mirror of
https://github.com/archlinuxarm/PKGBUILDs.git
synced 2024-12-08 23:03:46 +00:00
497 lines
24 KiB
Diff
497 lines
24 KiB
Diff
From 6e554a30893150793c2638e3689cf208ffc8e375 Mon Sep 17 00:00:00 2001
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From: Dale Curtis <dalecurtis@chromium.org>
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Date: Sat, 2 Apr 2022 05:13:53 +0000
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Subject: [PATCH] Roll src/third_party/ffmpeg/ 574c39cce..32b2d1d526 (1125
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commits)
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https://chromium.googlesource.com/chromium/third_party/ffmpeg.git/+log/574c39cce323..32b2d1d526
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Created with:
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roll-dep src/third_party/ffmpeg
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Fixed: 1293918
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Cq-Include-Trybots: luci.chromium.try:mac_chromium_asan_rel_ng,linux_chromium_asan_rel_ng,linux_chromium_chromeos_asan_rel_ng
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Change-Id: I41945d0f963e3d1f65940067bac22f63b68e37d2
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Reviewed-on: https://chromium-review.googlesource.com/c/chromium/src/+/3565647
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Auto-Submit: Dale Curtis <dalecurtis@chromium.org>
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Reviewed-by: Dan Sanders <sandersd@chromium.org>
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Commit-Queue: Dale Curtis <dalecurtis@chromium.org>
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Cr-Commit-Position: refs/heads/main@{#988253}
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---
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.../clear_key_cdm/ffmpeg_cdm_audio_decoder.cc | 29 ++++++++++---------
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media/ffmpeg/ffmpeg_common.cc | 11 +++----
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media/filters/audio_file_reader.cc | 9 +++---
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media/filters/audio_file_reader_unittest.cc | 6 ++--
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.../filters/audio_video_metadata_extractor.cc | 11 +++++--
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.../filters/ffmpeg_aac_bitstream_converter.cc | 7 +++--
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...ffmpeg_aac_bitstream_converter_unittest.cc | 2 +-
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media/filters/ffmpeg_audio_decoder.cc | 13 +++++----
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8 files changed, 51 insertions(+), 37 deletions(-)
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diff --git a/media/cdm/library_cdm/clear_key_cdm/ffmpeg_cdm_audio_decoder.cc b/media/cdm/library_cdm/clear_key_cdm/ffmpeg_cdm_audio_decoder.cc
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index e4fc3f460e2..9b1ad9f7675 100644
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--- a/media/cdm/library_cdm/clear_key_cdm/ffmpeg_cdm_audio_decoder.cc
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+++ b/media/cdm/library_cdm/clear_key_cdm/ffmpeg_cdm_audio_decoder.cc
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@@ -74,7 +74,7 @@ void CdmAudioDecoderConfigToAVCodecContext(
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codec_context->sample_fmt = AV_SAMPLE_FMT_NONE;
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}
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- codec_context->channels = config.channel_count;
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+ codec_context->ch_layout.nb_channels = config.channel_count;
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codec_context->sample_rate = config.samples_per_second;
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if (config.extra_data) {
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@@ -124,8 +124,8 @@ void CopySamples(cdm::AudioFormat cdm_format,
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case cdm::kAudioFormatPlanarS16:
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case cdm::kAudioFormatPlanarF32: {
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const int decoded_size_per_channel =
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- decoded_audio_size / av_frame.channels;
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- for (int i = 0; i < av_frame.channels; ++i) {
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+ decoded_audio_size / av_frame.ch_layout.nb_channels;
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+ for (int i = 0; i < av_frame.ch_layout.nb_channels; ++i) {
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memcpy(output_buffer, av_frame.extended_data[i],
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decoded_size_per_channel);
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output_buffer += decoded_size_per_channel;
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@@ -185,13 +185,14 @@ bool FFmpegCdmAudioDecoder::Initialize(
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// Success!
