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Sync from SUSE:SLFO:Main qt6-webengine revision 287e31e8e65586cfe85e16608597309f

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Adrian Schröter 2024-09-11 10:27:11 +02:00
parent 1b4392d338
commit a4140e5e23
6 changed files with 635 additions and 14 deletions
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33
qt6-webengine.changes
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@ -1,3 +1,36 @@
-------------------------------------------------------------------
Wed Aug 7 12:39:11 UTC 2024 - Christophe Marin <christophe@krop.fr>
- Add patch to build qtwebengine with ffmpeg 7 (picked from Arch)
* qtwebengine-ffmpeg-7.patch
-------------------------------------------------------------------
Wed Jun 19 07:26:07 UTC 2024 - Christophe Marin <christophe@krop.fr>
- Update to 6.7.2:
* https://www.qt.io/blog/qt-6.7.2-released
-------------------------------------------------------------------
Tue May 21 08:31:53 UTC 2024 - Christophe Marin <christophe@krop.fr>
- Update to 6.7.1:
* https://www.qt.io/blog/qt-6.7.1-released
- Drop patch, merged upstream:
* Add-missing-dependencies.patch
-------------------------------------------------------------------
Wed May 15 19:16:46 UTC 2024 - Christoph G <foss@grueninger.de>
- Backport Ninja 1.12 compatibility patch (and adjust paths)
Add-missing-dependencies.patch from upstream
-------------------------------------------------------------------
Tue Apr 2 13:40:04 UTC 2024 - Christophe Marin <christophe@krop.fr>
- Update to 6.7.0:
* https://www.qt.io/blog/qt-6.7-released
- Update rtc-dont-use-h264.patch
-------------------------------------------------------------------
Tue Mar 26 14:27:18 UTC 2024 - Christophe Marin <christophe@krop.fr>

22
qt6-webengine.spec
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@ -1,7 +1,7 @@
#
# spec file
# spec file for package qt6-webengine
#
# Copyright (c) 2023 SUSE LLC
# Copyright (c) 2024 SUSE LLC
#
# All modifications and additions to the file contributed by third parties
# remain the property of their copyright owners, unless otherwise agreed
@ -16,8 +16,8 @@
#
%define real_version 6.6.3
%define short_version 6.6
%define real_version 6.7.2
%define short_version 6.7
%define tar_name qtwebengine-everywhere-src
%define tar_suffix %{nil}
#
@ -37,20 +37,21 @@
%bcond_without system_tiff
%endif
Name: qt6-webengine%{?pkg_suffix}
Version: 6.6.3
Version: 6.7.2
Release: 0
Summary: Web browser engine for Qt applications
License: GPL-2.0-only OR LGPL-3.0-only OR GPL-3.0-only
URL: https://www.qt.io
Source: https://download.qt.io/official_releases/qt/%{short_version}/%{real_version}%{tar_suffix}/submodules/%{tar_name}-%{real_version}%{tar_suffix}.tar.xz
Source0: https://download.qt.io/official_releases/qt/%{short_version}/%{real_version}%{tar_suffix}/submodules/%{tar_name}-%{real_version}%{tar_suffix}.tar.xz
Source99: qt6-webengine-rpmlintrc
# Patches 0-100 are upstream patches #
# Patches 100-200 are openSUSE and/or non-upstream(able) patches #
Patch100: rtc-dont-use-h264.patch
Patch101: qtwebengine-ffmpeg-7.patch
#
# Chromium/blink don't support PowerPC and zSystems and build fails on
# 32 bits archs (https://bugreports.qt.io/browse/QTBUG-102143)
ExclusiveArch: aarch64 x86_64 riscv64
ExclusiveArch: aarch64 x86_64 %x86_64 riscv64
BuildRequires: Mesa-KHR-devel
BuildRequires: bison
# Not pulled automatically on Leap
@ -297,6 +298,7 @@ Requires: libQt6WebEngineQuick6 = %{version}
Requires: cmake(Qt6Gui) = %{real_version}
Requires: cmake(Qt6Qml) = %{real_version}
Requires: cmake(Qt6Quick) = %{real_version}
Requires: cmake(Qt6WebChannelQuick) = %{real_version}
Requires: cmake(Qt6WebEngineCore) = %{real_version}
%description -n qt6-webenginequick-devel
@ -344,6 +346,11 @@ ABI or API guarantees.
%prep
%autosetup -p1 -n %{tar_name}-%{real_version}%{tar_suffix}
%if %{pkg_vcmp pkgconfig(libavcodec) <= 5}
# The ffmpeg 7 compatibility patch would break build with older ffmpeg
%patch -P101 -p1 -R
%endif
%build
%if %{no_flavor}
# Determine the right number of parallel processes based on the available memory
@ -406,6 +413,7 @@ rm -r %{buildroot}%{_qt6_cmakedir}/Qt6BuildInternals
%{_qt6_datadir}/resources/
%{_qt6_translationsdir}/qtwebengine_locales/
%{_qt6_libexecdir}/QtWebEngineProcess
%{_qt6_libexecdir}/webenginedriver
%{_qt6_pluginsdir}/designer/libqwebengineview.so
%files imports

