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Bug 1069660 - Factor Vorbis and Opus decoding out.
This commit is contained in:
Matthew Gregan
parent
a0648a0cdf
commit
672372df86
@ -531,7 +531,6 @@ bool WebMReader::DecodeAudioPacket(nestegg_packet* aPacket, int64_t aOffset)
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return false;
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}
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const uint32_t rate = mInfo.mAudio.mRate;
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uint64_t tstamp_usecs = tstamp / NS_PER_USEC;
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if (mAudioStartUsec == -1) {
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// This is the first audio chunk. Assume the start time of our decode
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@ -542,8 +541,8 @@ bool WebMReader::DecodeAudioPacket(nestegg_packet* aPacket, int64_t aOffset)
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// the previous audio chunk, we need to increment the packet count so that
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// the vorbis decode doesn't use data from before the gap to help decode
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// from after the gap.
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CheckedInt64 tstamp_frames = UsecsToFrames(tstamp_usecs, rate);
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CheckedInt64 decoded_frames = UsecsToFrames(mAudioStartUsec, rate);
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CheckedInt64 tstamp_frames = UsecsToFrames(tstamp_usecs, mInfo.mAudio.mRate);
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CheckedInt64 decoded_frames = UsecsToFrames(mAudioStartUsec, mInfo.mAudio.mRate);
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if (!tstamp_frames.isValid() || !decoded_frames.isValid()) {
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NS_WARNING("Int overflow converting WebM times to frames");
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return false;
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@ -555,7 +554,7 @@ bool WebMReader::DecodeAudioPacket(nestegg_packet* aPacket, int64_t aOffset)
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}
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if (tstamp_frames.value() > decoded_frames.value()) {
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#ifdef DEBUG
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CheckedInt64 usecs = FramesToUsecs(tstamp_frames.value() - decoded_frames.value(), rate);
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CheckedInt64 usecs = FramesToUsecs(tstamp_frames.value() - decoded_frames.value(), mInfo.mAudio.mRate);
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LOG(PR_LOG_DEBUG, ("WebMReader detected gap of %lld, %lld frames, in audio stream\n",
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usecs.isValid() ? usecs.value() : -1,
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tstamp_frames.value() - decoded_frames.value()));
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@ -574,198 +573,218 @@ bool WebMReader::DecodeAudioPacket(nestegg_packet* aPacket, int64_t aOffset)
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return false;
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}
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if (mAudioCodec == NESTEGG_CODEC_VORBIS) {
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ogg_packet opacket = InitOggPacket(data, length, false, false, -1);
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if (vorbis_synthesis(&mVorbisBlock, &opacket) != 0) {
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if (!DecodeVorbis(data, length, aOffset, tstamp_usecs, &total_frames)) {
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return false;
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}
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if (vorbis_synthesis_blockin(&mVorbisDsp,
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&mVorbisBlock) != 0) {
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return false;
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}
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VorbisPCMValue** pcm = 0;
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int32_t frames = vorbis_synthesis_pcmout(&mVorbisDsp, &pcm);
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// If the first packet of audio in the media produces no data, we
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// still need to produce an AudioData for it so that the correct media
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// start time is calculated. Otherwise we'd end up with a media start
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// time derived from the timecode of the first packet that produced
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// data.
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if (frames == 0 && mAudioFrames == 0) {
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AudioQueue().Push(new AudioData(aOffset, tstamp_usecs, 0, 0, nullptr, mChannels, rate));
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}
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while (frames > 0) {
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nsAutoArrayPtr<AudioDataValue> buffer(new AudioDataValue[frames * mChannels]);
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for (uint32_t j = 0; j < mChannels; ++j) {
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VorbisPCMValue* channel = pcm[j];
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for (uint32_t i = 0; i < uint32_t(frames); ++i) {
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buffer[i*mChannels + j] = MOZ_CONVERT_VORBIS_SAMPLE(channel[i]);
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}
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}
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CheckedInt64 duration = FramesToUsecs(frames, rate);
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if (!duration.isValid()) {
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NS_WARNING("Int overflow converting WebM audio duration");
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return false;
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}
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CheckedInt64 total_duration = FramesToUsecs(total_frames, rate);
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if (!total_duration.isValid()) {
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NS_WARNING("Int overflow converting WebM audio total_duration");
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return false;
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}
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CheckedInt64 time = total_duration + tstamp_usecs;
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if (!time.isValid()) {
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NS_WARNING("Int overflow adding total_duration and tstamp_usecs");
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return false;
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};
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total_frames += frames;
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AudioQueue().Push(new AudioData(aOffset,
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time.value(),
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duration.value(),
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frames,
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buffer.forget(),
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mChannels,
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rate));
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mAudioFrames += frames;
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if (vorbis_synthesis_read(&mVorbisDsp, frames) != 0) {
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return false;
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}
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frames = vorbis_synthesis_pcmout(&mVorbisDsp, &pcm);
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}
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} else if (mAudioCodec == NESTEGG_CODEC_OPUS) {
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#ifdef MOZ_OPUS
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uint32_t channels = mOpusParser->mChannels;
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// Maximum value is 63*2880, so there's no chance of overflow.