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decoding_loop_.reset(new FFmpegDecodingLoop(codec_context_.get()));
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samples_per_second_ = config.samples_per_second;
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- bytes_per_frame_ = codec_context_->channels * config.bits_per_channel / 8;
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+ bytes_per_frame_ =
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+ codec_context_->ch_layout.nb_channels * config.bits_per_channel / 8;
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output_timestamp_helper_.reset(
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new AudioTimestampHelper(config.samples_per_second));
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is_initialized_ = true;
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// Store initial values to guard against midstream configuration changes.
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- channels_ = codec_context_->channels;
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+ channels_ = codec_context_->ch_layout.nb_channels;
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av_sample_format_ = codec_context_->sample_fmt;
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return true;
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@@ -291,17 +292,18 @@ cdm::Status FFmpegCdmAudioDecoder::DecodeBuffer(
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for (auto& frame : audio_frames) {
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int decoded_audio_size = 0;
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if (frame->sample_rate != samples_per_second_ ||
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- frame->channels != channels_ || frame->format != av_sample_format_) {
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+ frame->ch_layout.nb_channels != channels_ ||
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+ frame->format != av_sample_format_) {
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DLOG(ERROR) << "Unsupported midstream configuration change!"
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<< " Sample Rate: " << frame->sample_rate << " vs "
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- << samples_per_second_ << ", Channels: " << frame->channels
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+ << samples_per_second_ << ", Channels: " << frame->ch_layout.nb_channels
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<< " vs " << channels_ << ", Sample Format: " << frame->format
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<< " vs " << av_sample_format_;
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return cdm::kDecodeError;
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}
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decoded_audio_size = av_samples_get_buffer_size(
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- nullptr, codec_context_->channels, frame->nb_samples,
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+ nullptr, codec_context_->ch_layout.nb_channels, frame->nb_samples,
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codec_context_->sample_fmt, 1);
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if (!decoded_audio_size)
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continue;
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@@ -320,7 +323,7 @@ bool FFmpegCdmAudioDecoder::OnNewFrame(
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size_t* total_size,
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std::vector<std::unique_ptr<AVFrame, ScopedPtrAVFreeFrame>>* audio_frames,
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AVFrame* frame) {
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- *total_size += av_samples_get_buffer_size(nullptr, codec_context_->channels,
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+ *total_size += av_samples_get_buffer_size(nullptr, codec_context_->ch_layout.nb_channels,
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frame->nb_samples,
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codec_context_->sample_fmt, 1);
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audio_frames->emplace_back(av_frame_clone(frame));
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diff --git a/media/ffmpeg/ffmpeg_common.cc b/media/ffmpeg/ffmpeg_common.cc
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index 87ca8969626..76f03d6608e 100644
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--- a/media/ffmpeg/ffmpeg_common.cc
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+++ b/media/ffmpeg/ffmpeg_common.cc
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@@ -345,10 +345,11 @@ bool AVCodecContextToAudioDecoderConfig(const AVCodecContext* codec_context,
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codec_context->sample_fmt, codec_context->codec_id);
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ChannelLayout channel_layout =
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- codec_context->channels > 8
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+ codec_context->ch_layout.nb_channels > 8
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? CHANNEL_LAYOUT_DISCRETE
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- : ChannelLayoutToChromeChannelLayout(codec_context->channel_layout,
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- codec_context->channels);
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+ : ChannelLayoutToChromeChannelLayout(
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+ codec_context->ch_layout.u.mask,
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+ codec_context->ch_layout.nb_channels);
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int sample_rate = codec_context->sample_rate;
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switch (codec) {
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@@ -401,7 +402,7 @@ bool AVCodecContextToAudioDecoderConfig(const AVCodecContext* codec_context,
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extra_data, encryption_scheme, seek_preroll,
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codec_context->delay);
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if (channel_layout == CHANNEL_LAYOUT_DISCRETE)
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- config->SetChannelsForDiscrete(codec_context->channels);
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+ config->SetChannelsForDiscrete(codec_context->ch_layout.nb_channels);
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#if BUILDFLAG(ENABLE_PLATFORM_AC3_EAC3_AUDIO)
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// These are bitstream formats unknown to ffmpeg, so they don't have
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@@ -470,7 +471,7 @@ void AudioDecoderConfigToAVCodecContext(const AudioDecoderConfig& config,
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// TODO(scherkus): should we set |channel_layout|? I'm not sure if FFmpeg uses
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// said information to decode.
