electron28-xdg/REVERT-roll-src-third_party-ffmpeg-m102.patch

From 6e554a30893150793c2638e3689cf208ffc8e375 Mon Sep 17 00:00:00 2001
From: Dale Curtis <dalecurtis@chromium.org>
Date: Sat, 2 Apr 2022 05:13:53 +0000
Subject: [PATCH] Roll src/third_party/ffmpeg/ 574c39cce..32b2d1d526 (1125
 commits)

https://chromium.googlesource.com/chromium/third_party/ffmpeg.git/+log/574c39cce323..32b2d1d526

Created with:
  roll-dep src/third_party/ffmpeg

Fixed: 1293918
Cq-Include-Trybots: luci.chromium.try:mac_chromium_asan_rel_ng,linux_chromium_asan_rel_ng,linux_chromium_chromeos_asan_rel_ng
Change-Id: I41945d0f963e3d1f65940067bac22f63b68e37d2
Reviewed-on: https://chromium-review.googlesource.com/c/chromium/src/+/3565647
Auto-Submit: Dale Curtis <dalecurtis@chromium.org>
Reviewed-by: Dan Sanders <sandersd@chromium.org>
Commit-Queue: Dale Curtis <dalecurtis@chromium.org>
Cr-Commit-Position: refs/heads/main@{#988253}
---
 .../clear_key_cdm/ffmpeg_cdm_audio_decoder.cc | 29 ++++++++++---------
 media/ffmpeg/ffmpeg_common.cc                 | 11 +++----
 media/filters/audio_file_reader.cc            |  9 +++---
 media/filters/audio_file_reader_unittest.cc   |  6 ++--
 .../filters/audio_video_metadata_extractor.cc | 11 +++++--
 .../filters/ffmpeg_aac_bitstream_converter.cc |  7 +++--
 ...ffmpeg_aac_bitstream_converter_unittest.cc |  2 +-
 media/filters/ffmpeg_audio_decoder.cc         | 13 +++++----
 8 files changed, 51 insertions(+), 37 deletions(-)

diff --git a/media/cdm/library_cdm/clear_key_cdm/ffmpeg_cdm_audio_decoder.cc b/media/cdm/library_cdm/clear_key_cdm/ffmpeg_cdm_audio_decoder.cc
index e4fc3f460e2..9b1ad9f7675 100644
--- a/media/cdm/library_cdm/clear_key_cdm/ffmpeg_cdm_audio_decoder.cc
+++ b/media/cdm/library_cdm/clear_key_cdm/ffmpeg_cdm_audio_decoder.cc
@@ -74,7 +74,7 @@ void CdmAudioDecoderConfigToAVCodecContext(
       codec_context->sample_fmt = AV_SAMPLE_FMT_NONE;
   }
 
-  codec_context->channels = config.channel_count;
+  codec_context->ch_layout.nb_channels = config.channel_count;
   codec_context->sample_rate = config.samples_per_second;
 
   if (config.extra_data) {
@@ -124,8 +124,8 @@ void CopySamples(cdm::AudioFormat cdm_format,
     case cdm::kAudioFormatPlanarS16:
     case cdm::kAudioFormatPlanarF32: {
       const int decoded_size_per_channel =
-          decoded_audio_size / av_frame.channels;
-      for (int i = 0; i < av_frame.channels; ++i) {
+          decoded_audio_size / av_frame.ch_layout.nb_channels;
+      for (int i = 0; i < av_frame.ch_layout.nb_channels; ++i) {
         memcpy(output_buffer, av_frame.extended_data[i],
                decoded_size_per_channel);
         output_buffer += decoded_size_per_channel;
@@ -185,13 +185,14 @@ bool FFmpegCdmAudioDecoder::Initialize(
   // Success!
   decoding_loop_ = std::make_unique<FFmpegDecodingLoop>(codec_context_.get());
   samples_per_second_ = config.samples_per_second;
-  bytes_per_frame_ = codec_context_->channels * config.bits_per_channel / 8;
+  bytes_per_frame_ =
+      codec_context_->ch_layout.nb_channels * config.bits_per_channel / 8;
   output_timestamp_helper_ =
       std::make_unique<AudioTimestampHelper>(config.samples_per_second);
   is_initialized_ = true;
 