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int32_t frames_number = opus_packet_get_nb_frames(data, length);
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if (frames_number <= 0)
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return false; // Invalid packet header.
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int32_t samples = opus_packet_get_samples_per_frame(data,
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(opus_int32) rate);
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int32_t frames = frames_number*samples;
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// A valid Opus packet must be between 2.5 and 120 ms long.
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if (frames < 120 || frames > 5760)
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if (!DecodeOpus(data, length, aOffset, tstamp_usecs, aPacket)) {
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return false;
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nsAutoArrayPtr<AudioDataValue> buffer(new AudioDataValue[frames * channels]);
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// Decode to the appropriate sample type.
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#ifdef MOZ_SAMPLE_TYPE_FLOAT32
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int ret = opus_multistream_decode_float(mOpusDecoder,
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data, length,
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buffer, frames, false);
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#else
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int ret = opus_multistream_decode(mOpusDecoder,
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data, length,
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buffer, frames, false);
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#endif
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if (ret < 0)
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return false;
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NS_ASSERTION(ret == frames, "Opus decoded too few audio samples");
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CheckedInt64 startTime = tstamp_usecs;
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// Trim the initial frames while the decoder is settling.
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if (mSkip > 0) {
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int32_t skipFrames = std::min(mSkip, frames);
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if (skipFrames == frames) {
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// discard the whole packet
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mSkip -= frames;
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LOG(PR_LOG_DEBUG, ("Opus decoder skipping %d frames"
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" (whole packet)", frames));
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return true;
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}
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int32_t keepFrames = frames - skipFrames;
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int samples = keepFrames * channels;
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nsAutoArrayPtr<AudioDataValue> trimBuffer(new AudioDataValue[samples]);
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PodCopy(trimBuffer.get(), buffer.get() + skipFrames*channels, samples);
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startTime = startTime + FramesToUsecs(skipFrames, rate);
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frames = keepFrames;
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buffer = trimBuffer;
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mSkip -= skipFrames;
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LOG(PR_LOG_DEBUG, ("Opus decoder skipping %d frames", skipFrames));
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}
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int64_t discardPadding = 0;
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r = nestegg_packet_discard_padding(aPacket, &discardPadding);
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if (discardPadding > 0) {
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CheckedInt64 discardFrames = UsecsToFrames(discardPadding / NS_PER_USEC, rate);
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if (!discardFrames.isValid()) {
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NS_WARNING("Int overflow in DiscardPadding");
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return false;
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}
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if (discardFrames.value() >= frames) {
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LOG(PR_LOG_DEBUG, ("Opus decoder discarding whole packet"
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" (%d frames) as padding (%lld discarded)",
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frames, discardFrames.value()));
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return true;
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}
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int32_t keepFrames = frames - discardFrames.value();
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int32_t samples = keepFrames * channels;
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nsAutoArrayPtr<AudioDataValue> trimBuffer(new AudioDataValue[samples]);
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PodCopy(trimBuffer.get(), buffer.get(), samples);
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frames = keepFrames;
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buffer = trimBuffer;
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}
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// Apply the header gain if one was specified.
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#ifdef MOZ_SAMPLE_TYPE_FLOAT32
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if (mOpusParser->mGain != 1.0f) {
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float gain = mOpusParser->mGain;
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int samples = frames * channels;
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for (int i = 0; i < samples; i++) {
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buffer[i] *= gain;
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}
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}
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#else
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if (mOpusParser->mGain_Q16 != 65536) {
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int64_t gain_Q16 = mOpusParser->mGain_Q16;
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int samples = frames * channels;
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for (int i = 0; i < samples; i++) {
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int32_t val = static_cast<int32_t>((gain_Q16*buffer[i] + 32768)>>16);
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buffer[i] = static_cast<AudioDataValue>(MOZ_CLIP_TO_15(val));
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}
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}
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#endif
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// No channel mapping for more than 8 channels.