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- codec_context->channels = config.channels();
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+ codec_context->ch_layout.nb_channels = config.channels();
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codec_context->sample_rate = config.samples_per_second();
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if (config.extra_data().empty()) {
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diff --git a/media/filters/audio_file_reader.cc b/media/filters/audio_file_reader.cc
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index 5f257bdfaa6..e1be5aa9a5b 100644
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--- a/media/filters/audio_file_reader.cc
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+++ b/media/filters/audio_file_reader.cc
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@@ -113,14 +113,15 @@ bool AudioFileReader::OpenDecoder() {
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// Verify the channel layout is supported by Chrome. Acts as a sanity check
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// against invalid files. See http://crbug.com/171962
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- if (ChannelLayoutToChromeChannelLayout(codec_context_->channel_layout,
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- codec_context_->channels) ==
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+ if (ChannelLayoutToChromeChannelLayout(
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+ codec_context_->ch_layout.u.mask,
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+ codec_context_->ch_layout.nb_channels) ==
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CHANNEL_LAYOUT_UNSUPPORTED) {
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return false;
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}
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// Store initial values to guard against midstream configuration changes.
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- channels_ = codec_context_->channels;
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+ channels_ = codec_context_->ch_layout.nb_channels;
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audio_codec_ = CodecIDToAudioCodec(codec_context_->codec_id);
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sample_rate_ = codec_context_->sample_rate;
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av_sample_format_ = codec_context_->sample_fmt;
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@@ -223,7 +224,7 @@ bool AudioFileReader::OnNewFrame(
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if (frames_read < 0)
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return false;
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- const int channels = frame->channels;
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+ const int channels = frame->ch_layout.nb_channels;
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if (frame->sample_rate != sample_rate_ || channels != channels_ ||
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frame->format != av_sample_format_) {
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DLOG(ERROR) << "Unsupported midstream configuration change!"
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diff --git a/media/filters/ffmpeg_aac_bitstream_converter.cc b/media/filters/ffmpeg_aac_bitstream_converter.cc
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index 6f231c85729..ca5e5fb927d 100644
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--- a/media/filters/ffmpeg_aac_bitstream_converter.cc
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+++ b/media/filters/ffmpeg_aac_bitstream_converter.cc
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@@ -195,14 +195,15 @@ bool FFmpegAACBitstreamConverter::ConvertPacket(AVPacket* packet) {
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if (!header_generated_ || codec_ != stream_codec_parameters_->codec_id ||
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audio_profile_ != stream_codec_parameters_->profile ||
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sample_rate_index_ != sample_rate_index ||
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- channel_configuration_ != stream_codec_parameters_->channels ||
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+ channel_configuration_ !=
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+ stream_codec_parameters_->ch_layout.nb_channels ||
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frame_length_ != header_plus_packet_size) {
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header_generated_ =
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GenerateAdtsHeader(stream_codec_parameters_->codec_id,
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0, // layer
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stream_codec_parameters_->profile, sample_rate_index,
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0, // private stream
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- stream_codec_parameters_->channels,
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+ stream_codec_parameters_->ch_layout.nb_channels,
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0, // originality
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0, // home
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0, // copyrighted_stream
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@@ -214,7 +215,7 @@ bool FFmpegAACBitstreamConverter::ConvertPacket(AVPacket* packet) {
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codec_ = stream_codec_parameters_->codec_id;
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audio_profile_ = stream_codec_parameters_->profile;
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sample_rate_index_ = sample_rate_index;
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- channel_configuration_ = stream_codec_parameters_->channels;
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+ channel_configuration_ = stream_codec_parameters_->ch_layout.nb_channels;
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frame_length_ = header_plus_packet_size;
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}
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diff --git a/media/filters/ffmpeg_aac_bitstream_converter_unittest.cc b/media/filters/ffmpeg_aac_bitstream_converter_unittest.cc
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index 1fd4c5ccd7d..f59bcd8fdaf 100644
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--- a/media/filters/ffmpeg_aac_bitstream_converter_unittest.cc