   // Store initial values to guard against midstream configuration changes.
-  channels_ = codec_context_->channels;
+  channels_ = codec_context_->ch_layout.nb_channels;
   av_sample_format_ = codec_context_->sample_fmt;
 
   return true;
@@ -291,17 +292,19 @@ cdm::Status FFmpegCdmAudioDecoder::DecodeBuffer(
   for (auto& frame : audio_frames) {
     int decoded_audio_size = 0;
     if (frame->sample_rate != samples_per_second_ ||
-        frame->channels != channels_ || frame->format != av_sample_format_) {
+        frame->ch_layout.nb_channels != channels_ ||
+        frame->format != av_sample_format_) {
       DLOG(ERROR) << "Unsupported midstream configuration change!"
                   << " Sample Rate: " << frame->sample_rate << " vs "
-                  << samples_per_second_ << ", Channels: " << frame->channels
-                  << " vs " << channels_ << ", Sample Format: " << frame->format
-                  << " vs " << av_sample_format_;
+                  << samples_per_second_
+                  << ", Channels: " << frame->ch_layout.nb_channels << " vs "
+                  << channels_ << ", Sample Format: " << frame->format << " vs "
+                  << av_sample_format_;
       return cdm::kDecodeError;
     }
 
     decoded_audio_size = av_samples_get_buffer_size(
-        nullptr, codec_context_->channels, frame->nb_samples,
+        nullptr, codec_context_->ch_layout.nb_channels, frame->nb_samples,
         codec_context_->sample_fmt, 1);
     if (!decoded_audio_size)
       continue;
@@ -320,9 +323,9 @@ bool FFmpegCdmAudioDecoder::OnNewFrame(
     size_t* total_size,
     std::vector<std::unique_ptr<AVFrame, ScopedPtrAVFreeFrame>>* audio_frames,
     AVFrame* frame) {
-  *total_size += av_samples_get_buffer_size(nullptr, codec_context_->channels,
-                                            frame->nb_samples,
-                                            codec_context_->sample_fmt, 1);
+  *total_size += av_samples_get_buffer_size(
+      nullptr, codec_context_->ch_layout.nb_channels, frame->nb_samples,
+      codec_context_->sample_fmt, 1);
   audio_frames->emplace_back(av_frame_clone(frame));
   return true;
 }
diff --git a/media/ffmpeg/ffmpeg_common.cc b/media/ffmpeg/ffmpeg_common.cc
index 87ca8969626..76f03d6608e 100644
--- a/media/ffmpeg/ffmpeg_common.cc
+++ b/media/ffmpeg/ffmpeg_common.cc
@@ -345,10 +345,11 @@ bool AVCodecContextToAudioDecoderConfig(const AVCodecContext* codec_context,
       codec_context->sample_fmt, codec_context->codec_id);
 
   ChannelLayout channel_layout =
-      codec_context->channels > 8
+      codec_context->ch_layout.nb_channels > 8
           ? CHANNEL_LAYOUT_DISCRETE
-          : ChannelLayoutToChromeChannelLayout(codec_context->channel_layout,
-                                               codec_context->channels);
+          : ChannelLayoutToChromeChannelLayout(
+                codec_context->ch_layout.u.mask,
+                codec_context->ch_layout.nb_channels);
 
   int sample_rate = codec_context->sample_rate;
   switch (codec) {
@@ -401,7 +402,7 @@ bool AVCodecContextToAudioDecoderConfig(const AVCodecContext* codec_context,
                      extra_data, encryption_scheme, seek_preroll,
                      codec_context->delay);
   if (channel_layout == CHANNEL_LAYOUT_DISCRETE)
-    config->SetChannelsForDiscrete(codec_context->channels);
+    config->SetChannelsForDiscrete(codec_context->ch_layout.nb_channels);
 
 #if BUILDFLAG(ENABLE_PLATFORM_AC3_EAC3_AUDIO)
   // These are bitstream formats unknown to ffmpeg, so they don't have
@@ -470,7 +471,7 @@ void AudioDecoderConfigToAVCodecContext(const AudioDecoderConfig& config,
 
   // TODO(scherkus): should we set |channel_layout|? I'm not sure if FFmpeg uses
   // said information to decode.
-  codec_context->channels = config.channels();
+  codec_context->ch_layout.nb_channels = config.channels();
   codec_context->sample_rate = config.samples_per_second();
 
   if (config.extra_data().empty()) {
diff --git a/media/filters/audio_file_reader.cc b/media/filters/audio_file_reader.cc
index 5f257bdfaa6..e1be5aa9a5b 100644
--- a/media/filters/audio_file_reader.cc
+++ b/media/filters/audio_file_reader.cc
@@ -113,14 +113,15 @@ bool AudioFileReader::OpenDecoder() {
 