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if (channels > 8) {
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return false;
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}
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CheckedInt64 duration = FramesToUsecs(frames, rate);
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if (!duration.isValid()) {
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NS_WARNING("Int overflow converting WebM audio duration");
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return false;
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}
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CheckedInt64 time = startTime - mCodecDelay;
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if (!time.isValid()) {
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NS_WARNING("Int overflow shifting tstamp by codec delay");
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return false;
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};
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AudioQueue().Push(new AudioData(mDecoder->GetResource()->Tell(),
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time.value(),
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duration.value(),
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frames,
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buffer.forget(),
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mChannels,
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rate));
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mAudioFrames += frames;
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#else
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return false;
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#endif /* MOZ_OPUS */
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}
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}
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return true;
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}
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bool WebMReader::DecodeVorbis(unsigned char* aData, size_t aLength,
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int64_t aOffset, uint64_t aTstampUsecs, int32_t* aTotalFrames)
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{
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ogg_packet opacket = InitOggPacket(aData, aLength, false, false, -1);
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if (vorbis_synthesis(&mVorbisBlock, &opacket) != 0) {
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return false;
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}
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if (vorbis_synthesis_blockin(&mVorbisDsp,
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&mVorbisBlock) != 0) {
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return false;
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}
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VorbisPCMValue** pcm = 0;
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int32_t frames = vorbis_synthesis_pcmout(&mVorbisDsp, &pcm);
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// If the first packet of audio in the media produces no data, we
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// still need to produce an AudioData for it so that the correct media
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// start time is calculated. Otherwise we'd end up with a media start
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// time derived from the timecode of the first packet that produced
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// data.
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if (frames == 0 && mAudioFrames == 0) {
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AudioQueue().Push(new AudioData(aOffset, aTstampUsecs, 0, 0, nullptr, mChannels, mInfo.mAudio.mRate));
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}
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while (frames > 0) {
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nsAutoArrayPtr<AudioDataValue> buffer(new AudioDataValue[frames * mChannels]);
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for (uint32_t j = 0; j < mChannels; ++j) {
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VorbisPCMValue* channel = pcm[j];
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for (uint32_t i = 0; i < uint32_t(frames); ++i) {
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buffer[i*mChannels + j] = MOZ_CONVERT_VORBIS_SAMPLE(channel[i]);
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}
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}
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CheckedInt64 duration = FramesToUsecs(frames, mInfo.mAudio.mRate);
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if (!duration.isValid()) {
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NS_WARNING("Int overflow converting WebM audio duration");
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return false;
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}
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CheckedInt64 total_duration = FramesToUsecs(*aTotalFrames, mInfo.mAudio.mRate);
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if (!total_duration.isValid()) {
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NS_WARNING("Int overflow converting WebM audio total_duration");
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return false;
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}
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CheckedInt64 time = total_duration + aTstampUsecs;
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if (!time.isValid()) {
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NS_WARNING("Int overflow adding total_duration and aTstampUsecs");
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return false;
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};
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*aTotalFrames += frames;
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AudioQueue().Push(new AudioData(aOffset,
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time.value(),
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duration.value(),
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frames,
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buffer.forget(),
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mChannels,
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mInfo.mAudio.mRate));
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mAudioFrames += frames;
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if (vorbis_synthesis_read(&mVorbisDsp, frames) != 0) {
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return false;
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}
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frames = vorbis_synthesis_pcmout(&mVorbisDsp, &pcm);
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}
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return true;
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}
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#ifdef MOZ_OPUS
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bool WebMReader::DecodeOpus(unsigned char* aData, size_t aLength,
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int64_t aOffset, uint64_t aTstampUsecs, nestegg_packet* aPacket)
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{
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uint32_t channels = mOpusParser->mChannels;
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// Maximum value is 63*2880, so there's no chance of overflow.
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int32_t frames_number = opus_packet_get_nb_frames(aData, aLength);
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if (frames_number <= 0)
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return false; // Invalid packet header.
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int32_t samples = opus_packet_get_samples_per_frame(aData,
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(opus_int32) mInfo.mAudio.mRate);
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int32_t frames = frames_number*samples;
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// A valid Opus packet must be between 2.5 and 120 ms long.
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if (frames < 120 || frames > 5760)
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return false;
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nsAutoArrayPtr<AudioDataValue> buffer(new AudioDataValue[frames * channels]);
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// Decode to the appropriate sample type.
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#ifdef MOZ_SAMPLE_TYPE_FLOAT32
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int ret = opus_multistream_decode_float(mOpusDecoder,
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aData, aLength,
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buffer, frames, false);
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#else
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int ret = opus_multistream_decode(mOpusDecoder,
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aData, aLength,
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buffer, frames, false);
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#endif
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if (ret < 0)
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return false;
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NS_ASSERTION(ret == frames, "Opus decoded too few audio samples");
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CheckedInt64 startTime = aTstampUsecs;
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// Trim the initial frames while the decoder is settling.