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+++ b/media/filters/ffmpeg_aac_bitstream_converter_unittest.cc
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@@ -34,7 +34,7 @@ class FFmpegAACBitstreamConverterTest : public testing::Test {
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memset(&test_parameters_, 0, sizeof(AVCodecParameters));
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test_parameters_.codec_id = AV_CODEC_ID_AAC;
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test_parameters_.profile = FF_PROFILE_AAC_MAIN;
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- test_parameters_.channels = 2;
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+ test_parameters_.ch_layout.nb_channels = 2;
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test_parameters_.extradata = extradata_header_;
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test_parameters_.extradata_size = sizeof(extradata_header_);
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}
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diff --git a/media/filters/ffmpeg_audio_decoder.cc b/media/filters/ffmpeg_audio_decoder.cc
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index 6a56c675f7d..4615fdeb3fb 100644
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--- a/media/filters/ffmpeg_audio_decoder.cc
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+++ b/media/filters/ffmpeg_audio_decoder.cc
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@@ -28,7 +28,7 @@ namespace media {
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// Return the number of channels from the data in |frame|.
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static inline int DetermineChannels(AVFrame* frame) {
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- return frame->channels;
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+ return frame->ch_layout.nb_channels;
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}
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// Called by FFmpeg's allocation routine to allocate a buffer. Uses
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@@ -231,7 +231,7 @@ bool FFmpegAudioDecoder::OnNewFrame(const DecoderBuffer& buffer,
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// Translate unsupported into discrete layouts for discrete configurations;
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// ffmpeg does not have a labeled discrete configuration internally.
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ChannelLayout channel_layout = ChannelLayoutToChromeChannelLayout(
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- codec_context_->channel_layout, codec_context_->channels);
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+ codec_context_->ch_layout.u.mask, codec_context_->ch_layout.nb_channels);
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if (channel_layout == CHANNEL_LAYOUT_UNSUPPORTED &&
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config_.channel_layout() == CHANNEL_LAYOUT_DISCRETE) {
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channel_layout = CHANNEL_LAYOUT_DISCRETE;
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@@ -348,11 +348,11 @@ bool FFmpegAudioDecoder::ConfigureDecoder(const AudioDecoderConfig& config) {
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// Success!
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av_sample_format_ = codec_context_->sample_fmt;
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- if (codec_context_->channels != config.channels()) {
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+ if (codec_context_->ch_layout.nb_channels != config.channels()) {
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MEDIA_LOG(ERROR, media_log_)
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<< "Audio configuration specified " << config.channels()
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<< " channels, but FFmpeg thinks the file contains "
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- << codec_context_->channels << " channels";
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+ << codec_context_->ch_layout.nb_channels << " channels";
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ReleaseFFmpegResources();
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state_ = DecoderState::kUninitialized;
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return false;
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@@ -403,7 +403,7 @@ int FFmpegAudioDecoder::GetAudioBuffer(struct AVCodecContext* s,
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if (frame->nb_samples <= 0)
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return AVERROR(EINVAL);
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- if (s->channels != channels) {
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+ if (s->ch_layout.nb_channels != channels) {
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DLOG(ERROR) << "AVCodecContext and AVFrame disagree on channel count.";
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return AVERROR(EINVAL);
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}
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@@ -436,7 +436,8 @@ int FFmpegAudioDecoder::GetAudioBuffer(struct AVCodecContext* s,
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ChannelLayout channel_layout =
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config_.channel_layout() == CHANNEL_LAYOUT_DISCRETE
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? CHANNEL_LAYOUT_DISCRETE
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- : ChannelLayoutToChromeChannelLayout(s->channel_layout, s->channels);
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+ : ChannelLayoutToChromeChannelLayout(s->ch_layout.u.mask,
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+ s->ch_layout.nb_channels);
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if (channel_layout == CHANNEL_LAYOUT_UNSUPPORTED) {
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DLOG(ERROR) << "Unsupported channel layout.";
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commit 62274859104bd828373ae406aa9309e610449ac5
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Author: Ted Meyer <tmathmeyer@chromium.org>
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Date: Fri Mar 22 19:56:55 2024 +0000
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Replace deprecated use of AVCodecContext::reordered_opaque
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We can use the AV_CODEC_FLAG_COPY_OPAQUE flag on the codec context
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now to trigger timestamp propagation.
