Adding support for RTCRtpEncodingParameters.active flag.

webrtc/media/engine/webrtcvideoengine2.h

Index: webrtc/media/engine/webrtcvideoengine2.h
diff --git a/webrtc/media/engine/webrtcvideoengine2.h b/webrtc/media/engine/webrtcvideoengine2.h
index 27eadb299ae69160ba7d9a41ca38c74463f4b691..5efce960bf51d18969e70f818d3e00cedf7918aa 100644
--- a/webrtc/media/engine/webrtcvideoengine2.h
+++ b/webrtc/media/engine/webrtcvideoengine2.h
@@ -254,10 +254,12 @@ class WebRtcVideoChannel2 : public VideoMediaChannel, public webrtc::Transport {
     void MuteStream(bool mute);
     bool DisconnectCapturer();
 
-    void Start();
-    void Stop();
+    void SetSend(bool send);
 
-    webrtc::RtpParameters rtp_parameters() const { return rtp_parameters_; }
+    webrtc::RtpParameters rtp_parameters() const {
+      rtc::CritScope cs(&lock_);

[pbos-webrtc, 2016/03/22 18:13:43]

This became non-trivial, move to .cc file.

[Taylor Brandstetter, 2016/03/22 18:32:43]

Done.

+      return rtp_parameters_;
+    }
 
     // Implements webrtc::LoadObserver.
     void OnLoadUpdate(Load load) override;
@@ -267,6 +269,10 @@ class WebRtcVideoChannel2 : public VideoMediaChannel, public webrtc::Transport {
     void FillBandwidthEstimationInfo(BandwidthEstimationInfo* bwe_info);
 
    private:
+    // Calls Start or Stop according to whether or not |sending_| is true,
+    // and whether or not the encoding in |rtp_parameters_| is active.
+    void UpdateSendState();

[pbos-webrtc, 2016/03/22 18:13:43]

put EXCLUSIVE_LOCKS_REQUIRED(lock_) before ; to have it only callable from under
a lock.

Also move this down to around line 340 or so where other methods are (internal
classes/structs are declared before methods)

[Taylor Brandstetter, 2016/03/22 18:32:43]

Somehow I didn't notice the other methods and that macro, oops.

+
     // Parameters needed to reconstruct the underlying stream.
     // webrtc::VideoSendStream doesn't support setting a lot of options on the
     // fly, so when those need to be changed we tear down and reconstruct with
@@ -372,7 +378,7 @@ class WebRtcVideoChannel2 : public VideoMediaChannel, public webrtc::Transport {
     // TODO(skvlad): Move ssrcs_ and ssrc_groups_ into rtp_parameters_.
     // TODO(skvlad): Combine parameters_ and rtp_parameters_ once we have only
     // one stream per MediaChannel.
-    webrtc::RtpParameters rtp_parameters_;
+    webrtc::RtpParameters rtp_parameters_ GUARDED_BY(lock_);
     bool pending_encoder_reconfiguration_ GUARDED_BY(lock_);
     VideoEncoderSettings encoder_settings_ GUARDED_BY(lock_);
     AllocatedEncoder allocated_encoder_ GUARDED_BY(lock_);
@@ -482,9 +488,6 @@ class WebRtcVideoChannel2 : public VideoMediaChannel, public webrtc::Transport {
                const webrtc::PacketOptions& options) override;
   bool SendRtcp(const uint8_t* data, size_t len) override;
 
-  void StartAllSendStreams();
-  void StopAllSendStreams();
-
   static std::vector<VideoCodecSettings> MapCodecs(
       const std::vector<VideoCodec>& codecs);
   std::vector<VideoCodecSettings> FilterSupportedCodecs(