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PKGBUILDs/extra/qt5-webengine/qt5-webengine-ffmpeg7.patch
Kevin Mihelich 3555050152 extra/qt5-webengine to 5.15.17-4
2024-05-29 18:21:25 +00:00

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From 6e554a30893150793c2638e3689cf208ffc8e375 Mon Sep 17 00:00:00 2001
From: Dale Curtis <dalecurtis@chromium.org>
Date: Sat, 2 Apr 2022 05:13:53 +0000
Subject: [PATCH] Roll src/third_party/ffmpeg/ 574c39cce..32b2d1d526 (1125
commits)
https://chromium.googlesource.com/chromium/third_party/ffmpeg.git/+log/574c39cce323..32b2d1d526
Created with:
roll-dep src/third_party/ffmpeg
Fixed: 1293918
Cq-Include-Trybots: luci.chromium.try:mac_chromium_asan_rel_ng,linux_chromium_asan_rel_ng,linux_chromium_chromeos_asan_rel_ng
Change-Id: I41945d0f963e3d1f65940067bac22f63b68e37d2
Reviewed-on: https://chromium-review.googlesource.com/c/chromium/src/+/3565647
Auto-Submit: Dale Curtis <dalecurtis@chromium.org>
Reviewed-by: Dan Sanders <sandersd@chromium.org>
Commit-Queue: Dale Curtis <dalecurtis@chromium.org>
Cr-Commit-Position: refs/heads/main@{#988253}
---
.../clear_key_cdm/ffmpeg_cdm_audio_decoder.cc | 29 ++++++++++---------
media/ffmpeg/ffmpeg_common.cc | 11 +++----
media/filters/audio_file_reader.cc | 9 +++---
media/filters/audio_file_reader_unittest.cc | 6 ++--
.../filters/audio_video_metadata_extractor.cc | 11 +++++--
.../filters/ffmpeg_aac_bitstream_converter.cc | 7 +++--
...ffmpeg_aac_bitstream_converter_unittest.cc | 2 +-
media/filters/ffmpeg_audio_decoder.cc | 13 +++++----
8 files changed, 51 insertions(+), 37 deletions(-)
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
index e4fc3f460e2..9b1ad9f7675 100644
--- 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
@@ -74,7 +74,7 @@ void CdmAudioDecoderConfigToAVCodecContext(
codec_context->sample_fmt = AV_SAMPLE_FMT_NONE;
}
- codec_context->channels = config.channel_count;
+ codec_context->ch_layout.nb_channels = config.channel_count;
codec_context->sample_rate = config.samples_per_second;
if (config.extra_data) {
@@ -124,8 +124,8 @@ void CopySamples(cdm::AudioFormat cdm_format,
case cdm::kAudioFormatPlanarS16:
case cdm::kAudioFormatPlanarF32: {
const int decoded_size_per_channel =
- decoded_audio_size / av_frame.channels;
- for (int i = 0; i < av_frame.channels; ++i) {
+ decoded_audio_size / av_frame.ch_layout.nb_channels;
+ for (int i = 0; i < av_frame.ch_layout.nb_channels; ++i) {
memcpy(output_buffer, av_frame.extended_data[i],
decoded_size_per_channel);
output_buffer += decoded_size_per_channel;
@@ -185,13 +185,14 @@ bool FFmpegCdmAudioDecoder::Initialize(
// Success!
decoding_loop_.reset(new FFmpegDecodingLoop(codec_context_.get()));
samples_per_second_ = config.samples_per_second;
- bytes_per_frame_ = codec_context_->channels * config.bits_per_channel / 8;
+ bytes_per_frame_ =
+ codec_context_->ch_layout.nb_channels * config.bits_per_channel / 8;
output_timestamp_helper_.reset(
new AudioTimestampHelper(config.samples_per_second));
is_initialized_ = true;
// Store initial values to guard against midstream configuration changes.