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Bug: 330573128
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Change-Id: I6bc57241a35ab5283742aad8d42acb4dc5e85858
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Reviewed-on: https://chromium-review.googlesource.com/c/chromium/src/+/5384308
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Commit-Queue: Ted (Chromium) Meyer <tmathmeyer@chromium.org>
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Reviewed-by: Dan Sanders <sandersd@chromium.org>
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Cr-Commit-Position: refs/heads/main@{#1277051}
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diff --git a/media/filters/ffmpeg_video_decoder.cc b/media/filters/ffmpeg_video_decoder.cc
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index bd75477feeabb..8a658a58caac5 100644
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--- a/media/filters/ffmpeg_video_decoder.cc
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+++ b/media/filters/ffmpeg_video_decoder.cc
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@@ -134,7 +134,7 @@ bool FFmpegVideoDecoder::IsCodecSupported(VideoCodec codec) {
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}
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FFmpegVideoDecoder::FFmpegVideoDecoder(MediaLog* media_log)
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- : media_log_(media_log), state_(kUninitialized), decode_nalus_(false) {
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+ : media_log_(media_log), state_(kUninitialized), decode_nalus_(false), timestamp_map_(128) {
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DVLOG(1) << __func__;
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thread_checker_.DetachFromThread();
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}
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@@ -363,8 +363,10 @@ bool FFmpegVideoDecoder::FFmpegDecode(const DecoderBuffer& buffer) {
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DCHECK(packet.data);
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DCHECK_GT(packet.size, 0);
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- // Let FFmpeg handle presentation timestamp reordering.
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- codec_context_->reordered_opaque = buffer.timestamp().InMicroseconds();
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+ const int64_t timestamp = buffer.timestamp().InMicroseconds();
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+ const TimestampId timestamp_id = timestamp_id_generator_.GenerateNextId();
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+ timestamp_map_.Put(timestamp_id, timestamp);
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+ packet.opaque = reinterpret_cast<void*>(timestamp_id.GetUnsafeValue());
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}
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FFmpegDecodingLoop::DecodeStatus decode_status = decoding_loop_->DecodePacket(
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packet, base::BindRepeating(&FFmpegVideoDecoder::OnNewFrame,
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@@ -423,8 +425,13 @@ bool FFmpegVideoDecoder::OnNewFrame(AVFrame* frame) {
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scoped_refptr<VideoFrame> video_frame =
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reinterpret_cast<VideoFrame*>(av_buffer_get_opaque(frame->buf[0]));
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+ const auto ts_id = TimestampId(reinterpret_cast<size_t>(frame->opaque));
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+ const auto ts_lookup = timestamp_map_.Get(ts_id);
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+ if (ts_lookup == timestamp_map_.end()) {
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+ return false;
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+ }
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video_frame->set_timestamp(
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- base::TimeDelta::FromMicroseconds(frame->reordered_opaque));
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+ base::TimeDelta::FromMicroseconds(std::get<1>(*ts_lookup)));
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video_frame->metadata()->power_efficient = false;
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output_cb_.Run(video_frame);
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return true;
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@@ -498,8 +505,10 @@ bool FFmpegVideoDecoder::ConfigureDecoder(const VideoDecoderConfig& config,
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codec_context_->thread_count = GetFFmpegVideoDecoderThreadCount(config);
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codec_context_->thread_type =
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FF_THREAD_SLICE | (low_delay ? 0 : FF_THREAD_FRAME);
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+
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codec_context_->opaque = this;
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codec_context_->get_buffer2 = GetVideoBufferImpl;
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+ codec_context_->flags |= AV_CODEC_FLAG_COPY_OPAQUE;
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if (base::FeatureList::IsEnabled(kFFmpegAllowLists)) {
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// Note: FFmpeg will try to free this string, so we must duplicate it.
