mirror of
https://github.com/archlinuxarm/PKGBUILDs.git
synced 2024-11-08 22:45:43 +00:00
extra/qt6-webengine to 6.7.1-3
This commit is contained in:
parent
6f5174f922
commit
5c9b81fd6f
4 changed files with 584 additions and 3 deletions
|
@ -1,7 +1,7 @@
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pkgbase = qt6-webengine
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pkgdesc = Provides support for web applications using the Chromium browser project
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pkgver = 6.7.1
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pkgrel = 1
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pkgrel = 3
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url = https://www.qt.io
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arch = x86_64
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groups = qt6
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@ -68,8 +68,12 @@ pkgbase = qt6-webengine
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source = git+https://code.qt.io/qt/qtwebengine#tag=v6.7.1
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source = git+https://code.qt.io/qt/qtwebengine-chromium
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source = qtwebengine-6.7.0-ninja1.12.patch
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source = qtwebengine-ffmpeg-7.patch
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source = no-h264.patch
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sha256sums = 25cba11801528f31b1e043aa29395c695189ec145b993c633ca8e557b13f34f6
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sha256sums = SKIP
|
||||
sha256sums = c037cccc1d43bcd9d67045354ca48b405acec217149cb4b2bd3cfb7b5561cc33
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sha256sums = d31143fcb04acdb4d0a183eb0e1f37be66aad19454ebbbd326a8fa68ce6f0d35
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sha256sums = 5e41df98141a5a3c6ab1ef1f1a45ea4a27631cfa3fe8b0cdd524624424d74541
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pkgname = qt6-webengine
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|
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@ -10,7 +10,7 @@ highmem=1
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pkgname=qt6-webengine
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pkgver=6.7.1
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pkgrel=1
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pkgrel=3
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arch=(x86_64)
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url='https://www.qt.io'
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license=(GPL3 LGPL3 FDL custom)
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@ -78,11 +78,15 @@ _pkgfn=${pkgname/6-/}
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source=(git+https://code.qt.io/qt/$_pkgfn#tag=v$pkgver
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git+https://code.qt.io/qt/qtwebengine-chromium
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qtwebengine-6.7.0-ninja1.12.patch
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qtwebengine-ffmpeg-7.patch
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no-h264.patch
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0001-ARM-toolchain-fixes.patch
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0002-Run-blink-bindings-generation-single-threaded.patch)
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sha256sums=('25cba11801528f31b1e043aa29395c695189ec145b993c633ca8e557b13f34f6'
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'SKIP'
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'c037cccc1d43bcd9d67045354ca48b405acec217149cb4b2bd3cfb7b5561cc33'
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'd31143fcb04acdb4d0a183eb0e1f37be66aad19454ebbbd326a8fa68ce6f0d35'
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'5e41df98141a5a3c6ab1ef1f1a45ea4a27631cfa3fe8b0cdd524624424d74541'
|
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'772aeba76d16387e28ef629285965109a4bfaa02ca2dd466e9526fe2030773ec'
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'eff4ebf2e714185ef6e36e30c8d2158da6b41a46e3168d06472d35f6c9977aaf')
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@ -93,7 +97,10 @@ prepare() {
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git -c protocol.file.allow=always submodule update
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# Fix build with ninja 1.12 - Gentoo patch
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patch -Np1 -i ../qtwebengine-6.7.0-ninja1.12.patch
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patch -p1 -i ../qtwebengine-6.7.0-ninja1.12.patch
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# Fix build with ffmpeg 7 - OpenSUSE and Chromium patches
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patch -d src/3rdparty/chromium -p1 -i "$srcdir"/qtwebengine-ffmpeg-7.patch
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patch -p1 -i ../no-h264.patch
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if [[ $CARCH == "armv7h" ]]; then
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export MAKEFLAGS="-j1"
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17
extra/qt6-webengine/no-h264.patch
Normal file
17
extra/qt6-webengine/no-h264.patch
Normal file
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@ -0,0 +1,17 @@
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From: Fabian Vogt <fabian@ritter-vogt.de>
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Subject: Don't require open264 when proprietary_codecs are supported
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diff --git a/src/3rdparty/chromium/third_party/webrtc/webrtc.gni b/src/3rdparty/chromium/third_party/webrtc/webrtc.gni
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index 5a1c43c8888..d867f7e5330 100644
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--- a/src/3rdparty/chromium/third_party/webrtc/webrtc.gni
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+++ b/src/3rdparty/chromium/third_party/webrtc/webrtc.gni
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@@ -186,8 +186,7 @@ declare_args() {
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#
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# Enabling H264 when building with MSVC is currently not supported, see
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# bugs.webrtc.org/9213#c13 for more info.
