkipAccessPointPromise::CreateAndReject( + SkipFailureHolder(NS_ERROR_DOM_MEDIA_DEMUXER_ERR, 0), __func__); +} + +int64_t MP3TrackDemuxer::GetResourceOffset() const { return mOffset; } + +TimeIntervals MP3TrackDemuxer::GetBuffered() { + AutoPinned stream(mSource.GetResource()); + TimeIntervals buffered; + + if (Duration() && stream->IsDataCachedToEndOfResource(0)) { + // Special case completely cached files. This also handles local files. + buffered += TimeInterval(TimeUnit(), *Duration()); + MP3LOGV("buffered = [[%" PRId64 ", %" PRId64 "]]", + TimeUnit().ToMicroseconds(), Duration()->ToMicroseconds()); + return buffered; + } + + MediaByteRangeSet ranges; + nsresult rv = stream->GetCachedRanges(ranges); + NS_ENSURE_SUCCESS(rv, buffered); + + for (const auto& range : ranges) { + if (range.IsEmpty()) { + continue; + } + TimeUnit start = Duration(FrameIndexFromOffset(range.mStart)); + TimeUnit end = Duration(FrameIndexFromOffset(range.mEnd)); + MP3LOGV("buffered += [%" PRId64 ", %" PRId64 "]", start.ToMicroseconds(), + end.ToMicroseconds()); + buffered += TimeInterval(start, end); + } + + // If the number of frames reported by the header is valid, + // the duration calculated from it is the maximal duration. + if (ValidNumAudioFrames() && Duration()) { + TimeInterval duration = TimeInterval(TimeUnit(), *Duration()); + return buffered.Intersection(duration); + } + + return buffered; +} + +int64_t MP3TrackDemuxer::StreamLength() const { return mSource.GetLength(); } + +media::NullableTimeUnit NothingIfNegative(TimeUnit aDuration) { + if (aDuration.IsNegative()) { + return Nothing(); + } + return Some(aDuration); +} + +media::NullableTimeUnit MP3TrackDemuxer::Duration() const { + if (!mNumParsedFrames) { + return Nothing(); + } + + int64_t numFrames = 0; + const auto numAudioFrames = ValidNumAudioFrames(); + if (numAudioFrames) { + // VBR headers don't include the VBR header frame. + numFrames = numAudioFrames.value() + 1; + return NothingIfNegative(Duration(numFrames) - + (EncoderDelay() + Padding())); + } + + const int64_t streamLen = StreamLength(); + if (streamLen < 0) { // Live streams. + // Unknown length, we can't estimate duration. + return Nothing(); + } + // We can't early return when streamLen < 0 before checking numAudioFrames + // since some live radio will give an opening remark before playing music + // and the duration of the opening talk can be calculated by numAudioFrames. + + int64_t size = streamLen - mFirstFrameOffset; + MOZ_ASSERT(size); + + if (mParser.ID3v1MetadataFound() && size > 128) { + size -= 128; + } + + // If it's CBR, calculate the duration by bitrate. + if (!mParser.VBRInfo().IsValid()) { + const uint32_t bitrate = mParser.CurrentFrame().Header().Bitrate(); + return NothingIfNegative( + media::TimeUnit::FromSeconds(static_cast(size) * 8 / bitrate)); + } + + if (AverageFrameLength() > 0) { + numFrames = std::lround(AssertedCast(size) / AverageFrameLength()); + } + + return NothingIfNegative(Duration(numFrames) - (EncoderDelay() + Padding())); +} + +TimeUnit MP3TrackDemuxer::Duration(int64_t aNumFrames) const { + if (!mSamplesPerSecond) { + return TimeUnit::Invalid(); + } + + const int64_t frameCount = aNumFrames * mSamplesPerFrame; + return TimeUnit(frameCount, mSamplesPerSecond); +} + +MediaByteRange MP3TrackDemuxer::FindFirstFrame() { + // We attempt to find multiple successive frames to avoid locking onto a false + // positive if we're fed a stream that has been cut mid-frame. + // For compatibility reasons we