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diff --git a/media/filters/ffmpeg_video_decoder.h b/media/filters/ffmpeg_video_decoder.h
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index d02cb89c3ddf7..0a2de1c623fff 100644
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--- a/media/filters/ffmpeg_video_decoder.h
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+++ b/media/filters/ffmpeg_video_decoder.h
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@@ -7,6 +7,8 @@
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#include <list>
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#include <memory>
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+#include "base/containers/mru_cache.h"
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+#include "base/util/type_safety/id_type.h"
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#include "base/callback.h"
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#include "base/macros.h"
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#include "base/memory/ref_counted.h"
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@@ -87,6 +89,20 @@ class MEDIA_EXPORT FFmpegVideoDecoder : public VideoDecoder {
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// FFmpeg structures owned by this object.
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std::unique_ptr<AVCodecContext, ScopedPtrAVFreeContext> codec_context_;
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+ // The gist here is that timestamps need to be 64 bits to store microsecond
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+ // precision. A 32 bit integer would overflow at ~35 minutes at this level of
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+ // precision. We can't cast the timestamp to the void ptr object used by the
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+ // opaque field in ffmpeg then, because it would lose data on a 32 bit build.
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+ // However, we don't actually have 2^31 timestamped frames in a single
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+ // playback, so it's fine to use the 32 bit value as a key in a map which
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+ // contains the actual timestamps. Additionally, we've in the past set 128
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+ // outstanding frames for re-ordering as a limit for cross-thread decoding
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+ // tasks, so we'll do that here too with the LRU cache.
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+ using TimestampId = util::IdType<int64_t, size_t, 0>;
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+
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+ TimestampId::Generator timestamp_id_generator_;
|
|
+ base::MRUCache<TimestampId, int64_t> timestamp_map_;
|
|
+
|
|
VideoDecoderConfig config_;
|
|
|
|
scoped_refptr<FrameBufferPool> frame_pool_;
|
|
diff --git a/media/ffmpeg/ffmpeg_common.cc b/media/ffmpeg/ffmpeg_common.cc
|
|
index 3331581a6fee6..69539fd6594ec 100644
|
|
--- a/media/ffmpeg/ffmpeg_common.cc
|
|
+++ b/media/ffmpeg/ffmpeg_common.cc
|
|
@@ -404,7 +404,9 @@ bool AVCodecContextToAudioDecoderConfig(const AVCodecContext* codec_context,
|
|
#if BUILDFLAG(USE_PROPRIETARY_CODECS)
|
|
// TODO(dalecurtis): Just use the profile from the codec context if ffmpeg
|
|
// ever starts supporting xHE-AAC.
|
|
- if (codec == kCodecAAC && codec_context->profile == FF_PROFILE_UNKNOWN) {
|
|
+ constexpr uint8_t kXHEAAc = 41;
|
|
+ if (codec == kCodecAAC && codec_context->profile == FF_PROFILE_UNKNOWN ||
|
|
+ codec_context->profile == kXHEAAc) {
|
|
// Errors aren't fatal here, so just drop any MediaLog messages.