   // Verify the channel layout is supported by Chrome.  Acts as a sanity check
   // against invalid files.  See http://crbug.com/171962
-  if (ChannelLayoutToChromeChannelLayout(codec_context_->channel_layout,
-                                         codec_context_->channels) ==
+  if (ChannelLayoutToChromeChannelLayout(
+          codec_context_->ch_layout.u.mask,
+          codec_context_->ch_layout.nb_channels) ==
       CHANNEL_LAYOUT_UNSUPPORTED) {
     return false;
   }
 
   // Store initial values to guard against midstream configuration changes.
-  channels_ = codec_context_->channels;
+  channels_ = codec_context_->ch_layout.nb_channels;
   audio_codec_ = CodecIDToAudioCodec(codec_context_->codec_id);
   sample_rate_ = codec_context_->sample_rate;
   av_sample_format_ = codec_context_->sample_fmt;
@@ -223,7 +224,7 @@ bool AudioFileReader::OnNewFrame(
   if (frames_read < 0)
     return false;
 
-  const int channels = frame->channels;
+  const int channels = frame->ch_layout.nb_channels;
   if (frame->sample_rate != sample_rate_ || channels != channels_ ||
       frame->format != av_sample_format_) {
     DLOG(ERROR) << "Unsupported midstream configuration change!"
diff --git a/media/filters/audio_file_reader_unittest.cc b/media/filters/audio_file_reader_unittest.cc
index 2aba7927a31..1f45a50cace 100644
--- a/media/filters/audio_file_reader_unittest.cc
+++ b/media/filters/audio_file_reader_unittest.cc
@@ -121,11 +121,11 @@ class AudioFileReaderTest : public testing::Test {
     EXPECT_FALSE(reader_->Open());
   }
 
-  void RunTestFailingDecode(const char* fn) {
+  void RunTestFailingDecode(const char* fn, int expect_read = 0) {
     Initialize(fn);
     EXPECT_TRUE(reader_->Open());
     std::vector<std::unique_ptr<AudioBus>> decoded_audio_packets;
-    EXPECT_EQ(reader_->Read(&decoded_audio_packets), 0);
+    EXPECT_EQ(reader_->Read(&decoded_audio_packets), expect_read);
   }
 
   void RunTestPartialDecode(const char* fn) {
@@ -219,7 +219,7 @@ TEST_F(AudioFileReaderTest, AAC_ADTS) {
 }
 
 TEST_F(AudioFileReaderTest, MidStreamConfigChangesFail) {
-  RunTestFailingDecode("midstream_config_change.mp3");
+  RunTestFailingDecode("midstream_config_change.mp3", 42624);
 }
 #endif
 
diff --git a/media/filters/audio_video_metadata_extractor.cc b/media/filters/audio_video_metadata_extractor.cc
index 185819eb936..69ff508c221 100644
--- a/media/filters/audio_video_metadata_extractor.cc
+++ b/media/filters/audio_video_metadata_extractor.cc
@@ -113,6 +113,15 @@ bool AudioVideoMetadataExtractor::Extract(DataSource* source,
     if (!stream)
       continue;
 
+    void* display_matrix =
+        av_stream_get_side_data(stream, AV_PKT_DATA_DISPLAYMATRIX, nullptr);
+    if (display_matrix) {
+      rotation_ = VideoTransformation::FromFFmpegDisplayMatrix(
+                      static_cast<int32_t*>(display_matrix))
+                      .rotation;
+      info.tags["rotate"] = base::NumberToString(rotation_);
+    }
+
     // Extract dictionary from streams also. Needed for containers that attach
     // metadata to contained streams instead the container itself, like OGG.
     ExtractDictionary(stream->metadata, &info.tags);
@@ -255,8 +264,6 @@ void AudioVideoMetadataExtractor::ExtractDictionary(AVDictionary* metadata,
     if (raw_tags->find(tag->key) == raw_tags->end())
       (*raw_tags)[tag->key] = tag->value;
 
-    if (ExtractInt(tag, "rotate", &rotation_))
-      continue;
     if (ExtractString(tag, "album", &album_))
       continue;
     if (ExtractString(tag, "artist", &artist_))
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.";