commit 6c02541abea929b01a4a2b656baa2d47fed2321a
author Christian Fremerey <chfremer@webrtc.org> Wed Feb 13 19:43:28 2019
committer Commit Bot <commit-bot@chromium.org> Wed Feb 13 22:42:43 2019
tree 359e3540145d99b84fa2a967d0c301ed27e080df
parent 3588394b26abf20bef02f886e0440483132a15a5

Revert "Delete video source proxying in WebRtcVideoSendStream"

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}

media/engine/webrtc_video_engine.cc

diff --git a/media/engine/webrtc_video_engine.cc b/media/engine/webrtc_video_engine.cc
index 9c97bd1..e001872 100644
--- a/media/engine/webrtc_video_engine.cc
+++ b/media/engine/webrtc_video_engine.cc
@@ -1567,6 +1567,7 @@
       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 @@
   // Switch to the new source.
   source_ = source;
   if (source && stream_) {
-    stream_->SetSource(source_, GetDegradationPreference());
+    stream_->SetSource(this, GetDegradationPreference());
   }
   return true;
 }
@@ -1842,7 +1843,7 @@
     UpdateSendState();
   }
   if (new_degradation_preference) {
-    stream_->SetSource(source_, GetDegradationPreference());
+    stream_->SetSource(this, GetDegradationPreference());
   }
   return webrtc::RTCError::OK();
 }
@@ -2023,6 +2024,39 @@
   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 @@
   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.

media/engine/webrtc_video_engine.h

diff --git a/media/engine/webrtc_video_engine.h b/media/engine/webrtc_video_engine.h
index 464c78f..a3d5a2f 100644
--- a/media/engine/webrtc_video_engine.h
+++ b/media/engine/webrtc_video_engine.h
@@ -254,7 +254,8 @@
       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 @@
     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 @@
         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.