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- rtc_use_h264 =
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- proprietary_codecs && !is_android && !is_ios && !(is_win && !is_clang)
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+ rtc_use_h264 = false
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# Enable to use H265
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rtc_use_h265 = proprietary_codecs
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553
extra/qt6-webengine/qtwebengine-ffmpeg-7.patch
Normal file
553
extra/qt6-webengine/qtwebengine-ffmpeg-7.patch
Normal file
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@ -0,0 +1,553 @@
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From 6e554a30893150793c2638e3689cf208ffc8e375 Mon Sep 17 00:00:00 2001
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From: Dale Curtis <dalecurtis@chromium.org>
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Date: Sat, 2 Apr 2022 05:13:53 +0000
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Subject: [PATCH] Roll src/third_party/ffmpeg/ 574c39cce..32b2d1d526 (1125
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commits)
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https://chromium.googlesource.com/chromium/third_party/ffmpeg.git/+log/574c39cce323..32b2d1d526
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Created with:
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roll-dep src/third_party/ffmpeg
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Fixed: 1293918
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Cq-Include-Trybots: luci.chromium.try:mac_chromium_asan_rel_ng,linux_chromium_asan_rel_ng,linux_chromium_chromeos_asan_rel_ng
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Change-Id: I41945d0f963e3d1f65940067bac22f63b68e37d2
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Reviewed-on: https://chromium-review.googlesource.com/c/chromium/src/+/3565647
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Auto-Submit: Dale Curtis <dalecurtis@chromium.org>
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Reviewed-by: Dan Sanders <sandersd@chromium.org>
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Commit-Queue: Dale Curtis <dalecurtis@chromium.org>
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Cr-Commit-Position: refs/heads/main@{#988253}
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---
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.../clear_key_cdm/ffmpeg_cdm_audio_decoder.cc | 29 ++++++++++---------
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media/ffmpeg/ffmpeg_common.cc | 11 +++----
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media/filters/audio_file_reader.cc | 9 +++---
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media/filters/audio_file_reader_unittest.cc | 6 ++--
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.../filters/audio_video_metadata_extractor.cc | 11 +++++--
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.../filters/ffmpeg_aac_bitstream_converter.cc | 7 +++--
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...ffmpeg_aac_bitstream_converter_unittest.cc | 2 +-
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media/filters/ffmpeg_audio_decoder.cc | 13 +++++----
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8 files changed, 51 insertions(+), 37 deletions(-)
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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
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index e4fc3f460e2..9b1ad9f7675 100644
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--- a/media/cdm/library_cdm/clear_key_cdm/ffmpeg_cdm_audio_decoder.cc
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+++ b/media/cdm/library_cdm/clear_key_cdm/ffmpeg_cdm_audio_decoder.cc
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@@ -74,7 +74,7 @@ void CdmAudioDecoderConfigToAVCodecContext(
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codec_context->sample_fmt = AV_SAMPLE_FMT_NONE;
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}
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- codec_context->channels = config.channel_count;
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+ codec_context->ch_layout.nb_channels = config.channel_count;
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codec_context->sample_rate = config.samples_per_second;
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if (config.extra_data) {
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@@ -124,8 +124,8 @@ void CopySamples(cdm::AudioFormat cdm_format,
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case cdm::kAudioFormatPlanarS16:
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case cdm::kAudioFormatPlanarF32: {
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const int decoded_size_per_channel =
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- decoded_audio_size / av_frame.channels;
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- for (int i = 0; i < av_frame.channels; ++i) {
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+ decoded_audio_size / av_frame.ch_layout.nb_channels;
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+ for (int i = 0; i < av_frame.ch_layout.nb_channels; ++i) {
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memcpy(output_buffer, av_frame.extended_data[i],
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decoded_size_per_channel);
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output_buffer += decoded_size_per_channel;
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@@ -185,13 +185,14 @@ bool FFmpegCdmAudioDecoder::Initialize(
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// Success!
