File qt5-webengine-ffmpeg7.patch of Package libqt5-qtwebengine

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File qt5-webengine-ffmpeg7.patch of Package libqt5-qtwebengine (Revision f030ef36cde615da547e30ca6021e819)

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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/src/3rdparty/chromium/media/cdm/library_cdm/clear_key_cdm/ffmpeg_cdm_audio_decoder.cc b/src/3rdparty/chromium/media/cdm/library_cdm/clear_key_cdm/ffmpeg_cdm_audio_decoder.cc
index a043005..9ae2ca9 100644
--- a/src/3rdparty/chromium/media/cdm/library_cdm/clear_key_cdm/ffmpeg_cdm_audio_decoder.cc
+++ b/src/3rdparty/chromium/media/cdm/library_cdm/clear_key_cdm/ffmpeg_cdm_audio_decoder.cc
@@ -73,7 +73,7 @@ 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) {
@@ -123,8 +123,8 @@ 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;
@@ -184,13 +184,14 @@ 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;
@@ -290,17 +291,18 @@ 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;
@@ -319,7 +321,7 @@ 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/src/3rdparty/chromium/media/ffmpeg/ffmpeg_common.cc b/src/3rdparty/chromium/media/ffmpeg/ffmpeg_common.cc
index c17dd9f..0448cb5 100644
--- a/src/3rdparty/chromium/media/ffmpeg/ffmpeg_common.cc
+++ b/src/3rdparty/chromium/media/ffmpeg/ffmpeg_common.cc
@@ -341,10 +341,11 @@ 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) {
@@ -397,7 +398,7 @@ 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
@@ -462,7 +463,7 @@ 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/src/3rdparty/chromium/media/filters/audio_file_reader.cc b/src/3rdparty/chromium/media/filters/audio_file_reader.cc
index bd73908..745c4c7 100644
--- a/src/3rdparty/chromium/media/filters/audio_file_reader.cc
+++ b/src/3rdparty/chromium/media/filters/audio_file_reader.cc
@@ -112,14 +112,15 @@ 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;
@@ -222,7 +223,7 @@ 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/src/3rdparty/chromium/media/filters/ffmpeg_aac_bitstream_converter.cc b/src/3rdparty/chromium/media/filters/ffmpeg_aac_bitstream_converter.cc
index 6f231c8..ca5e5fb 100644
--- a/src/3rdparty/chromium/media/filters/ffmpeg_aac_bitstream_converter.cc
+++ b/src/3rdparty/chromium/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/src/3rdparty/chromium/media/filters/ffmpeg_aac_bitstream_converter_unittest.cc b/src/3rdparty/chromium/media/filters/ffmpeg_aac_bitstream_converter_unittest.cc
index ac8bb13..3e4e3f6 100644
--- a/src/3rdparty/chromium/media/filters/ffmpeg_aac_bitstream_converter_unittest.cc
+++ b/src/3rdparty/chromium/media/filters/ffmpeg_aac_bitstream_converter_unittest.cc
@@ -29,7 +29,7 @@ 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/src/3rdparty/chromium/media/filters/ffmpeg_audio_decoder.cc b/src/3rdparty/chromium/media/filters/ffmpeg_audio_decoder.cc
index 72fac61..ab49fd5 100644
--- a/src/3rdparty/chromium/media/filters/ffmpeg_audio_decoder.cc
+++ b/src/3rdparty/chromium/media/filters/ffmpeg_audio_decoder.cc
@@ -27,7 +27,7 @@ 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
@@ -227,7 +227,7 @@ 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;
@@ -344,11 +344,11 @@ 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_ = kUninitialized;
     return false;
@@ -398,7 +398,7 @@ 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);
   }
@@ -431,7 +431,8 @@ 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/src/3rdparty/chromium/media/ffmpeg/ffmpeg_common.cc b/src/3rdparty/chromium/media/ffmpeg/ffmpeg_common.cc
index 0448cb5..89e9cf9 100644
--- a/src/3rdparty/chromium/media/ffmpeg/ffmpeg_common.cc
+++ b/src/3rdparty/chromium/media/ffmpeg/ffmpeg_common.cc
@@ -414,7 +414,9 @@ 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/src/3rdparty/chromium/media/ffmpeg/ffmpeg_regression_tests.cc b/src/3rdparty/chromium/media/ffmpeg/ffmpeg_regression_tests.cc
index ebd1bab..04d5ecc 100644
--- a/src/3rdparty/chromium/media/ffmpeg/ffmpeg_regression_tests.cc
+++ b/src/3rdparty/chromium/media/ffmpeg/ffmpeg_regression_tests.cc
@@ -86,16 +86,16 @@ FFMPEG_TEST_CASE(Cr47761, "crbug47761.ogg", PIPELINE_OK, PIPELINE_OK);
 FFMPEG_TEST_CASE(Cr50045, "crbug50045.mp4", PIPELINE_OK, PIPELINE_OK);
 FFMPEG_TEST_CASE(Cr62127, "crbug62127.webm", PIPELINE_OK, PIPELINE_OK);
 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,
@@ -150,7 +150,10 @@ 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,
@@ -371,8 +374,8 @@ 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/src/3rdparty/chromium/media/filters/audio_file_reader.cc b/src/3rdparty/chromium/media/filters/audio_file_reader.cc
index 745c4c7..2b3abba 100644
--- a/src/3rdparty/chromium/media/filters/audio_file_reader.cc
+++ b/src/3rdparty/chromium/media/filters/audio_file_reader.cc
@@ -242,10 +242,10 @@ 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/src/3rdparty/chromium/media/filters/ffmpeg_video_decoder.cc b/src/3rdparty/chromium/media/filters/ffmpeg_video_decoder.cc
index 7996606..a15aafc 100644
--- a/src/3rdparty/chromium/media/filters/ffmpeg_video_decoder.cc
+++ b/src/3rdparty/chromium/media/filters/ffmpeg_video_decoder.cc
@@ -86,7 +86,7 @@ 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();
 }
@@ -183,7 +183,6 @@ 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.
@@ -318,8 +317,10 @@ 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());
   }
 
   switch (decoding_loop_->DecodePacket(
@@ -358,8 +359,13 @@ 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;
@@ -385,8 +391,10 @@ 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 (decode_nalus_)
     codec_context_->flags2 |= AV_CODEC_FLAG2_CHUNKS;
diff --git a/src/3rdparty/chromium/media/filters/ffmpeg_video_decoder.h b/src/3rdparty/chromium/media/filters/ffmpeg_video_decoder.h
index f13ce41..ee2444b 100644
--- a/src/3rdparty/chromium/media/filters/ffmpeg_video_decoder.h
+++ b/src/3rdparty/chromium/media/filters/ffmpeg_video_decoder.h
@@ -8,6 +8,8 @@
 #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"
@@ -85,6 +87,20 @@ 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_;
 
   VideoFramePool frame_pool_;
diff --git a/src/3rdparty/chromium/third_party/webrtc/modules/video_coding/codecs/h264/h264_decoder_impl.cc b/src/3rdparty/chromium/third_party/webrtc/modules/video_coding/codecs/h264/h264_decoder_impl.cc
index d12fade..8abfbbf 100644
--- a/src/3rdparty/chromium/third_party/webrtc/modules/video_coding/codecs/h264/h264_decoder_impl.cc
+++ b/src/3rdparty/chromium/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());

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File qt5-webengine-ffmpeg7.patch of Package libqt5-qtwebengine (Revision f030ef36cde615da547e30ca6021e819)

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