|
|
NullMediaLog media_log;
|
|
mp4::AAC aac_parser;
|
|
diff --git a/media/ffmpeg/ffmpeg_regression_tests.cc b/media/ffmpeg/ffmpeg_regression_tests.cc
|
|
index 05dcb1cd62c75..866f446698947 100644
|
|
--- a/media/ffmpeg/ffmpeg_regression_tests.cc
|
|
+++ b/media/ffmpeg/ffmpeg_regression_tests.cc
|
|
@@ -90,16 +90,16 @@ FFMPEG_TEST_CASE(Cr62127,
|
|
PIPELINE_ERROR_DECODE,
|
|
PIPELINE_ERROR_DECODE);
|
|
FFMPEG_TEST_CASE(Cr93620, "security/93620.ogg", PIPELINE_OK, PIPELINE_OK);
|
|
-FFMPEG_TEST_CASE(Cr100492,
|
|
- "security/100492.webm",
|
|
- DECODER_ERROR_NOT_SUPPORTED,
|
|
- DECODER_ERROR_NOT_SUPPORTED);
|
|
+FFMPEG_TEST_CASE(Cr100492, "security/100492.webm", PIPELINE_OK, PIPELINE_OK);
|
|
FFMPEG_TEST_CASE(Cr100543, "security/100543.webm", PIPELINE_OK, PIPELINE_OK);
|
|
FFMPEG_TEST_CASE(Cr101458,
|
|
"security/101458.webm",
|
|
PIPELINE_ERROR_DECODE,
|
|
PIPELINE_ERROR_DECODE);
|
|
-FFMPEG_TEST_CASE(Cr108416, "security/108416.webm", PIPELINE_OK, PIPELINE_OK);
|
|
+FFMPEG_TEST_CASE(Cr108416,
|
|
+ "security/108416.webm",
|
|
+ PIPELINE_ERROR_DECODE,
|
|
+ PIPELINE_ERROR_DECODE);
|
|
FFMPEG_TEST_CASE(Cr110849,
|
|
"security/110849.mkv",
|
|
DEMUXER_ERROR_COULD_NOT_OPEN,
|
|
@@ -154,7 +154,10 @@ FFMPEG_TEST_CASE(Cr234630b,
|
|
"security/234630b.mov",
|
|
DEMUXER_ERROR_NO_SUPPORTED_STREAMS,
|
|
DEMUXER_ERROR_NO_SUPPORTED_STREAMS);
|
|
-FFMPEG_TEST_CASE(Cr242786, "security/242786.webm", PIPELINE_OK, PIPELINE_OK);
|
|
+FFMPEG_TEST_CASE(Cr242786,
|
|
+ "security/242786.webm",
|
|
+ PIPELINE_OK,
|
|
+ PIPELINE_ERROR_DECODE);
|
|
// Test for out-of-bounds access with slightly corrupt file (detection logic
|
|
// thinks it's a MONO file, but actually contains STEREO audio).
|
|
FFMPEG_TEST_CASE(Cr275590,
|
|
@@ -372,8 +375,8 @@ FFMPEG_TEST_CASE(WEBM_2,
|
|
DEMUXER_ERROR_NO_SUPPORTED_STREAMS);
|
|
FFMPEG_TEST_CASE(WEBM_4,
|
|
"security/out.webm.68798.1929",
|
|
- DECODER_ERROR_NOT_SUPPORTED,
|
|
- DECODER_ERROR_NOT_SUPPORTED);
|
|
+ PIPELINE_OK,
|
|
+ PIPELINE_OK);
|
|
FFMPEG_TEST_CASE(WEBM_5, "frame_size_change.webm", PIPELINE_OK, PIPELINE_OK);
|
|
|
|
// General MKV test cases.
|
|
diff --git a/media/filters/ffmpeg_video_decoder.cc b/media/filters/ffmpeg_video_decoder.cc
|
|
index 8a658a58caac5..9d6ed8aeb5c48 100644
|
|
--- a/media/filters/ffmpeg_video_decoder.cc
|
|
+++ b/media/filters/ffmpeg_video_decoder.cc
|
|
@@ -213,7 +213,6 @@ int FFmpegVideoDecoder::GetVideoBuffer(struct AVCodecContext* codec_context,
|
|
frame->width = coded_size.width();
|
|
frame->height = coded_size.height();
|
|
frame->format = codec_context->pix_fmt;
|
|
- frame->reordered_opaque = codec_context->reordered_opaque;
|
|
|
|
// Now create an AVBufferRef for the data just allocated. It will own the
|
|
// reference to the VideoFrame object.
