From b66003ca79cd34f65ef964a5e3b4766bc97a5659 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Niels=20M=C3=B6ller?= <nisse@webrtc.org>
Date: Tue, 5 Feb 2019 22:41:20 +0100
Subject: [PATCH] Delete video source proxying in WebRtcVideoSendStream
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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}
---
 media/engine/webrtc_video_engine.cc | 40 +++--------------------------
 media/engine/webrtc_video_engine.h  | 14 ++--------
 2 files changed, 5 insertions(+), 49 deletions(-)

diff --git a/media/engine/webrtc_video_engine.cc b/media/engine/webrtc_video_engine.cc
index e0018726f1..9c97bd1567 100644
--- a/media/engine/webrtc_video_engine.cc
+++ b/media/engine/webrtc_video_engine.cc
@@ -1567,7 +1567,6 @@ 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) {
@@ -1663,7 +1662,7 @@ bool WebRtcVideoChannel::WebRtcVideoSendStream::SetVideoSend(
   // Switch to the new source.
   source_ = source;
   if (source && stream_) {
-    stream_->SetSource(this, GetDegradationPreference());
+    stream_->SetSource(source_, GetDegradationPreference());
   }
   return true;
 }
@@ -1843,7 +1842,7 @@ webrtc::RTCError WebRtcVideoChannel::WebRtcVideoSendStream::SetRtpParameters(
     UpdateSendState();
   }
   if (new_degradation_preference) {
-    stream_->SetSource(this, GetDegradationPreference());
+    stream_->SetSource(source_, GetDegradationPreference());
   }
   return webrtc::RTCError::OK();
 }
@@ -2024,39 +2023,6 @@ 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;
@@ -2179,7 +2145,7 @@ void WebRtcVideoChannel::WebRtcVideoSendStream::RecreateWebRtcStream() {
   parameters_.encoder_config.encoder_specific_settings = NULL;
 
   if (source_) {
-    stream_->SetSource(this, GetDegradationPreference());
+    stream_->SetSource(source_, GetDegradationPreference());
   }
 
   // Call stream_->Start() if necessary conditions are met.
diff --git a/media/engine/webrtc_video_engine.h b/media/engine/webrtc_video_engine.h
index a3d5a2f42b..464c78f43e 100644
--- a/media/engine/webrtc_video_engine.h
+++ b/media/engine/webrtc_video_engine.h
@@ -254,8 +254,7 @@ class WebRtcVideoChannel : public VideoMediaChannel, public webrtc::Transport {
       const std::vector<VideoCodecSettings>& codecs);
 
   // Wrapper for the sender part.
-  class WebRtcVideoSendStream
-      : public rtc::VideoSourceInterface<webrtc::VideoFrame> {
+  class WebRtcVideoSendStream {
    public:
     WebRtcVideoSendStream(
         webrtc::Call* call,
@@ -276,14 +275,6 @@ 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);
 
@@ -341,8 +332,7 @@ 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.