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if (mSkip > 0) {
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int32_t skipFrames = std::min(mSkip, frames);
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if (skipFrames == frames) {
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// discard the whole packet
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mSkip -= frames;
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LOG(PR_LOG_DEBUG, ("Opus decoder skipping %d frames"
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" (whole packet)", frames));
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return true;
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}
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int32_t keepFrames = frames - skipFrames;
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int samples = keepFrames * channels;
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nsAutoArrayPtr<AudioDataValue> trimBuffer(new AudioDataValue[samples]);
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PodCopy(trimBuffer.get(), buffer.get() + skipFrames*channels, samples);
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startTime = startTime + FramesToUsecs(skipFrames, mInfo.mAudio.mRate);
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frames = keepFrames;
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buffer = trimBuffer;
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mSkip -= skipFrames;
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LOG(PR_LOG_DEBUG, ("Opus decoder skipping %d frames", skipFrames));
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}
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int64_t discardPadding = 0;
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(void) nestegg_packet_discard_padding(aPacket, &discardPadding);
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if (discardPadding > 0) {
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CheckedInt64 discardFrames = UsecsToFrames(discardPadding / NS_PER_USEC, mInfo.mAudio.mRate);
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if (!discardFrames.isValid()) {
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NS_WARNING("Int overflow in DiscardPadding");
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return false;
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}
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if (discardFrames.value() >= frames) {
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LOG(PR_LOG_DEBUG, ("Opus decoder discarding whole packet"
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" (%d frames) as padding (%lld discarded)",
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frames, discardFrames.value()));
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return true;
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}
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int32_t keepFrames = frames - discardFrames.value();
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int32_t samples = keepFrames * channels;
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nsAutoArrayPtr<AudioDataValue> trimBuffer(new AudioDataValue[samples]);
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PodCopy(trimBuffer.get(), buffer.get(), samples);
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frames = keepFrames;
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buffer = trimBuffer;
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}
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// Apply the header gain if one was specified.
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#ifdef MOZ_SAMPLE_TYPE_FLOAT32
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if (mOpusParser->mGain != 1.0f) {
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float gain = mOpusParser->mGain;
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int samples = frames * channels;
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for (int i = 0; i < samples; i++) {
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buffer[i] *= gain;
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}
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}
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#else
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if (mOpusParser->mGain_Q16 != 65536) {
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int64_t gain_Q16 = mOpusParser->mGain_Q16;
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int samples = frames * channels;
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for (int i = 0; i < samples; i++) {
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int32_t val = static_cast<int32_t>((gain_Q16*buffer[i] + 32768)>>16);
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buffer[i] = static_cast<AudioDataValue>(MOZ_CLIP_TO_15(val));
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}
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}
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#endif
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// No channel mapping for more than 8 channels.
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if (channels > 8) {
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return false;
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}
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CheckedInt64 duration = FramesToUsecs(frames, mInfo.mAudio.mRate);
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if (!duration.isValid()) {
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NS_WARNING("Int overflow converting WebM audio duration");
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return false;
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}
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CheckedInt64 time = startTime - mCodecDelay;
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if (!time.isValid()) {
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NS_WARNING("Int overflow shifting tstamp by codec delay");
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return false;
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};
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AudioQueue().Push(new AudioData(aOffset,
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time.value(),
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duration.value(),
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frames,
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buffer.forget(),
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mChannels,
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mInfo.mAudio.mRate));
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mAudioFrames += frames;
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return true;
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}
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#endif /* MOZ_OPUS */
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nsReturnRef<NesteggPacketHolder> WebMReader::NextPacket(TrackType aTrackType)
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{
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// The packet queue that packets will be pushed on if they
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@ -176,6 +176,12 @@ protected:
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// must be held during this call. The caller is responsible for freeing
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// aPacket.
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bool DecodeAudioPacket(nestegg_packet* aPacket, int64_t aOffset);
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bool DecodeVorbis(unsigned char* aData, size_t aLength,
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int64_t aOffset, uint64_t aTstampUsecs, int32_t* aTotalFrames);
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#ifdef MOZ_OPUS
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bool DecodeOpus(unsigned char* aData, size_t aLength,
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int64_t aOffset, uint64_t aTstampUsecs, nestegg_packet* aPacket);
|
||||
#endif
|
||||
|
||||
// Release context and set to null. Called when an error occurs during
|
||||
// reading metadata or destruction of the reader itself.
|
||||
|
||||
Loading…
Reference in New Issue
Block a user