automatic version update by autodist [release 6.7.2-1mamba;Fri Jun 21 2024]

qt6-qtwebengine-6.7.2-ffmpeg-7.0.1.patch

@@ -0,0 +1,553 @@
+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_ = std::make_unique<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_ =
+       std::make_unique<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;
+@@ -291,18 +292,19 @@ 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->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;
+@@ -320,9 +323,9 @@ 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, frame->nb_samples,
+-      codec_context_->sample_fmt, 1);
++  *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));
+   return true;
+ }
+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) {
++    : media_log_(media_log), timestamp_map_(128) {
+   DVLOG(1) << __func__;
+   DETACH_FROM_SEQUENCE(sequence_checker_);
+ }
+@@ -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(std::make_pair(timestamp_id, timestamp));
++    packet->opaque = reinterpret_cast<void*>(timestamp_id.GetUnsafeValue());
+   }
+   FFmpegDecodingLoop::DecodeStatus decode_status = decoding_loop_->DecodePacket(
+       packet, base::BindRepeating(&FFmpegVideoDecoder::OnNewFrame,
+@@ -423,7 +425,12 @@ bool FFmpegVideoDecoder::OnNewFrame(AVFrame* frame) {
+   }
+   gfx::Size natural_size = aspect_ratio.GetNaturalSize(visible_rect);
+ 
+-  const auto pts = base::Microseconds(frame->reordered_opaque);
++  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;
++  }
++  const auto pts = base::Microseconds(std::get<1>(*ts_lookup));
+   auto video_frame = VideoFrame::WrapExternalDataWithLayout(
+       opaque->layout, visible_rect, natural_size, opaque->data, opaque->size,
+       pts);
+@@ -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
+@@ -7,10 +7,12 @@
+ 
+ #include <memory>
+ 
++#include "base/containers/lru_cache.h"
+ #include "base/functional/callback.h"
+ #include "base/memory/raw_ptr.h"
+ #include "base/memory/scoped_refptr.h"
+ #include "base/sequence_checker.h"
++#include "base/types/id_type.h"
+ #include "media/base/supported_video_decoder_config.h"
+ #include "media/base/video_decoder.h"
+ #include "media/base/video_decoder_config.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 = base::IdType<int64_t, size_t, 0>;
++
++  TimestampId::Generator timestamp_id_generator_;
++  base::LRUCache<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,
+ 
+     // TODO(dalecurtis): Just use the profile from the codec context if ffmpeg
+     // ever starts supporting xHE-AAC.
+-    if (codec_context->profile == FF_PROFILE_UNKNOWN) {
++    constexpr uint8_t kXHEAAc = 41;
++    if (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/audio_decoder_unittest.cc b/media/filters/audio_decoder_unittest.cc
+index a31823cfe3b58..e43f408b79e5c 100644
+--- a/media/filters/audio_decoder_unittest.cc
++++ b/media/filters/audio_decoder_unittest.cc
+@@ -484,7 +484,7 @@ constexpr TestParams kXheAacTestParams[] = {
+      }},
+      0,
+      29400,
+-     CHANNEL_LAYOUT_MONO,
++     CHANNEL_LAYOUT_UNSUPPORTED,
+      AudioCodecProfile::kXHE_AAC},
+ #endif
+     {AudioCodec::kAAC,
+diff --git a/media/filters/audio_file_reader_unittest.cc b/media/filters/audio_file_reader_unittest.cc
+index c0cc568d63019..edf9470f2f8b3 100644
+--- a/media/filters/audio_file_reader_unittest.cc
++++ b/media/filters/audio_file_reader_unittest.cc
+@@ -62,15 +62,14 @@ class AudioFileReaderTest : public testing::Test {
+   // Verify packets are consistent across demuxer runs.  Reads the first few