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decoding_loop_ = std::make_unique<FFmpegDecodingLoop>(codec_context_.get());
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samples_per_second_ = config.samples_per_second;
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- bytes_per_frame_ = codec_context_->channels * config.bits_per_channel / 8;
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+ bytes_per_frame_ =
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+ codec_context_->ch_layout.nb_channels * config.bits_per_channel / 8;
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output_timestamp_helper_ =
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std::make_unique<AudioTimestampHelper>(config.samples_per_second);
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is_initialized_ = true;
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// Store initial values to guard against midstream configuration changes.
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- channels_ = codec_context_->channels;
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+ channels_ = codec_context_->ch_layout.nb_channels;
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av_sample_format_ = codec_context_->sample_fmt;
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return true;
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@@ -291,18 +292,19 @@ cdm::Status FFmpegCdmAudioDecoder::DecodeBuffer(
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for (auto& frame : audio_frames) {
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int decoded_audio_size = 0;
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if (frame->sample_rate != samples_per_second_ ||
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- frame->channels != channels_ || frame->format != av_sample_format_) {
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+ frame->ch_layout.nb_channels != channels_ ||
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+ frame->format != av_sample_format_) {
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DLOG(ERROR) << "Unsupported midstream configuration change!"
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<< " Sample Rate: " << frame->sample_rate << " vs "
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<< samples_per_second_
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<< ", Channels: " << frame->ch_layout.nb_channels << " vs "
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<< channels_ << ", Sample Format: " << frame->format << " vs "
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<< av_sample_format_;
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return cdm::kDecodeError;
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}
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decoded_audio_size = av_samples_get_buffer_size(
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- nullptr, codec_context_->channels, frame->nb_samples,
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+ nullptr, codec_context_->ch_layout.nb_channels, frame->nb_samples,
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codec_context_->sample_fmt, 1);
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if (!decoded_audio_size)
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continue;
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@@ -320,9 +323,9 @@ bool FFmpegCdmAudioDecoder::OnNewFrame(
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size_t* total_size,
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std::vector<std::unique_ptr<AVFrame, ScopedPtrAVFreeFrame>>* audio_frames,
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AVFrame* frame) {
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- *total_size += av_samples_get_buffer_size(
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- nullptr, codec_context_->channels, frame->nb_samples,
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- codec_context_->sample_fmt, 1);
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+ *total_size += av_samples_get_buffer_size(
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+ nullptr, codec_context_->ch_layout.nb_channels, frame->nb_samples,
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+ codec_context_->sample_fmt, 1);
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audio_frames->emplace_back(av_frame_clone(frame));
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return true;
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}
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diff --git a/media/ffmpeg/ffmpeg_common.cc b/media/ffmpeg/ffmpeg_common.cc
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index 87ca8969626..76f03d6608e 100644
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--- a/media/ffmpeg/ffmpeg_common.cc
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+++ b/media/ffmpeg/ffmpeg_common.cc
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@@ -345,10 +345,11 @@ bool AVCodecContextToAudioDecoderConfig(const AVCodecContext* codec_context,
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codec_context->sample_fmt, codec_context->codec_id);
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ChannelLayout channel_layout =
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- codec_context->channels > 8
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+ codec_context->ch_layout.nb_channels > 8
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? CHANNEL_LAYOUT_DISCRETE
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- : ChannelLayoutToChromeChannelLayout(codec_context->channel_layout,
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- codec_context->channels);
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+ : ChannelLayoutToChromeChannelLayout(
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+ codec_context->ch_layout.u.mask,
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+ codec_context->ch_layout.nb_channels);
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int sample_rate = codec_context->sample_rate;
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switch (codec) {
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@@ -401,7 +402,7 @@ bool AVCodecContextToAudioDecoderConfig(const AVCodecContext* codec_context,
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extra_data, encryption_scheme, seek_preroll,
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codec_context->delay);
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if (channel_layout == CHANNEL_LAYOUT_DISCRETE)
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- config->SetChannelsForDiscrete(codec_context->channels);
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+ config->SetChannelsForDiscrete(codec_context->ch_layout.nb_channels);
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#if BUILDFLAG(ENABLE_PLATFORM_AC3_EAC3_AUDIO)
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// These are bitstream formats unknown to ffmpeg, so they don't have
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@@ -470,7 +471,7 @@ void AudioDecoderConfigToAVCodecContext(const AudioDecoderConfig& config,
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// TODO(scherkus): should we set |channel_layout|? I'm not sure if FFmpeg uses
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// said information to decode.