have to use the same frame count as Chrome, + // since some web sites actually use a file that short to test our playback + // capabilities. + static const int MIN_SUCCESSIVE_FRAMES = 3; + mFrameLock = false; + + MediaByteRange candidateFrame = FindNextFrame(); + int numSuccFrames = candidateFrame.Length() > 0; + MediaByteRange currentFrame = candidateFrame; + MP3LOGV("FindFirst() first candidate frame: mOffset=%" PRIu64 + " Length()=%" PRIu64, + candidateFrame.mStart, candidateFrame.Length()); + + while (candidateFrame.Length()) { + mParser.EndFrameSession(); + mOffset = currentFrame.mEnd; + const MediaByteRange prevFrame = currentFrame; + + // FindNextFrame() here will only return frames consistent with our + // candidate frame. + currentFrame = FindNextFrame(); + numSuccFrames += currentFrame.Length() > 0; + // Multiple successive false positives, which wouldn't be caught by the + // consistency checks alone, can be detected by wrong alignment (non-zero + // gap between frames). + const int64_t frameSeparation = currentFrame.mStart - prevFrame.mEnd; + + if (!currentFrame.Length() || frameSeparation != 0) { + MP3LOGV( + "FindFirst() not enough successive frames detected, " + "rejecting candidate frame: successiveFrames=%d, last " + "Length()=%" PRIu64 ", last frameSeparation=%" PRId64, + numSuccFrames, currentFrame.Length(), frameSeparation); + + mParser.ResetFrameData(); + mOffset = candidateFrame.mStart + 1; + candidateFrame = FindNextFrame(); + numSuccFrames = candidateFrame.Length() > 0; + currentFrame = candidateFrame; + MP3LOGV("FindFirst() new candidate frame: mOffset=%" PRIu64 + " Length()=%" PRIu64, + candidateFrame.mStart, candidateFrame.Length()); + } else if (numSuccFrames >= MIN_SUCCESSIVE_FRAMES) { + MP3LOG( + "FindFirst() accepting candidate frame: " + "successiveFrames=%d", + numSuccFrames); + mFrameLock = true; + return candidateFrame; + } else if (prevFrame.mStart == mParser.TotalID3HeaderSize() && + currentFrame.mEnd == StreamLength()) { + // We accept streams with only two frames if both frames are valid. This + // is to handle very short files and provide parity with Chrome. See + // bug 1432195 for more information. This will not handle short files + // with a trailing tag, but as of writing we lack infrastructure to + // handle such tags. + MP3LOG( + "FindFirst() accepting candidate frame for short stream: " + "successiveFrames=%d", + numSuccFrames); + mFrameLock = true; + return candidateFrame; + } + } + + MP3LOG("FindFirst() no suitable first frame found"); + return candidateFrame; +} + +static bool VerifyFrameConsistency(const FrameParser::Frame& aFrame1, + const FrameParser::Frame& aFrame2) { + const auto& h1 = aFrame1.Header(); + const auto& h2 = aFrame2.Header(); + + return h1.IsValid() && h2.IsValid() && h1.Layer() == h2.Layer() && + h1.SlotSize() == h2.SlotSize() && + h1.SamplesPerFrame() == h2.SamplesPerFrame() && + h1.Channels() == h2.Channels() && h1.SampleRate() == h2.SampleRate() && + h1.RawVersion() == h2.RawVersion() && + h1.RawProtection() == h2.RawProtection(); +} + +MediaByteRange MP3TrackDemuxer::FindNextFrame() { + static const int BUFFER_SIZE = 64; + static const uint32_t MAX_SKIPPABLE_BYTES = 1024 * BUFFER_SIZE; + + MP3LOGV("FindNext() Begin mOffset=%" PRIu64 " mNumParsedFrames=%" PRIu64 + " mFrameIndex=%" PRId64 " mTotalFrameLen=%" PRIu64 + " mSamplesPerFrame=%d mSamplesPerSecond=%d mChannels=%d", + mOffset, mNumParsedFrames, mFrameIndex, mTotalFrameLen, + mSamplesPerFrame, mSamplesPerSecond, mChannels); + + uint8_t