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chromium-browser-stable-sla.../fix-ffmpeg.106.0.5219.0.patch
Sergey Zhemoytel eaa89e5f73 109.0.5414.119
2023-02-05 18:40:55 +00:00

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diff --git a/media/filters/ffmpeg_demuxer.h b/media/filters/ffmpeg_demuxer.h
index c147309..48a8f6a 100644
--- a/media/filters/ffmpeg_demuxer.h
+++ b/media/filters/ffmpeg_demuxer.h
@@ -151,6 +151,8 @@
base::TimeDelta start_time() const { return start_time_; }
void set_start_time(base::TimeDelta time) { start_time_ = time; }
+ int64_t first_dts() const { return first_dts_; }
+
private:
friend class FFmpegDemuxerTest;
@@ -208,6 +210,7 @@
bool fixup_chained_ogg_;
int num_discarded_packet_warnings_;
+ int64_t first_dts_;
int64_t last_packet_pos_;
int64_t last_packet_dts_;
};
--- a/media/filters/ffmpeg_demuxer.cc 2021-06-28 16:40:34.456861225 +0200
+++ b/media/filters/ffmpeg_demuxer.cc 2021-06-28 16:49:27.746115753 +0200
@@ -58,7 +58,7 @@
namespace {
-constexpr int64_t kInvalidPTSMarker = static_cast<int64_t>(0x8000000000000000);
+constexpr int64_t kRelativeTsBase = static_cast<int64_t>(0x7ffeffffffffffff);
void SetAVStreamDiscard(AVStream* stream, AVDiscard discard) {
DCHECK(stream);
@@ -96,7 +96,7 @@
sample_rate);
}
-static base::TimeDelta ExtractStartTime(AVStream* stream) {
+static base::TimeDelta ExtractStartTime(AVStream* stream, int64_t first_dts) {
// The default start time is zero.
base::TimeDelta start_time;
@@ -106,12 +106,12 @@
// Next try to use the first DTS value, for codecs where we know PTS == DTS
// (excludes all H26x codecs). The start time must be returned in PTS.
- if (av_stream_get_first_dts(stream) != kNoFFmpegTimestamp &&
+ if (first_dts != AV_NOPTS_VALUE &&
stream->codecpar->codec_id != AV_CODEC_ID_HEVC &&
stream->codecpar->codec_id != AV_CODEC_ID_H264 &&
stream->codecpar->codec_id != AV_CODEC_ID_MPEG4) {
const base::TimeDelta first_pts =
- ConvertFromTimeBase(stream->time_base, av_stream_get_first_dts(stream));
+ ConvertFromTimeBase(stream->time_base, first_dts);
if (first_pts < start_time)
start_time = first_pts;
}
@@ -280,6 +280,7 @@
fixup_negative_timestamps_(false),
fixup_chained_ogg_(false),
num_discarded_packet_warnings_(0),
+ first_dts_(AV_NOPTS_VALUE),
last_packet_pos_(AV_NOPTS_VALUE),
last_packet_dts_(AV_NOPTS_VALUE) {
DCHECK(demuxer_);
@@ -346,6 +347,11 @@
int64_t packet_dts =
packet->dts == AV_NOPTS_VALUE ? packet->pts : packet->dts;
+ if (first_dts_ == AV_NOPTS_VALUE && packet->dts != AV_NOPTS_VALUE &&
+ last_packet_dts_ != AV_NOPTS_VALUE) {
+ first_dts_ = packet->dts - (last_packet_dts_ + kRelativeTsBase);
+ }
+
// Chained ogg files have non-monotonically increasing position and time stamp
// values, which prevents us from using them to determine if a packet should
// be dropped. Since chained ogg is only allowed on single track audio only
@@ -1439,7 +1445,7 @@
max_duration = std::max(max_duration, streams_[i]->duration());
- base::TimeDelta start_time = ExtractStartTime(stream);
+ base::TimeDelta start_time = ExtractStartTime(stream, streams_[i]->first_dts());
// Note: This value is used for seeking, so we must take the true value and
// not the one possibly clamped to zero below.
@@ -1596,7 +1602,7 @@
for (const auto& stream : streams_) {
if (!stream || stream->IsEnabled() != enabled)
continue;
- if (av_stream_get_first_dts(stream->av_stream()) == kInvalidPTSMarker)
+ if (stream->first_dts() == AV_NOPTS_VALUE)
continue;
if (!lowest_start_time_stream ||
stream->start_time() < lowest_start_time_stream->start_time()) {
@@ -1617,7 +1623,7 @@
if (stream->type() != DemuxerStream::VIDEO)
continue;
- if (av_stream_get_first_dts(stream->av_stream()) == kInvalidPTSMarker)
+ if (stream->first_dts() == AV_NOPTS_VALUE)
continue;
if (!stream->IsEnabled())
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 e4fc3f46..9b1ad9f 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 @@
codec_context->sample_fmt = AV_SAMPLE_FMT_NONE;
}
- codec_context->ch_layout.nb_channels = config.channel_count;
+ codec_context->channels = config.channel_count;
codec_context->sample_rate = config.samples_per_second;
if (config.extra_data) {
@@ -124,8 +124,8 @@
case cdm::kAudioFormatPlanarS16:
case cdm::kAudioFormatPlanarF32: {
const int decoded_size_per_channel =
- decoded_audio_size / av_frame.ch_layout.nb_channels;
- for (int i = 0; i < av_frame.ch_layout.nb_channels; ++i) {
+ decoded_audio_size / av_frame.channels;
+ for (int i = 0; i < av_frame.channels; ++i) {
memcpy(output_buffer, av_frame.extended_data[i],
decoded_size_per_channel);
output_buffer += decoded_size_per_channel;
@@ -185,14 +185,13 @@
// Success!