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- codec_context->channels = config.channels();
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+ codec_context->ch_layout.nb_channels = config.channels();
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codec_context->sample_rate = config.samples_per_second();
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if (config.extra_data().empty()) {
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diff --git a/media/filters/audio_file_reader.cc b/media/filters/audio_file_reader.cc
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index 5f257bdfaa6..e1be5aa9a5b 100644
|
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--- a/media/filters/audio_file_reader.cc
|
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+++ b/media/filters/audio_file_reader.cc
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@@ -113,14 +113,15 @@ bool AudioFileReader::OpenDecoder() {
|
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|
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// Verify the channel layout is supported by Chrome. Acts as a sanity check
|
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// against invalid files. See http://crbug.com/171962
|
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- if (ChannelLayoutToChromeChannelLayout(codec_context_->channel_layout,
|
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- codec_context_->channels) ==
|
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+ if (ChannelLayoutToChromeChannelLayout(
|
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+ codec_context_->ch_layout.u.mask,
|
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+ codec_context_->ch_layout.nb_channels) ==
|
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CHANNEL_LAYOUT_UNSUPPORTED) {
|
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return false;
|
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}
|
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|
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// Store initial values to guard against midstream configuration changes.
|
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- channels_ = codec_context_->channels;
|
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+ channels_ = codec_context_->ch_layout.nb_channels;
|
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audio_codec_ = CodecIDToAudioCodec(codec_context_->codec_id);
|
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sample_rate_ = codec_context_->sample_rate;
|
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av_sample_format_ = codec_context_->sample_fmt;
|
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@@ -223,7 +224,7 @@ bool AudioFileReader::OnNewFrame(
|
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if (frames_read < 0)
|
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return false;
|
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|
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- const int channels = frame->channels;
|
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+ const int channels = frame->ch_layout.nb_channels;
|
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if (frame->sample_rate != sample_rate_ || channels != channels_ ||
|
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frame->format != av_sample_format_) {
|
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DLOG(ERROR) << "Unsupported midstream configuration change!"
|
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diff --git a/media/filters/ffmpeg_aac_bitstream_converter.cc b/media/filters/ffmpeg_aac_bitstream_converter.cc
|
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index 6f231c85729..ca5e5fb927d 100644
|
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--- a/media/filters/ffmpeg_aac_bitstream_converter.cc
|
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+++ b/media/filters/ffmpeg_aac_bitstream_converter.cc
|
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@@ -195,14 +195,15 @@ bool FFmpegAACBitstreamConverter::ConvertPacket(AVPacket* packet) {
|
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if (!header_generated_ || codec_ != stream_codec_parameters_->codec_id ||
|
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audio_profile_ != stream_codec_parameters_->profile ||
|
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sample_rate_index_ != sample_rate_index ||
|
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- channel_configuration_ != stream_codec_parameters_->channels ||
|
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+ channel_configuration_ !=
|
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+ stream_codec_parameters_->ch_layout.nb_channels ||
|
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frame_length_ != header_plus_packet_size) {
|
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header_generated_ =
|
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GenerateAdtsHeader(stream_codec_parameters_->codec_id,
|
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0, // layer
|
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stream_codec_parameters_->profile, sample_rate_index,
|
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0, // private stream
|
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- stream_codec_parameters_->channels,
|
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+ stream_codec_parameters_->ch_layout.nb_channels,
|
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0, // originality
|
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0, // home
|
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0, // copyrighted_stream
|
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@@ -214,7 +215,7 @@ bool FFmpegAACBitstreamConverter::ConvertPacket(AVPacket* packet) {
|
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codec_ = stream_codec_parameters_->codec_id;
|
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audio_profile_ = stream_codec_parameters_->profile;
|
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sample_rate_index_ = sample_rate_index;
|
||||
- channel_configuration_ = stream_codec_parameters_->channels;
|
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+ channel_configuration_ = stream_codec_parameters_->ch_layout.nb_channels;
|
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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) {
|
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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_));
|
||||
|
Loading…
Reference in a new issue