+   // packets and then seeks back to the start timestamp and verifies that the
+   // hashes match on the packets just read.
+-  void VerifyPackets() {
+-    const int kReads = 3;
++  void VerifyPackets(int packet_reads) {
+     const int kTestPasses = 2;
+ 
+     AVPacket packet;
+     base::TimeDelta start_timestamp;
+     std::vector<std::string> packet_md5_hashes_;
+     for (int i = 0; i < kTestPasses; ++i) {
+-      for (int j = 0; j < kReads; ++j) {
++      for (int j = 0; j < packet_reads; ++j) {
+         ASSERT_TRUE(reader_->ReadPacketForTesting(&packet));
+ 
+         // On the first pass save the MD5 hash of each packet, on subsequent
+@@ -99,7 +98,8 @@ class AudioFileReaderTest : public testing::Test {
+                int sample_rate,
+                base::TimeDelta duration,
+                int frames,
+-               int expected_frames) {
++               int expected_frames,
++               int packet_reads = 3) {
+     Initialize(fn);
+     ASSERT_TRUE(reader_->Open());
+     EXPECT_EQ(channels, reader_->channels());
+@@ -113,7 +113,7 @@ class AudioFileReaderTest : public testing::Test {
+       EXPECT_EQ(reader_->HasKnownDuration(), false);
+     }
+     if (!packet_verification_disabled_)
+-      ASSERT_NO_FATAL_FAILURE(VerifyPackets());
++      ASSERT_NO_FATAL_FAILURE(VerifyPackets(packet_reads));
+     ReadAndVerify(hash, expected_frames);
+   }
+ 
+@@ -220,7 +220,7 @@ TEST_F(AudioFileReaderTest, AAC_ADTS) {
+ }
+ 
+ TEST_F(AudioFileReaderTest, MidStreamConfigChangesFail) {
+-  RunTestFailingDecode("midstream_config_change.mp3", 42624);
++  RunTestFailingDecode("midstream_config_change.mp3", 0);
+ }
+ #endif
+ 
+@@ -230,7 +230,7 @@ TEST_F(AudioFileReaderTest, VorbisInvalidChannelLayout) {
+ 
+ TEST_F(AudioFileReaderTest, WaveValidFourChannelLayout) {
+   RunTest("4ch.wav", "131.71,38.02,130.31,44.89,135.98,42.52,", 4, 44100,
+-          base::Microseconds(100001), 4411, 4410);
++          base::Microseconds(100001), 4411, 4410, /*packet_reads=*/2);
+ }
+ 
+ TEST_F(AudioFileReaderTest, ReadPartialMP3) {
+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
+@@ -213,10 +213,6 @@ int FFmpegVideoDecoder::GetVideoBuffer(struct AVCodecContext* codec_context,
+     frame->linesize[plane] = layout->planes()[plane].stride;
+   }
+ 
+-  // This seems unsafe, given threaded decoding.  However, `reordered_opaque` is
+-  // also going away upstream, so we need a whole new mechanism either way.
+-  frame->reordered_opaque = codec_context->reordered_opaque;
+-
+   // This will be freed by `ReleaseVideoBufferImpl`.
+   auto* opaque = new OpaqueData(fb_priv, frame_pool_, data, allocation_size,
+                                 std::move(*layout));
+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
+@@ -243,18 +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 BUILDFLAG(USE_SYSTEM_FFMPEG)
+-  if (audio_codec_ == AudioCodec::kAAC && frame->pkt_duration) {
+-#else
+   if (audio_codec_ == AudioCodec::kAAC && frame->duration) {
+-#endif  // BUILDFLAG(USE_SYSTEM_FFMPEG)
+     const base::TimeDelta pkt_duration = ConvertFromTimeBase(
+         glue_->format_context()->streams[stream_index_]->time_base,
+-#if BUILDFLAG(USE_SYSTEM_FFMPEG)
+-        frame->pkt_duration + std::min(static_cast<int64_t>(0), frame->pts));
+-#else
+         frame->duration + std::min(static_cast<int64_t>(0), frame->pts));
+-#endif  // BUILDFLAG(USE_SYSTEM_FFMPEG)
+     const base::TimeDelta frame_duration =
+         base::Seconds(frames_read / static_cast<double>(sample_rate_));
+ 