- channels_ = codec_context_->channels;
+ channels_ = codec_context_->ch_layout.nb_channels;
av_sample_format_ = codec_context_->sample_fmt;
return true;
@@ -291,17 +292,18 @@ cdm::Status FFmpegCdmAudioDecoder::DecodeBuffer(
for (auto& frame : audio_frames) {
int decoded_audio_size = 0;
if (frame->sample_rate != samples_per_second_ ||
- frame->channels != channels_ || frame->format != av_sample_format_) {
+ frame->ch_layout.nb_channels != channels_ ||
+ frame->format != av_sample_format_) {
DLOG(ERROR) << "Unsupported midstream configuration change!"
<< " Sample Rate: " << frame->sample_rate << " vs "
- << samples_per_second_ << ", Channels: " << frame->channels
+ << samples_per_second_ << ", Channels: " << frame->ch_layout.nb_channels
<< " vs " << channels_ << ", Sample Format: " << frame->format
<< " vs " << av_sample_format_;
return cdm::kDecodeError;
}
decoded_audio_size = av_samples_get_buffer_size(
- nullptr, codec_context_->channels, frame->nb_samples,
+ nullptr, codec_context_->ch_layout.nb_channels, frame->nb_samples,
codec_context_->sample_fmt, 1);
if (!decoded_audio_size)
continue;
@@ -320,7 +323,7 @@ bool FFmpegCdmAudioDecoder::OnNewFrame(
size_t* total_size,
std::vector<std::unique_ptr<AVFrame, ScopedPtrAVFreeFrame>>* audio_frames,
AVFrame* frame) {
- *total_size += av_samples_get_buffer_size(nullptr, codec_context_->channels,
+ *total_size += av_samples_get_buffer_size(nullptr, codec_context_->ch_layout.nb_channels,
frame->nb_samples,
codec_context_->sample_fmt, 1);
audio_frames->emplace_back(av_frame_clone(frame));
diff --git a/media/ffmpeg/ffmpeg_common.cc b/media/ffmpeg/ffmpeg_common.cc
index 87ca8969626..76f03d6608e 100644
--- a/media/ffmpeg/ffmpeg_common.cc
+++ b/media/ffmpeg/ffmpeg_common.cc
@@ -345,10 +345,11 @@ bool AVCodecContextToAudioDecoderConfig(const AVCodecContext* codec_context,
codec_context->sample_fmt, codec_context->codec_id);
ChannelLayout channel_layout =
- codec_context->channels > 8
+ codec_context->ch_layout.nb_channels > 8
? CHANNEL_LAYOUT_DISCRETE
- : ChannelLayoutToChromeChannelLayout(codec_context->channel_layout,
- codec_context->channels);
+ : ChannelLayoutToChromeChannelLayout(
+ codec_context->ch_layout.u.mask,
+ codec_context->ch_layout.nb_channels);
int sample_rate = codec_context->sample_rate;
switch (codec) {
@@ -401,7 +402,7 @@ bool AVCodecContextToAudioDecoderConfig(const AVCodecContext* codec_context,
extra_data, encryption_scheme, seek_preroll,
codec_context->delay);
if (channel_layout == CHANNEL_LAYOUT_DISCRETE)
- config->SetChannelsForDiscrete(codec_context->channels);
+ config->SetChannelsForDiscrete(codec_context->ch_layout.nb_channels);
#if BUILDFLAG(ENABLE_PLATFORM_AC3_EAC3_AUDIO)
// These are bitstream formats unknown to ffmpeg, so they don't have
@@ -470,7 +471,7 @@ void AudioDecoderConfigToAVCodecContext(const AudioDecoderConfig& config,
// TODO(scherkus): should we set |channel_layout|? I'm not sure if FFmpeg uses
// said information to decode.
- codec_context->channels = config.channels();
+ codec_context->ch_layout.nb_channels = config.channels();
codec_context->sample_rate = config.samples_per_second();
if (config.extra_data().empty()) {
diff --git a/media/filters/audio_file_reader.cc b/media/filters/audio_file_reader.cc
index 5f257bdfaa6..e1be5aa9a5b 100644
--- a/media/filters/audio_file_reader.cc
+++ b/media/filters/audio_file_reader.cc
@@ -113,14 +113,15 @@ bool AudioFileReader::OpenDecoder() {
// Verify the channel layout is supported by Chrome. Acts as a sanity check
// against invalid files. See http://crbug.com/171962
- if (ChannelLayoutToChromeChannelLayout(codec_context_->channel_layout,
- codec_context_->channels) ==
+ if (ChannelLayoutToChromeChannelLayout(
+ codec_context_->ch_layout.u.mask,
+ codec_context_->ch_layout.nb_channels) ==
CHANNEL_LAYOUT_UNSUPPORTED) {
return false;
}
// Store initial values to guard against midstream configuration changes.
