| author | Antonio Rojas
<arojas@archlinux.org> 2024-05-23 22:29:56 UTC |
| committer | Antonio Rojas
<arojas@archlinux.org> 2024-05-23 22:29:56 UTC |
| parent | 55a02804953a9035cdee7e6ff2e2dae0bf5c5fea |
| .SRCINFO | +7 | -1 |
| PKGBUILD | +13 | -3 |
| ffmpeg-7.patch | +270 | -0 |
| ffmpeg-7b.patch | +195 | -0 |
| no-h264.patch | +33 | -0 |
diff --git a/.SRCINFO b/.SRCINFO index f012abf..7267794 100644 --- a/.SRCINFO +++ b/.SRCINFO @@ -1,7 +1,7 @@ pkgbase = qt5-webengine pkgdesc = Provides support for web applications using the Chromium browser project pkgver = 5.15.17 - pkgrel = 1 + pkgrel = 2 url = https://www.qt.io arch = x86_64 groups = qt5 @@ -44,6 +44,9 @@ pkgbase = qt5-webengine source = qt5-webengine-ninja-1.12.patch source = python3.12-imp.patch source = python3.12-six.patch + source = ffmpeg-7.patch + source = ffmpeg-7b.patch + source = no-h264.patch sha256sums = c2488e6d6778cf9954fcc45e94acb762376581a99df7f889f13a93cefd356940 sha256sums = SKIP sha256sums = c50d3019626183e753c53a997dc8a55938847543aa3178d4c51f377be741c693 @@ -52,5 +55,8 @@ pkgbase = qt5-webengine sha256sums = 6672741b64d896dc555c8ee42ca2329c4f20d5f406095a69fe72da44b3a142f4 sha256sums = 1f4357708e985bb5aca61a7e5dc4c0c1285d2af00994bb49ff89ede78198e0d2 sha256sums = ac87ec55ee5cbcf2d520e1ea433d041c0bf754271a17f859edbb9976f192ce3f + sha256sums = d4d0d077169a71bac55b8455065fbd608d194dda4c06cd1cdf6b1a959fb01e65 + sha256sums = d852831642aec200bb7bcbe0b0d3c0cb8151dc4164f4fb20d380a84759fb0459 + sha256sums = b8f65c5bd5f0ce5ee297d3642c9e0ef58fbc525e304386d05cfa950b66cf206b pkgname = qt5-webengine diff --git a/PKGBUILD b/PKGBUILD index 2495ab5..b2892a2 100644 --- a/PKGBUILD +++ b/PKGBUILD @@ -4,7 +4,7 @@ pkgname=qt5-webengine _basever=5.15.13 pkgver=5.15.17 -pkgrel=1 +pkgrel=2 arch=('x86_64') url='https://www.qt.io' license=('LGPL3' 'LGPL2.1' 'BSD') @@ -22,7 +22,10 @@ source=(kde-$_pkgfqn::git+https://code.qt.io/qt/qtwebengine.git#tag=v${pkgver}-l qt5-webengine-icu-75.patch qt5-webengine-ninja-1.12.patch python3.12-imp.patch - python3.12-six.patch) + python3.12-six.patch + ffmpeg-7.patch + ffmpeg-7b.patch + no-h264.patch) sha256sums=('c2488e6d6778cf9954fcc45e94acb762376581a99df7f889f13a93cefd356940' 'SKIP' 'c50d3019626183e753c53a997dc8a55938847543aa3178d4c51f377be741c693' @@ -30,7 +33,10 @@ sha256sums=('c2488e6d6778cf9954fcc45e94acb762376581a99df7f889f13a93cefd356940' '7cac28ba784d24b4abf6414079548ada165343af507ecd8e23cbe7e4f63ae52f' '6672741b64d896dc555c8ee42ca2329c4f20d5f406095a69fe72da44b3a142f4' '1f4357708e985bb5aca61a7e5dc4c0c1285d2af00994bb49ff89ede78198e0d2' - 'ac87ec55ee5cbcf2d520e1ea433d041c0bf754271a17f859edbb9976f192ce3f') + 'ac87ec55ee5cbcf2d520e1ea433d041c0bf754271a17f859edbb9976f192ce3f' + 'd4d0d077169a71bac55b8455065fbd608d194dda4c06cd1cdf6b1a959fb01e65' + 'd852831642aec200bb7bcbe0b0d3c0cb8151dc4164f4fb20d380a84759fb0459' + 'b8f65c5bd5f0ce5ee297d3642c9e0ef58fbc525e304386d05cfa950b66cf206b') prepare() { mkdir -p build @@ -47,6 +53,10 @@ prepare() { patch -p2 -d src/3rdparty/chromium -i "$srcdir"/qt5-webengine-ninja-1.12.patch # Fix build with ninja 1.12 patch -p1 -d src/3rdparty/chromium -i "$srcdir"/python3.12-imp.patch # Fix build with python 3.12 - patch from Debian patch -p1 -d src/3rdparty/chromium -i "$srcdir"/python3.12-six.patch # Fix build with python 3.12 - patch from Debian +# Fix build with ffmpeg 7 - OpenSUSE and Chromium patches + patch -p1 -d src/3rdparty/chromium -i "$srcdir"/ffmpeg-7.patch + patch -p1 -d src/3rdparty/chromium -i "$srcdir"/ffmpeg-7b.patch + patch -p1 -i ../no-h264.patch } build() { diff --git a/ffmpeg-7.patch b/ffmpeg-7.patch new file mode 100644 index 0000000..7d96e47 --- /dev/null +++ b/ffmpeg-7.patch @@ -0,0 +1,270 @@ +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."; diff --git a/ffmpeg-7b.patch b/ffmpeg-7b.patch new file mode 100644 index 0000000..e4fcc3d --- /dev/null +++ b/ffmpeg-7b.patch @@ -0,0 +1,195 @@ +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_)); + diff --git a/no-h264.patch b/no-h264.patch new file mode 100644 index 0000000..68781bf --- /dev/null +++ b/no-h264.patch @@ -0,0 +1,33 @@ +From: Fabian Vogt <fabian@ritter-vogt.de> +Subject: Don't require open264 when proprietary_codecs are supported + +Amended on 2024-01-30: also disable h265 + +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 this flag to make webrtc::Mutex be implemented by absl::Mutex. + rtc_use_absl_mutex = false +diff --git a/src/core/config/common.pri b/src/core/config/common.pri +index fc46d55ce..27558a500 100644 +--- a/src/core/config/common.pri ++++ b/src/core/config/common.pri +@@ -26,9 +26,6 @@ qtConfig(webengine-webrtc) { + + qtConfig(webengine-proprietary-codecs) { + gn_args += proprietary_codecs=true ffmpeg_branding=\"Chrome\" +- qtConfig(webengine-webrtc) { +- gn_args += rtc_use_h264=true +- } + } else { + gn_args += proprietary_codecs=false + }