decoding_loop_ = std::make_unique<FFmpegDecodingLoop>(codec_context_.get());
samples_per_second_ = config.samples_per_second;
- bytes_per_frame_ =
- codec_context_->ch_layout.nb_channels * config.bits_per_channel / 8;
+ bytes_per_frame_ = codec_context_->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_->ch_layout.nb_channels;
+ channels_ = codec_context_->channels;
av_sample_format_ = codec_context_->sample_fmt;
return true;
@@ -291,9 +291,8 @@
for (auto& frame : audio_frames) {
int decoded_audio_size = 0;
if (frame->sample_rate != samples_per_second_ ||
- frame->ch_layout.nb_channels != channels_ ||
- frame->format != av_sample_format_) {
- DLOG(ERROR) << "Unsupported midstream configuration change!"
+ frame->channels != channels_ || frame->format != av_sample_format_) {
+ DLOG(ERROR) << "Unsupported midstream configuration change!"
<< " Sample Rate: " << frame->sample_rate << " vs "
<< samples_per_second_
<< ", Channels: " << frame->ch_layout.nb_channels << " vs "
@@ -303,7 +302,7 @@
}
decoded_audio_size = av_samples_get_buffer_size(
- nullptr, codec_context_->ch_layout.nb_channels, frame->nb_samples,
+ nullptr, codec_context_->channels, frame->nb_samples,
codec_context_->sample_fmt, 1);
if (!decoded_audio_size)
continue;
@@ -323,7 +322,7 @@
std::vector<std::unique_ptr<AVFrame, ScopedPtrAVFreeFrame>>* audio_frames,
AVFrame* frame) {
*total_size += av_samples_get_buffer_size(
- nullptr, codec_context_->ch_layout.nb_channels, frame->nb_samples,
+ nullptr, codec_context_->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 87ca896..76f03d6 100644
--- a/media/ffmpeg/ffmpeg_common.cc
+++ b/media/ffmpeg/ffmpeg_common.cc
@@ -345,11 +345,10 @@
codec_context->sample_fmt, codec_context->codec_id);
ChannelLayout channel_layout =
- codec_context->ch_layout.nb_channels > 8
+ codec_context->channels > 8
? CHANNEL_LAYOUT_DISCRETE
- : ChannelLayoutToChromeChannelLayout(
- codec_context->ch_layout.u.mask,
- codec_context->ch_layout.nb_channels);
+ : ChannelLayoutToChromeChannelLayout(codec_context->channel_layout,
+ codec_context->channels);
int sample_rate = codec_context->sample_rate;
switch (codec) {
@@ -401,7 +402,7 @@
extra_data, encryption_scheme, seek_preroll,
codec_context->delay);
if (channel_layout == CHANNEL_LAYOUT_DISCRETE)
- config->SetChannelsForDiscrete(codec_context->ch_layout.nb_channels);
+ config->SetChannelsForDiscrete(codec_context->channels);
#if BUILDFLAG(ENABLE_PLATFORM_AC3_EAC3_AUDIO)
// These are bitstream formats unknown to ffmpeg, so they don't have
@@ -470,7 +471,7 @@
// TODO(scherkus): should we set |channel_layout|? I'm not sure if FFmpeg uses
// said information to decode.
- codec_context->ch_layout.nb_channels = config.channels();
+ codec_context->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 5f257bd..e1be5aa 100644
--- a/media/filters/audio_file_reader.cc
+++ b/media/filters/audio_file_reader.cc
@@ -113,15 +113,14 @@
// 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_->ch_layout.u.mask,
- codec_context_->ch_layout.nb_channels) ==
+ if (ChannelLayoutToChromeChannelLayout(codec_context_->channel_layout,
+ codec_context_->channels) ==
CHANNEL_LAYOUT_UNSUPPORTED) {
return false;
}
// Store initial values to guard against midstream configuration changes.
- channels_ = codec_context_->ch_layout.nb_channels;
+ channels_ = codec_context_->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 @@
if (frames_read < 0)
return false;
- const int channels = frame->ch_layout.nb_channels;
+ const int channels = frame->channels;
if (frame->sample_rate != sample_rate_ || channels != channels_ ||
frame->format != av_sample_format_) {
DLOG(ERROR) << "Unsupported midstream configuration change!"