- channels_ = codec_context_->channels;
+ channels_ = codec_context_->ch_layout.nb_channels;
audio_codec_ = CodecIDToAudioCodec(codec_context_->codec_id);
sample_rate_ = codec_context_->sample_rate;
av_sample_format_ = codec_context_->sample_fmt;
@@ -223,7 +224,7 @@ bool AudioFileReader::OnNewFrame(
if (frames_read < 0)
return false;
- const int channels = frame->channels;
+ const int channels = frame->ch_layout.nb_channels;
if (frame->sample_rate != sample_rate_ || channels != channels_ ||
frame->format != av_sample_format_) {
DLOG(ERROR) << "Unsupported midstream configuration change!"
diff --git a/media/filters/ffmpeg_aac_bitstream_converter.cc b/media/filters/ffmpeg_aac_bitstream_converter.cc
index 6f231c85729..ca5e5fb927d 100644
--- a/media/filters/ffmpeg_aac_bitstream_converter.cc
+++ b/media/filters/ffmpeg_aac_bitstream_converter.cc
@@ -195,14 +195,15 @@ bool FFmpegAACBitstreamConverter::ConvertPacket(AVPacket* packet) {
if (!header_generated_ || codec_ != stream_codec_parameters_->codec_id ||
audio_profile_ != stream_codec_parameters_->profile ||
sample_rate_index_ != sample_rate_index ||
- channel_configuration_ != stream_codec_parameters_->channels ||
+ channel_configuration_ !=
+ stream_codec_parameters_->ch_layout.nb_channels ||
frame_length_ != header_plus_packet_size) {
header_generated_ =
GenerateAdtsHeader(stream_codec_parameters_->codec_id,
0, // layer
stream_codec_parameters_->profile, sample_rate_index,
0, // private stream
- stream_codec_parameters_->channels,
+ stream_codec_parameters_->ch_layout.nb_channels,
0, // originality
0, // home
0, // copyrighted_stream
@@ -214,7 +215,7 @@ bool FFmpegAACBitstreamConverter::ConvertPacket(AVPacket* packet) {
codec_ = stream_codec_parameters_->codec_id;
audio_profile_ = stream_codec_parameters_->profile;
sample_rate_index_ = sample_rate_index;
- channel_configuration_ = stream_codec_parameters_->channels;
+ channel_configuration_ = stream_codec_parameters_->ch_layout.nb_channels;
frame_length_ = header_plus_packet_size;
}
diff --git a/media/filters/ffmpeg_aac_bitstream_converter_unittest.cc b/media/filters/ffmpeg_aac_bitstream_converter_unittest.cc
index 1fd4c5ccd7d..f59bcd8fdaf 100644
--- a/media/filters/ffmpeg_aac_bitstream_converter_unittest.cc
+++ b/media/filters/ffmpeg_aac_bitstream_converter_unittest.cc
@@ -34,7 +34,7 @@ class FFmpegAACBitstreamConverterTest : public testing::Test {
memset(&test_parameters_, 0, sizeof(AVCodecParameters));
test_parameters_.codec_id = AV_CODEC_ID_AAC;
test_parameters_.profile = FF_PROFILE_AAC_MAIN;
- test_parameters_.channels = 2;
+ test_parameters_.ch_layout.nb_channels = 2;
test_parameters_.extradata = extradata_header_;
test_parameters_.extradata_size = sizeof(extradata_header_);
}
diff --git a/media/filters/ffmpeg_audio_decoder.cc b/media/filters/ffmpeg_audio_decoder.cc
index 6a56c675f7d..4615fdeb3fb 100644
--- a/media/filters/ffmpeg_audio_decoder.cc
+++ b/media/filters/ffmpeg_audio_decoder.cc
@@ -28,7 +28,7 @@ namespace media {
// Return the number of channels from the data in |frame|.
static inline int DetermineChannels(AVFrame* frame) {
- return frame->channels;
+ return frame->ch_layout.nb_channels;
}
// Called by FFmpeg's allocation routine to allocate a buffer. Uses
@@ -231,7 +231,7 @@ bool FFmpegAudioDecoder::OnNewFrame(const DecoderBuffer& buffer,
// Translate unsupported into discrete layouts for discrete configurations;
// ffmpeg does not have a labeled discrete configuration internally.