buffer[BUFFER_SIZE]; + uint32_t read = 0; + + bool foundFrame = false; + int64_t frameHeaderOffset = 0; + int64_t startOffset = mOffset; + const bool searchingForID3 = !mParser.ID3Header().HasSizeBeenSet(); + + // Check whether we've found a valid MPEG frame. + while (!foundFrame) { + // How many bytes we can go without finding a valid MPEG frame + // (effectively rounded up to the next full buffer size multiple, as we + // only check this before reading the next set of data into the buffer). + + // This default value of 0 will be used during testing whether we're being + // fed a valid stream, which shouldn't have any gaps between frames. + uint32_t maxSkippableBytes = 0; + + if (!mParser.FirstFrame().Length()) { + // We're looking for the first valid frame. A well-formed file should + // have its first frame header right at the start (skipping an ID3 tag + // if necessary), but in order to support files that might have been + // improperly cut, we search the first few kB for a frame header. + maxSkippableBytes = MAX_SKIPPABLE_BYTES; + // Since we're counting the skipped bytes from the offset we started + // this parsing session with, we need to discount the ID3 tag size only + // if we were looking for one during the current frame parsing session. + if (searchingForID3) { + maxSkippableBytes += mParser.TotalID3HeaderSize(); + } + } else if (mFrameLock) { + // We've found a valid MPEG stream, so don't impose any limits + // to allow skipping corrupted data until we hit EOS. + maxSkippableBytes = std::numeric_limits::max(); + } + + if ((mOffset - startOffset > maxSkippableBytes) || + (read = Read(buffer, mOffset, BUFFER_SIZE)) == 0) { + MP3LOG( + "FindNext() EOS or exceeded maxSkippeableBytes without a frame " + "(read: %d)", + read); + // This is not a valid MPEG audio stream or we've reached EOS, give up. + break; + } + + BufferReader reader(buffer, read); + uint32_t bytesToSkip = 0; + auto res = mParser.Parse(&reader, &bytesToSkip); + foundFrame = res.unwrapOr(false); + int64_t readerOffset = static_cast(reader.Offset()); + frameHeaderOffset = mOffset + readerOffset - FrameParser::FrameHeader::SIZE; + + // If we've found neither an MPEG frame header nor an ID3v2 tag, + // the reader shouldn't have any bytes remaining. + MOZ_ASSERT(foundFrame || bytesToSkip || !reader.Remaining()); + + if (foundFrame && mParser.FirstFrame().Length() && + !VerifyFrameConsistency(mParser.FirstFrame(), mParser.CurrentFrame())) { + MP3LOG("Skipping frame"); + // We've likely hit a false-positive, ignore it and proceed with the + // search for the next valid frame. + foundFrame = false; + mOffset = frameHeaderOffset + 1; + mParser.EndFrameSession(); + } else { + // Advance mOffset by the amount of bytes read and if necessary, + // skip an ID3v2 tag which stretches beyond the current buffer. + NS_ENSURE_TRUE(mOffset + read + bytesToSkip > mOffset, + MediaByteRange(0, 0)); + mOffset += static_cast(read + bytesToSkip); + } + } + + if (StreamLength() != -1) { + mEOS = frameHeaderOffset + mParser.CurrentFrame().Length() + BUFFER_SIZE > + StreamLength(); + } + + if (!foundFrame || !mParser.CurrentFrame().Length()) { + MP3LOG("FindNext() Exit foundFrame=%d mParser.CurrentFrame().Length()=%d ", + foundFrame, mParser.CurrentFrame().Length()); + return {0, 0}; + } + + MP3LOGV("FindNext() End mOffset=%" PRIu64 " mNumParsedFrames=%" PRIu64 + " mFrameIndex=%" PRId64 " frameHeaderOffset=%" PRId64 + " mTotalFrameLen=%" PRIu64 + " mSamplesPerFrame=%d