|
|
diff --git a/media/filters/audio_file_reader.cc b/media/filters/audio_file_reader.cc
|
|
index e1be5aa9a5b13..951c003956fb5 100644
|
|
--- a/media/filters/audio_file_reader.cc
|
|
+++ b/media/filters/audio_file_reader.cc
|
|
@@ -243,10 +243,10 @@ bool AudioFileReader::OnNewFrame(
|
|
// silence from being output. In the case where we are also discarding some
|
|
// portion of the packet (as indicated by a negative pts), we further want to
|
|
// adjust the duration downward by however much exists before zero.
|
|
- if (audio_codec_ == kCodecAAC && frame->pkt_duration) {
|
|
+ if (audio_codec_ == kCodecAAC && frame->duration) {
|
|
const base::TimeDelta pkt_duration = ConvertFromTimeBase(
|
|
glue_->format_context()->streams[stream_index_]->time_base,
|
|
- frame->pkt_duration + std::min(static_cast<int64_t>(0), frame->pts));
|
|
+ frame->duration + std::min(static_cast<int64_t>(0), frame->pts));
|
|
const base::TimeDelta frame_duration = base::TimeDelta::FromSecondsD(
|
|
frames_read / static_cast<double>(sample_rate_));
|
|
|
|
diff --git a/chromium/third_party/webrtc/modules/video_coding/codecs/h264/h264_decoder_impl.cc b/chromium/third_party/webrtc/modules/video_coding/codecs/h264/h264_decoder_impl.cc
|
|
index c6446c2..805b95b 100644
|
|
--- a/third_party/webrtc/modules/video_coding/codecs/h264/h264_decoder_impl.cc
|
|
+++ b/third_party/webrtc/modules/video_coding/codecs/h264/h264_decoder_impl.cc
|
|
@@ -114,7 +114,6 @@ int H264DecoderImpl::AVGetBuffer2(AVCodecContext* context,
|
|
int total_size = y_size + 2 * uv_size;
|
|
|
|
av_frame->format = context->pix_fmt;
|
|
- av_frame->reordered_opaque = context->reordered_opaque;
|
|
|
|
// Set |av_frame| members as required by FFmpeg.
|
|
av_frame->data[kYPlaneIndex] = frame_buffer->MutableDataY();
|
|
@@ -273,8 +272,6 @@ int32_t H264DecoderImpl::Decode(const EncodedImage& input_image,
|
|
return WEBRTC_VIDEO_CODEC_ERROR;
|
|
}
|
|
packet.size = static_cast<int>(input_image.size());
|
|
- int64_t frame_timestamp_us = input_image.ntp_time_ms_ * 1000; // ms -> μs
|
|
- av_context_->reordered_opaque = frame_timestamp_us;
|
|
|
|
int result = avcodec_send_packet(av_context_.get(), &packet);
|
|
if (result < 0) {
|
|
@@ -290,10 +287,6 @@ int32_t H264DecoderImpl::Decode(const EncodedImage& input_image,
|
|
return WEBRTC_VIDEO_CODEC_ERROR;
|
|
}
|
|
|
|
- // We don't expect reordering. Decoded frame tamestamp should match
|
|
- // the input one.
|
|
- RTC_DCHECK_EQ(av_frame_->reordered_opaque, frame_timestamp_us);
|
|
-
|
|
absl::optional<uint8_t> qp;
|
|
// TODO(sakal): Maybe it is possible to get QP directly from FFmpeg.
|
|
h264_bitstream_parser_.ParseBitstream(input_image.data(), input_image.size());
|