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Revert "Delete video source proxying in WebRtcVideoSendStream"

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This reverts commit b66003ca79cd34f65ef964a5e3b4766bc97a5659.

Reason for revert: Causes bot failures in Chromium, see https://chromium-review.googlesource.com/c/chromium/src/+/1470391

Original change's description:
> Delete video source proxying in WebRtcVideoSendStream
>
> Bug: webrtc:10147
> Change-Id: Ib9f399e79d99f7d8db53fa38ef4b92986913ac26
> Reviewed-on: https://webrtc-review.googlesource.com/c/121569
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26633}

TBR=nisse@webrtc.org,sprang@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:10147
No-Try: True
Change-Id: I80395333d2be8fd3329c0bcdd6ed33d994a01ae3
Reviewed-on: https://webrtc-review.googlesource.com/c/122940
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Christian Fremerey <chfremer@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26672}
This commit is contained in:
Christian Fremerey
2019-02-13 19:43:28 +00:00
committed by Commit Bot
parent 3588394b26
commit 6c02541abe
2 changed files with 49 additions and 5 deletions
Download Patch File Download Diff File Expand all files Collapse all files

40
media/engine/webrtc_video_engine.cc
View File

@ -1567,6 +1567,7 @@ WebRtcVideoChannel::WebRtcVideoSendStream::WebRtcVideoSendStream(
enable_cpu_overuse_detection_(enable_cpu_overuse_detection),
source_(nullptr),
stream_(nullptr),
encoder_sink_(nullptr),
parameters_(std::move(config), options, max_bitrate_bps, codec_settings),
rtp_parameters_(CreateRtpParametersWithEncodings(sp)),
sending_(false) {
@ -1662,7 +1663,7 @@ bool WebRtcVideoChannel::WebRtcVideoSendStream::SetVideoSend(
// Switch to the new source.
source_ = source;
if (source && stream_) {
stream_->SetSource(source_, GetDegradationPreference());
stream_->SetSource(this, GetDegradationPreference());
}
return true;
}
@ -1842,7 +1843,7 @@ webrtc::RTCError WebRtcVideoChannel::WebRtcVideoSendStream::SetRtpParameters(
UpdateSendState();
}
if (new_degradation_preference) {
stream_->SetSource(source_, GetDegradationPreference());
stream_->SetSource(this, GetDegradationPreference());
}
return webrtc::RTCError::OK();
}
@ -2023,6 +2024,39 @@ void WebRtcVideoChannel::WebRtcVideoSendStream::SetSend(bool send) {
UpdateSendState();
}
void WebRtcVideoChannel::WebRtcVideoSendStream::RemoveSink(
rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) {
RTC_DCHECK_RUN_ON(&thread_checker_);
RTC_DCHECK(encoder_sink_ == sink);
encoder_sink_ = nullptr;
source_->RemoveSink(sink);
}
void WebRtcVideoChannel::WebRtcVideoSendStream::AddOrUpdateSink(
rtc::VideoSinkInterface<webrtc::VideoFrame>* sink,
const rtc::VideoSinkWants& wants) {
if (worker_thread_ == rtc::Thread::Current()) {
// AddOrUpdateSink is called on |worker_thread_| if this is the first
// registration of |sink|.
RTC_DCHECK_RUN_ON(&thread_checker_);
encoder_sink_ = sink;
source_->AddOrUpdateSink(encoder_sink_, wants);
} else {
// Subsequent calls to AddOrUpdateSink will happen on the encoder task
// queue.
invoker_.AsyncInvoke<void>(
RTC_FROM_HERE, worker_thread_, [this, sink, wants] {
RTC_DCHECK_RUN_ON(&thread_checker_);
// |sink| may be invalidated after this task was posted since
// RemoveSink is called on the worker thread.
bool encoder_sink_valid = (sink == encoder_sink_);
if (source_ && encoder_sink_valid) {
source_->AddOrUpdateSink(encoder_sink_, wants);
}
});
}
}
VideoSenderInfo WebRtcVideoChannel::WebRtcVideoSendStream::GetVideoSenderInfo(
bool log_stats) {
VideoSenderInfo info;
@ -2145,7 +2179,7 @@ void WebRtcVideoChannel::WebRtcVideoSendStream::RecreateWebRtcStream() {
parameters_.encoder_config.encoder_specific_settings = NULL;
if (source_) {
stream_->SetSource(source_, GetDegradationPreference());
stream_->SetSource(this, GetDegradationPreference());
}
// Call stream_->Start() if necessary conditions are met.

14
media/engine/webrtc_video_engine.h
View File

@ -254,7 +254,8 @@ class WebRtcVideoChannel : public VideoMediaChannel, public webrtc::Transport {
const std::vector<VideoCodecSettings>& codecs);
// Wrapper for the sender part.
class WebRtcVideoSendStream {
class WebRtcVideoSendStream
: public rtc::VideoSourceInterface<webrtc::VideoFrame> {
public:
WebRtcVideoSendStream(
webrtc::Call* call,
@ -275,6 +276,14 @@ class WebRtcVideoChannel : public VideoMediaChannel, public webrtc::Transport {
void SetFrameEncryptor(
rtc::scoped_refptr<webrtc::FrameEncryptorInterface> frame_encryptor);
// Implements rtc::VideoSourceInterface<webrtc::VideoFrame>.
// WebRtcVideoSendStream acts as a source to the webrtc::VideoSendStream
// in |stream_|. This is done to proxy VideoSinkWants from the encoder to
// the worker thread.
void AddOrUpdateSink(rtc::VideoSinkInterface<webrtc::VideoFrame>* sink,
const rtc::VideoSinkWants& wants) override;
void RemoveSink(rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) override;
bool SetVideoSend(const VideoOptions* options,
rtc::VideoSourceInterface<webrtc::VideoFrame>* source);
@ -332,7 +341,8 @@ class WebRtcVideoChannel : public VideoMediaChannel, public webrtc::Transport {
RTC_GUARDED_BY(&thread_checker_);
webrtc::VideoSendStream* stream_ RTC_GUARDED_BY(&thread_checker_);
rtc::VideoSinkInterface<webrtc::VideoFrame>* encoder_sink_
RTC_GUARDED_BY(&thread_checker_);
// Contains settings that are the same for all streams in the MediaChannel,
// such as codecs, header extensions, and the global bitrate limit for the
// entire channel.