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

webrtc/video_receive_stream.h

 /*
  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 144 matching lines @@
       // restored to the last non-RTX packet payload type received.
       bool use_rtx_payload_mapping_on_restore = false;
 
       // RTP header extensions used for the received stream.
       std::vector<RtpExtension> extensions;
     } rtp;
 
     // 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
     // this many milliseconds, if possible, earlier than the ideal render time.
     // Only valid if 'renderer' is set.
     int render_delay_ms = 10;
 
     // If set, pass frames on to the renderer as soon as they are
     // available.
     bool disable_prerenderer_smoothing = false;
@@ skipping 31 matching lines @@
   // TODO(pbos): Add info on currently-received codec to Stats.
   virtual Stats GetStats() const = 0;
 
  protected:
   virtual ~VideoReceiveStream() {}
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_VIDEO_RECEIVE_STREAM_H_