ChannelLayout channel_layout = ChannelLayoutToChromeChannelLayout(
- codec_context_->channel_layout, codec_context_->channels);
+ codec_context_->ch_layout.u.mask, codec_context_->ch_layout.nb_channels);
if (channel_layout == CHANNEL_LAYOUT_UNSUPPORTED &&
config_.channel_layout() == CHANNEL_LAYOUT_DISCRETE) {
channel_layout = CHANNEL_LAYOUT_DISCRETE;
@@ -348,11 +348,11 @@ bool FFmpegAudioDecoder::ConfigureDecoder(const AudioDecoderConfig& config) {
// Success!
av_sample_format_ = codec_context_->sample_fmt;
- if (codec_context_->channels != config.channels()) {
+ if (codec_context_->ch_layout.nb_channels != config.channels()) {
MEDIA_LOG(ERROR, media_log_)
<< "Audio configuration specified " << config.channels()
<< " channels, but FFmpeg thinks the file contains "
- << codec_context_->channels << " channels";
+ << codec_context_->ch_layout.nb_channels << " channels";
ReleaseFFmpegResources();
state_ = DecoderState::kUninitialized;
return false;
@@ -403,7 +403,7 @@ int FFmpegAudioDecoder::GetAudioBuffer(struct AVCodecContext* s,
if (frame->nb_samples <= 0)
return AVERROR(EINVAL);
- if (s->channels != channels) {
+ if (s->ch_layout.nb_channels != channels) {
DLOG(ERROR) << "AVCodecContext and AVFrame disagree on channel count.";
return AVERROR(EINVAL);
}
@@ -436,7 +436,8 @@ int FFmpegAudioDecoder::GetAudioBuffer(struct AVCodecContext* s,
ChannelLayout channel_layout =
config_.channel_layout() == CHANNEL_LAYOUT_DISCRETE
? CHANNEL_LAYOUT_DISCRETE
- : ChannelLayoutToChromeChannelLayout(s->channel_layout, s->channels);
+ : ChannelLayoutToChromeChannelLayout(s->ch_layout.u.mask,
+ s->ch_layout.nb_channels);
if (channel_layout == CHANNEL_LAYOUT_UNSUPPORTED) {
DLOG(ERROR) << "Unsupported channel layout.";
commit 62274859104bd828373ae406aa9309e610449ac5
Author: Ted Meyer <tmathmeyer@chromium.org>
Date: Fri Mar 22 19:56:55 2024 +0000
Replace deprecated use of AVCodecContext::reordered_opaque
We can use the AV_CODEC_FLAG_COPY_OPAQUE flag on the codec context
now to trigger timestamp propagation.
Bug: 330573128
Change-Id: I6bc57241a35ab5283742aad8d42acb4dc5e85858
Reviewed-on: https://chromium-review.googlesource.com/c/chromium/src/+/5384308
Commit-Queue: Ted (Chromium) Meyer <tmathmeyer@chromium.org>
Reviewed-by: Dan Sanders <sandersd@chromium.org>
Cr-Commit-Position: refs/heads/main@{#1277051}
diff --git a/media/filters/ffmpeg_video_decoder.cc b/media/filters/ffmpeg_video_decoder.cc
index bd75477feeabb..8a658a58caac5 100644
--- a/media/filters/ffmpeg_video_decoder.cc
+++ b/media/filters/ffmpeg_video_decoder.cc
@@ -134,7 +134,7 @@ bool FFmpegVideoDecoder::IsCodecSupported(VideoCodec codec) {
}
FFmpegVideoDecoder::FFmpegVideoDecoder(MediaLog* media_log)
- : media_log_(media_log), state_(kUninitialized), decode_nalus_(false) {
+ : media_log_(media_log), state_(kUninitialized), decode_nalus_(false), timestamp_map_(128) {
DVLOG(1) << __func__;
thread_checker_.DetachFromThread();
}
@@ -363,8 +363,10 @@ bool FFmpegVideoDecoder::FFmpegDecode(const DecoderBuffer& buffer) {
DCHECK(packet.data);
DCHECK_GT(packet.size, 0);
- // Let FFmpeg handle presentation timestamp reordering.