mSamplesPerSecond=%d" + " mChannels=%d, mEOS=%s", + mOffset, mNumParsedFrames, mFrameIndex, frameHeaderOffset, + mTotalFrameLen, mSamplesPerFrame, mSamplesPerSecond, mChannels, + mEOS ? "true" : "false"); + + return {frameHeaderOffset, + frameHeaderOffset + mParser.CurrentFrame().Length()}; +} + +bool MP3TrackDemuxer::SkipNextFrame(const MediaByteRange& aRange) { + if (!mNumParsedFrames || !aRange.Length()) { + // We can't skip the first frame, since it could contain VBR headers. + RefPtr frame(GetNextFrame(aRange)); + return frame; + } + + UpdateState(aRange); + + MP3LOGV("SkipNext() End mOffset=%" PRIu64 " mNumParsedFrames=%" PRIu64 + " mFrameIndex=%" PRId64 " mTotalFrameLen=%" PRIu64 + " mSamplesPerFrame=%d mSamplesPerSecond=%d mChannels=%d", + mOffset, mNumParsedFrames, mFrameIndex, mTotalFrameLen, + mSamplesPerFrame, mSamplesPerSecond, mChannels); + + return true; +} + +media::TimeUnit MP3TrackDemuxer::EncoderDelay() const { + return media::TimeUnit(mEncoderDelay, mSamplesPerSecond); +} + +uint32_t MP3TrackDemuxer::EncoderDelayFrames() const { return mEncoderDelay; } + +media::TimeUnit MP3TrackDemuxer::Padding() const { + return media::TimeUnit(mEncoderPadding, mSamplesPerSecond); +} + +uint32_t MP3TrackDemuxer::PaddingFrames() const { return mEncoderPadding; } + +already_AddRefed MP3TrackDemuxer::GetNextFrame( + const MediaByteRange& aRange) { + MP3LOG("GetNext() Begin({mStart=%" PRId64 " Length()=%" PRId64 "})", + aRange.mStart, aRange.Length()); + if (!aRange.Length()) { + return nullptr; + } + + RefPtr frame = new MediaRawData(); + frame->mOffset = aRange.mStart; + + UniquePtr frameWriter(frame->CreateWriter()); + if (!frameWriter->SetSize(static_cast(aRange.Length()))) { + MP3LOG("GetNext() Exit failed to allocated media buffer"); + return nullptr; + } + + const uint32_t read = + Read(frameWriter->Data(), frame->mOffset, frame->Size()); + + if (read != aRange.Length()) { + MP3LOG("GetNext() Exit read=%u frame->Size()=%zu", read, frame->Size()); + return nullptr; + } + + UpdateState(aRange); + + if (mNumParsedFrames == 1) { + // First frame parsed, let's read VBR info if available. + BufferReader reader(frame->Data(), frame->Size()); + mFirstFrameOffset = frame->mOffset; + + if (mParser.ParseVBRHeader(&reader)) { + // Parsing was successful + if (mParser.VBRInfo().Type() == FrameParser::VBRHeader::XING) { + MP3LOG("XING header present, skipping encoder delay (%u frames)", + mParser.VBRInfo().EncoderDelay()); + mEncoderDelay = mParser.VBRInfo().EncoderDelay(); + mEncoderPadding = mParser.VBRInfo().EncoderPadding(); + // Padding is encoded as a 12-bit unsigned number so this is fine. + mRemainingEncoderPadding = AssertedCast(mEncoderPadding); + if (mEncoderDelay == 0) { + // Skip the VBR frame + the decoder delay, that is always 529 frames + // in practice for the decoder we're using. + mEncoderDelay = mSamplesPerFrame + 529; + MP3LOG( + "No explicit delay present in vbr header, delay is assumed to be " + "%u frames\n", + mEncoderDelay); + } + } else if (mParser.VBRInfo().Type() == FrameParser::VBRHeader::VBRI) { + MP3LOG("VBRI header present, skipping encoder delay (%u frames)", + mParser.VBRInfo().EncoderDelay()); + mEncoderDelay = mParser.VBRInfo().EncoderDelay(); + } + } + } + + TimeUnit rawPts = Duration(mFrameIndex - 1) - EncoderDelay(); + TimeUnit rawDuration = Duration(1); + TimeUnit rawEnd = rawPts + rawDuration; + + frame->mTime = std::max(TimeUnit::Zero(mSamplesPerSecond), rawPts); + + frame->mDuration = Duration(1); + frame->mTimecode = frame->mTime; + frame->mKeyframe = true; + frame->mEOS = mEOS; + + // Handle decoder delay. A packet must be trimmed if its pts, adjusted for + // decoder delay, is negative. A packet can be trimmed entirely. + if (rawPts.IsNegative()) { + frame->mDuration = + std::max(TimeUnit::Zero(mSamplesPerSecond), rawEnd - frame->mTime); + } + + // It's possible to create an mp3 file that has a padding value that somehow + // spans multiple packets. In that case the duration is probably known, + // because it's probably a VBR file with a XING header (that has a duration + // field). Use the duration to be able to set the correct duration on + // packets that aren't the last one. + // For most files, the padding is less than a packet, it's simply substracted. + if (mParser.VBRInfo().Type() == FrameParser::VBRHeader::XING && + mRemainingEncoderPadding > 0 && + frame->GetEndTime() > Duration().valueOr(TimeUnit::FromInfinity())) { + TimeUnit duration = Duration().value(); + TimeUnit inPaddingZone = frame->GetEndTime() - duration; + TimeUnit originalEnd = frame->GetEndTime(); + TimeUnit originalPts = frame->mTime; + frame->mDuration -= inPaddingZone; + // Packet is entirely padding and will be completely discarded by the + // decoder. + if (frame->mDuration.IsNegative()) { + frame->mDuration = TimeUnit::Zero(mSamplesPerSecond); + } + int32_t paddingFrames = + AssertedCast(inPaddingZone.ToTicksAtRate(mSamplesPerSecond)); + if (mRemainingEncoderPadding >= paddingFrames) { + mRemainingEncoderPadding -= paddingFrames; + } else { + mRemainingEncoderPadding = 0; + } + MP3LOG("Trimming [%s, %s] to [%s,%s] (padding) (stream duration: %s)", + originalPts.ToString().get(), originalEnd.ToString().get(), + frame->mTime.ToString().get(), frame->GetEndTime().ToString().get(), + duration.ToString().get()); + } else if (frame->mEOS && + mRemainingEncoderPadding <= + frame->mDuration.ToTicksAtRate(mSamplesPerSecond)) { + frame->mDuration -= TimeUnit(mRemainingEncoderPadding, mSamplesPerSecond); + MOZ_ASSERT(frame->mDuration.IsPositiveOrZero()); + MP3LOG("Trimming last packet %s to [%s,%s]", Padding().ToString().get(), + frame->mTime.ToString().get(), frame->GetEndTime().ToString().get()); + } + + MP3LOGV("GetNext() End mOffset=%" PRIu64 " mNumParsedFrames=%" PRIu64 + " mFrameIndex=%" PRId64 " mTotalFrameLen=%" PRIu64 + " mSamplesPerFrame=%d mSamplesPerSecond=%d mChannels=%d, mEOS=%s", + mOffset, mNumParsedFrames, mFrameIndex, mTotalFrameLen, + mSamplesPerFrame, mSamplesPerSecond, mChannels, + mEOS ? "true" : "false"); + + // It's possible for the duration of a frame to be zero if the frame is to be + // trimmed entirely because it's fully comprised of decoder delay samples. + // This is common at the beginning of an stream. + MOZ_ASSERT(frame->mDuration.IsPositiveOrZero()); + + MP3LOG("Packet demuxed: pts [%s, %s] (duration: %s)", + frame->mTime.ToString().get(), frame->GetEndTime().ToString().get(), + frame->mDuration.ToString().get()); + + // Indicate original packet information to trim after decoding. + if (frame->mDuration != rawDuration) { + frame->mOriginalPresentationWindow = Some(TimeInterval{rawPts, rawEnd}); + MP3LOG("Total packet time excluding trimming: [%s, %s]", + rawPts.ToString().get(), rawEnd.ToString().get()); + } + + return frame.forget(); +} + +int64_t MP3TrackDemuxer::OffsetFromFrameIndex(int64_t aFrameIndex) const { + int64_t offset = 0; + const auto& vbr = mParser.VBRInfo(); + + if (vbr.IsComplete()) { + offset = mFirstFrameOffset + aFrameIndex * vbr.NumBytes().value() / + vbr.NumAudioFrames().value(); + } else if (AverageFrameLength() > 0) { + offset = mFirstFrameOffset + + AssertedCast(static_cast(aFrameIndex) * + AverageFrameLength()); + } + + MP3LOGV("OffsetFromFrameIndex(%" PRId64 ") -> %" PRId64, aFrameIndex, offset); + return std::max(mFirstFrameOffset, offset); +} + +int64_t MP3TrackDemuxer::FrameIndexFromOffset(int64_t aOffset) const { + int64_t frameIndex = 0; + const auto& vbr = mParser.VBRInfo(); + + if (vbr.IsComplete()) { + frameIndex = + AssertedCast(static_cast(aOffset - mFirstFrameOffset) / + static_cast(vbr.NumBytes().value()) * + static_cast(vbr.NumAudioFrames().value())); + frameIndex = std::min(vbr.NumAudioFrames().value(), frameIndex); + } else if (AverageFrameLength() > 0) { + frameIndex = AssertedCast( + static_cast(aOffset - mFirstFrameOffset) / AverageFrameLength()); + } + + MP3LOGV("FrameIndexFromOffset(%" PRId64 ") -> %" PRId64, aOffset, frameIndex); + return std::max(0, frameIndex); +} + +int64_t MP3TrackDemuxer::FrameIndexFromTime( + const media::TimeUnit& aTime) const { + int64_t frameIndex = 0; + if (mSamplesPerSecond > 0 && mSamplesPerFrame > 0) { + frameIndex = AssertedCast( + aTime.ToSeconds() * mSamplesPerSecond / mSamplesPerFrame - 1); + } + + MP3LOGV("FrameIndexFromOffset(%fs) -> %" PRId64, aTime.ToSeconds(), + frameIndex); + return std::max(0, frameIndex); +} + +void MP3TrackDemuxer::UpdateState(const MediaByteRange& aRange) { + // Prevent overflow. + if (mTotalFrameLen + aRange.Length() < mTotalFrameLen) { + // These variables have a linear dependency and are only used to derive the + // average frame length. + mTotalFrameLen /= 2; + mNumParsedFrames /= 2; + } + + // Full frame parsed, move offset to its end. + mOffset = aRange.mEnd; + + mTotalFrameLen += aRange.Length(); + + if (!mSamplesPerFrame) { + mSamplesPerFrame = mParser.CurrentFrame().Header().SamplesPerFrame(); + mSamplesPerSecond = mParser.CurrentFrame().Header().SampleRate(); + mChannels = mParser.CurrentFrame().Header().Channels(); + } + + ++mNumParsedFrames; + ++mFrameIndex; + MOZ_ASSERT(mFrameIndex > 0); + + // Prepare the parser for the next frame parsing session. + mParser.EndFrameSession(); +} + +uint32_t MP3TrackDemuxer::Read(uint8_t* aBuffer, int64_t aOffset, + uint32_t aSize) { + MP3LOGV("MP3TrackDemuxer::Read(%p %" PRId64 " %d)", aBuffer, aOffset, aSize); + + const int64_t streamLen = StreamLength(); + if (mInfo && streamLen > 0) { + // Prevent blocking reads after successful initialization. + int64_t max = streamLen > aOffset ? streamLen - aOffset : 0; + aSize = std::min(aSize, max); + } + + uint32_t read = 0; + MP3LOGV("MP3TrackDemuxer::Read -> ReadAt(%u)", aSize); + const nsresult rv = mSource.ReadAt(aOffset, reinterpret_cast(aBuffer), + static_cast(aSize), &read); + NS_ENSURE_SUCCESS(rv, 0); + return read; +} + +double MP3TrackDemuxer::AverageFrameLength() const { + if (mNumParsedFrames) { + return static_cast(mTotalFrameLen) / + static_cast(mNumParsedFrames); + } + const auto& vbr = mParser.VBRInfo(); + if (vbr.IsComplete() && vbr.NumAudioFrames().value() + 1) { + return static_cast(vbr.NumBytes().value()) / + (vbr.NumAudioFrames().value() + 1); + } + return 0.0; +} + +Maybe MP3TrackDemuxer::ValidNumAudioFrames() const { + return mParser.VBRInfo().IsValid() && + mParser.VBRInfo().NumAudioFrames().valueOr(0) + 1 > 1 + ? mParser.VBRInfo().NumAudioFrames() + : Nothing(); +} + +} // namespace mozilla + +#undef MP3LOG +#undef MP3LOGV -- cgit v1.2.3