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qtwebengine-everywhere-src-6.6.3.tar.xz (Stored with Git LFS)
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qtwebengine-everywhere-src-6.7.2.tar.xz (Stored with Git LFS) Normal file
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574
qtwebengine-ffmpeg-7.patch Normal file
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@ -0,0 +1,574 @@
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 c535d2b..62ddbc8 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
@@ -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;
@@ -291,7 +292,8 @@ 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_
@@ -302,7 +304,7 @@ cdm::Status FFmpegCdmAudioDecoder::DecodeBuffer(
}
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;
@@ -322,7 +324,7 @@ bool FFmpegCdmAudioDecoder::OnNewFrame(
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,
+ 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/src/3rdparty/chromium/media/ffmpeg/ffmpeg_common.cc b/src/3rdparty/chromium/media/ffmpeg/ffmpeg_common.cc
index 2665355..910f9ad 100644
--- a/src/3rdparty/chromium/media/ffmpeg/ffmpeg_common.cc
+++ b/src/3rdparty/chromium/media/ffmpeg/ffmpeg_common.cc
@@ -336,10 +336,11 @@ 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);
switch (codec) {
// For AC3/EAC3 we enable only demuxing, but not decoding, so FFmpeg does
@@ -391,7 +392,7 @@ 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
@@ -460,7 +461,7 @@ 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 777eabc..2b58dd7 100644
--- a/src/3rdparty/chromium/media/filters/audio_file_reader.cc
+++ b/src/3rdparty/chromium/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/src/3rdparty/chromium/media/filters/ffmpeg_aac_bitstream_converter.cc b/src/3rdparty/chromium/media/filters/ffmpeg_aac_bitstream_converter.cc
index 76b41aa..e26b6cd 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 3b46f7f..1897eb0 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
@@ -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/src/3rdparty/chromium/media/filters/ffmpeg_audio_decoder.cc b/src/3rdparty/chromium/media/filters/ffmpeg_audio_decoder.cc
index bf3ed00..d564ee9 100644
--- a/src/3rdparty/chromium/media/filters/ffmpeg_audio_decoder.cc
+++ b/src/3rdparty/chromium/media/filters/ffmpeg_audio_decoder.cc
@@ -29,7 +29,7 @@ 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
@@ -243,7 +243,7 @@ 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;
@@ -360,11 +360,11 @@ 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;
@@ -415,7 +415,7 @@ 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);
}
@@ -448,7 +448,8 @@ 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 910f9ad..8be165c 100644
--- a/src/3rdparty/chromium/media/ffmpeg/ffmpeg_common.cc
+++ b/src/3rdparty/chromium/media/ffmpeg/ffmpeg_common.cc
@@ -411,7 +411,9 @@ 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/src/3rdparty/chromium/media/ffmpeg/ffmpeg_regression_tests.cc b/src/3rdparty/chromium/media/ffmpeg/ffmpeg_regression_tests.cc
index 05dcb1c..866f446 100644
--- a/src/3rdparty/chromium/media/ffmpeg/ffmpeg_regression_tests.cc
+++ b/src/3rdparty/chromium/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/src/3rdparty/chromium/media/filters/audio_decoder_unittest.cc b/src/3rdparty/chromium/media/filters/audio_decoder_unittest.cc
index a7b2533..ba3c308 100644
--- a/src/3rdparty/chromium/media/filters/audio_decoder_unittest.cc
+++ b/src/3rdparty/chromium/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/src/3rdparty/chromium/media/filters/audio_file_reader.cc b/src/3rdparty/chromium/media/filters/audio_file_reader.cc