- codec_context_->reordered_opaque = buffer.timestamp().InMicroseconds();
+ const int64_t timestamp = buffer.timestamp().InMicroseconds();
+ const TimestampId timestamp_id = timestamp_id_generator_.GenerateNextId();
+ timestamp_map_.Put(timestamp_id, timestamp);
+ packet.opaque = reinterpret_cast<void*>(timestamp_id.GetUnsafeValue());
}
FFmpegDecodingLoop::DecodeStatus decode_status = decoding_loop_->DecodePacket(
packet, base::BindRepeating(&FFmpegVideoDecoder::OnNewFrame,
@@ -423,8 +425,13 @@ bool FFmpegVideoDecoder::OnNewFrame(AVFrame* frame) {
scoped_refptr<VideoFrame> video_frame =
reinterpret_cast<VideoFrame*>(av_buffer_get_opaque(frame->buf[0]));
+ const auto ts_id = TimestampId(reinterpret_cast<size_t>(frame->opaque));
+ const auto ts_lookup = timestamp_map_.Get(ts_id);
+ if (ts_lookup == timestamp_map_.end()) {
+ return false;
+ }
video_frame->set_timestamp(
- base::TimeDelta::FromMicroseconds(frame->reordered_opaque));
+ base::TimeDelta::FromMicroseconds(std::get<1>(*ts_lookup)));
video_frame->metadata()->power_efficient = false;
output_cb_.Run(video_frame);
return true;
@@ -498,8 +505,10 @@ bool FFmpegVideoDecoder::ConfigureDecoder(const VideoDecoderConfig& config,
codec_context_->thread_count = GetFFmpegVideoDecoderThreadCount(config);
codec_context_->thread_type =
FF_THREAD_SLICE | (low_delay ? 0 : FF_THREAD_FRAME);
+
codec_context_->opaque = this;
codec_context_->get_buffer2 = GetVideoBufferImpl;
+ codec_context_->flags |= AV_CODEC_FLAG_COPY_OPAQUE;
if (base::FeatureList::IsEnabled(kFFmpegAllowLists)) {
// Note: FFmpeg will try to free this string, so we must duplicate it.
diff --git a/media/filters/ffmpeg_video_decoder.h b/media/filters/ffmpeg_video_decoder.h
index d02cb89c3ddf7..0a2de1c623fff 100644
--- a/media/filters/ffmpeg_video_decoder.h
+++ b/media/filters/ffmpeg_video_decoder.h
@@ -7,6 +7,8 @@
#include <list>
#include <memory>
+#include "base/containers/mru_cache.h"
+#include "base/util/type_safety/id_type.h"
#include "base/callback.h"
#include "base/macros.h"
#include "base/memory/ref_counted.h"
@@ -87,6 +89,20 @@ class MEDIA_EXPORT FFmpegVideoDecoder : public VideoDecoder {
// FFmpeg structures owned by this object.
std::unique_ptr<AVCodecContext, ScopedPtrAVFreeContext> codec_context_;
+ // The gist here is that timestamps need to be 64 bits to store microsecond
+ // precision. A 32 bit integer would overflow at ~35 minutes at this level of
+ // precision. We can't cast the timestamp to the void ptr object used by the
+ // opaque field in ffmpeg then, because it would lose data on a 32 bit build.
+ // However, we don't actually have 2^31 timestamped frames in a single
+ // playback, so it's fine to use the 32 bit value as a key in a map which
+ // contains the actual timestamps. Additionally, we've in the past set 128
+ // outstanding frames for re-ordering as a limit for cross-thread decoding
+ // tasks, so we'll do that here too with the LRU cache.
+ using TimestampId = util::IdType<int64_t, size_t, 0>;
+
+ 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_));
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