qt6-qtwebengine-6.7.2-no-h264.patch

@@ -0,0 +1,17 @@
+From: Fabian Vogt <fabian@ritter-vogt.de>
+Subject: Don't require open264 when proprietary_codecs are supported
+
+diff --git a/src/3rdparty/chromium/third_party/webrtc/webrtc.gni b/src/3rdparty/chromium/third_party/webrtc/webrtc.gni
+index 5a1c43c8888..d867f7e5330 100644
+--- a/src/3rdparty/chromium/third_party/webrtc/webrtc.gni
++++ b/src/3rdparty/chromium/third_party/webrtc/webrtc.gni
+@@ -186,8 +186,7 @@ declare_args() {
+   #
+   # Enabling H264 when building with MSVC is currently not supported, see
+   # bugs.webrtc.org/9213#c13 for more info.
+-  rtc_use_h264 =
+-      proprietary_codecs && !is_android && !is_ios && !(is_win && !is_clang)
++  rtc_use_h264 = false
+
+   # Enable to use H265
+   rtc_use_h265 = proprietary_codecs

qt6-qtwebengine.spec

@@ -1,6 +1,6 @@
 %define majver %(echo %version | cut -d. -f1-2)
 Name:          qt6-qtwebengine
-Version:       6.7.1
+Version:       6.7.2
 Release:       1mamba
 Summary:       Qt6 WebEngine component
 Group:         System/Libraries
@@ -11,6 +11,8 @@ URL:           https://www.qt.io/
 Source:        https://download.qt.io/official_releases/qt/%{majver}/%{version}/submodules/qtwebengine-everywhere-src-%{version}.tar.xz
 Patch0:        qt6-qtwebengine-6.7.0-ninja-1.12.0.patch
 Patch1:        qt6-qtwebengine-6.7.0-ninja-1.12.0-2.patch
+Patch2:        qt6-qtwebengine-6.7.2-ffmpeg-7.0.1.patch
+Patch3:        qt6-qtwebengine-6.7.2-no-h264.patch
 License:       GPL
 ## AUTOBUILDREQ-BEGIN
 BuildRequires: glibc-devel
@@ -26,9 +28,9 @@ BuildRequires: libXtst-devel
 BuildRequires: libabseil-cpp-devel
 BuildRequires: libalsa-devel
 BuildRequires: libatomic-devel
-BuildRequires: libavcodec-ffmpeg-devel
+BuildRequires: libavcodec-devel
-BuildRequires: libavformat-ffmpeg-devel
+BuildRequires: libavformat-devel
-BuildRequires: libavutil-ffmpeg-devel
+BuildRequires: libavutil-devel
 BuildRequires: libdbus-devel
 BuildRequires: libdrm-devel
 BuildRequires: libevent-devel
@@ -89,12 +91,17 @@ This package contains libraries and header files for developing applications tha
 # Don't clean build at the end
 %global __spec_rmbuild_cmd /bin/true
 
+%define _default_patch_fuzz 2
+
 %prep
-%setup -q -n qtwebengine-everywhere-src-%{version}
+%setup -q -n qtwebengine-everywhere-src-%{version} -D -T
-#-D -T
+:<< _EOF
-#:<< _EOF
 %patch 0 -p1
 %patch 1 -p1
+cd src/3rdparty/chromium
+%patch 2 -p1 -b .ffmpeg-7.0.1
+cd ../../..
+%patch 3 -p1 -b .no-h264
 
 %build
 #:<< _EOF
@@ -184,6 +191,9 @@ DESTDIR="%{buildroot}" cmake --install build
 %{_libdir}/pkgconfig/Qt6*.pc
 
 %changelog
+* Fri Jun 21 2024 Automatic Build System <autodist@openmamba.org> 6.7.2-1mamba
+- automatic version update by autodist
+
 * Mon Jun 03 2024 Automatic Build System <autodist@openmamba.org> 6.7.1-1mamba
 - automatic version update by autodist