@@ -243,10 +243,10 @@
// 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_ == AudioCodec::kAAC && frame->duration) {
+ if (audio_codec_ == AudioCodec::kAAC && frame->pkt_duration) {
const base::TimeDelta pkt_duration = ConvertFromTimeBase(
glue_->format_context()->streams[stream_index_]->time_base,
- frame->duration + std::min(static_cast<int64_t>(0), frame->pts));
+ frame->pkt_duration + std::min(static_cast<int64_t>(0), frame->pts));
const base::TimeDelta frame_duration =
base::Seconds(frames_read / static_cast<double>(sample_rate_));
diff --git a/media/filters/ffmpeg_aac_bitstream_converter.cc b/media/filters/ffmpeg_aac_bitstream_converter.cc
index 6f231c8..ca5e5fb 100644
--- a/media/filters/ffmpeg_aac_bitstream_converter.cc
+++ b/media/filters/ffmpeg_aac_bitstream_converter.cc
@@ -195,15 +195,14 @@
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_->ch_layout.nb_channels ||
+ channel_configuration_ != stream_codec_parameters_->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_->ch_layout.nb_channels,
+ stream_codec_parameters_->channels,
0, // originality
0, // home
0, // copyrighted_stream
@@ -214,7 +215,7 @@
codec_ = stream_codec_parameters_->codec_id;
audio_profile_ = stream_codec_parameters_->profile;
sample_rate_index_ = sample_rate_index;
- channel_configuration_ = stream_codec_parameters_->ch_layout.nb_channels;
+ channel_configuration_ = stream_codec_parameters_->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 1fd4c5c..f59bcd8f 100644
--- a/media/filters/ffmpeg_aac_bitstream_converter_unittest.cc
+++ b/media/filters/ffmpeg_aac_bitstream_converter_unittest.cc
@@ -34,7 +34,7 @@
memset(&test_parameters_, 0, sizeof(AVCodecParameters));
test_parameters_.codec_id = AV_CODEC_ID_AAC;
test_parameters_.profile = FF_PROFILE_AAC_MAIN;
- test_parameters_.ch_layout.nb_channels = 2;
+ test_parameters_.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 6a56c67..4615fdeb 100644
--- a/media/filters/ffmpeg_audio_decoder.cc
+++ b/media/filters/ffmpeg_audio_decoder.cc
@@ -28,7 +28,7 @@
// Return the number of channels from the data in |frame|.
static inline int DetermineChannels(AVFrame* frame) {
- return frame->ch_layout.nb_channels;
+ return frame->channels;
}
// Called by FFmpeg's allocation routine to allocate a buffer. Uses
@@ -231,7 +231,7 @@
// Translate unsupported into discrete layouts for discrete configurations;
// ffmpeg does not have a labeled discrete configuration internally.
ChannelLayout channel_layout = ChannelLayoutToChromeChannelLayout(
- codec_context_->ch_layout.u.mask, codec_context_->ch_layout.nb_channels);
+ codec_context_->channel_layout, codec_context_->channels);
if (channel_layout == CHANNEL_LAYOUT_UNSUPPORTED &&
config_.channel_layout() == CHANNEL_LAYOUT_DISCRETE) {
channel_layout = CHANNEL_LAYOUT_DISCRETE;
@@ -348,11 +348,11 @@
// Success!
av_sample_format_ = codec_context_->sample_fmt;
- if (codec_context_->ch_layout.nb_channels != config.channels()) {
+ if (codec_context_->channels != config.channels()) {
MEDIA_LOG(ERROR, media_log_)
<< "Audio configuration specified " << config.channels()
<< " channels, but FFmpeg thinks the file contains "
- << codec_context_->ch_layout.nb_channels << " channels";
+ << codec_context_->channels << " channels";
ReleaseFFmpegResources();
state_ = DecoderState::kUninitialized;
return false;
@@ -403,7 +403,7 @@
if (frame->nb_samples <= 0)
return AVERROR(EINVAL);
- if (s->ch_layout.nb_channels != channels) {
+ if (s->channels != channels) {
DLOG(ERROR) << "AVCodecContext and AVFrame disagree on channel count.";
return AVERROR(EINVAL);
}
@@ -436,8 +436,7 @@
ChannelLayout channel_layout =
config_.channel_layout() == CHANNEL_LAYOUT_DISCRETE
? CHANNEL_LAYOUT_DISCRETE
- : ChannelLayoutToChromeChannelLayout(s->ch_layout.u.mask,
- s->ch_layout.nb_channels);
+ : ChannelLayoutToChromeChannelLayout(s->channel_layout, s->channels);
if (channel_layout == CHANNEL_LAYOUT_UNSUPPORTED) {
DLOG(ERROR) << "Unsupported channel layout.";
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