index 2b58dd7..9d37f32 100644
--- a/src/3rdparty/chromium/media/filters/audio_file_reader.cc
+++ b/src/3rdparty/chromium/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_));
diff --git a/src/3rdparty/chromium/media/filters/audio_file_reader_unittest.cc b/src/3rdparty/chromium/media/filters/audio_file_reader_unittest.cc
index a1c633d..5784fe1 100644
--- a/src/3rdparty/chromium/media/filters/audio_file_reader_unittest.cc
+++ b/src/3rdparty/chromium/media/filters/audio_file_reader_unittest.cc
@@ -61,15 +61,14 @@ 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
@@ -98,7 +97,8 @@ 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());
@@ -112,7 +112,7 @@ 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);
}
@@ -219,7 +219,7 @@ TEST_F(AudioFileReaderTest, AAC_ADTS) {
}
TEST_F(AudioFileReaderTest, MidStreamConfigChangesFail) {
- RunTestFailingDecode("midstream_config_change.mp3", 42624);
+ RunTestFailingDecode("midstream_config_change.mp3", 0);
}
#endif
@@ -229,7 +229,7 @@ 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/src/3rdparty/chromium/media/filters/ffmpeg_video_decoder.cc b/src/3rdparty/chromium/media/filters/ffmpeg_video_decoder.cc
index e62b2af..ab39796 100644
--- a/src/3rdparty/chromium/media/filters/ffmpeg_video_decoder.cc
+++ b/src/3rdparty/chromium/media/filters/ffmpeg_video_decoder.cc
@@ -125,7 +125,7 @@ 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_);
}
@@ -204,10 +204,6 @@ 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));
@@ -354,8 +350,10 @@ 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,
@@ -414,7 +412,12 @@ 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);
@@ -489,8 +492,10 @@ 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 60cb9d5..4fa8628 100644
--- a/src/3rdparty/chromium/media/filters/ffmpeg_video_decoder.h
+++ b/src/3rdparty/chromium/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/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 f67718c..fe42aef 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
@@ -229,7 +229,6 @@ 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;
// Create a VideoFrame object, to keep a reference to the buffer.
// TODO(nisse): The VideoFrame's timestamp and rotation info is not used.
@@ -377,8 +376,6 @@ 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.get());
@@ -395,10 +392,6 @@ int32_t H264DecoderImpl::Decode(const EncodedImage& input_image,
return WEBRTC_VIDEO_CODEC_ERROR;
}
- // We don't expect reordering. Decoded frame timestamp should match
- // the input one.
- RTC_DCHECK_EQ(av_frame_->reordered_opaque, frame_timestamp_us);
-
// TODO(sakal): Maybe it is possible to get QP directly from FFmpeg.
h264_bitstream_parser_.ParseBitstream(input_image);
absl::optional<int> qp = h264_bitstream_parser_.GetLastSliceQp();

12
rtc-dont-use-h264.patch
View File

@ -1,11 +1,13 @@
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/chromium/third_party/webrtc/webrtc.gni
index ca8acdbf259..36897a72aa8 100644
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
@@ -151,8 +151,7 @@ declare_args() {
@@ -186,11 +186,10 @@ declare_args() {
#
# Enabling H264 when building with MSVC is currently not supported, see
# bugs.webrtc.org/9213#c13 for more info.
@ -13,5 +15,9 @@ index ca8acdbf259..36897a72aa8 100644
- proprietary_codecs && !is_android && !is_ios && !(is_win && !is_clang)
+ rtc_use_h264 = false
# Enable to use H265
- rtc_use_h265 = proprietary_codecs
+ rtc_use_h265 = false
# Enable this flag to make webrtc::Mutex be implemented by absl::Mutex.
rtc_use_absl_mutex = false