Add logging statements to places where the frame might be dropped in WebRTC pipeline.

webrtc/video_receive_stream.h

Index: webrtc/video_receive_stream.h
diff --git a/webrtc/video_receive_stream.h b/webrtc/video_receive_stream.h
index 0adcf3fc0d4cd3f499ae562dbd91ffbcfb03a21b..f87b7a440e29d35c2ef18045e4e870a178e3db20 100644
--- a/webrtc/video_receive_stream.h
+++ b/webrtc/video_receive_stream.h
@@ -162,8 +162,7 @@ class VideoReceiveStream {
     // Transport for outgoing packets (RTCP).
     Transport* rtcp_send_transport = nullptr;
 
-    // VideoRenderer will be called for each decoded frame. 'nullptr' disables
-    // rendering of this stream.
+    // Must not be 'nullptr' when the stream is started.
     rtc::VideoSinkInterface<VideoFrame>* renderer = nullptr;
 
     // Expected delay needed by the renderer, i.e. the frame will be delivered