New flag is_screencast in VideoOptions.

webrtc/video/video_send_stream.cc

 /*
  *  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 456 matching lines @@
                                  static_cast<unsigned int>(streams[i].max_qp));
   }
 
   // Set to zero to not update the bitrate controller from ViEEncoder, as
   // the bitrate controller is already set from Call.
   video_codec.startBitrate = 0;
 
   RTC_DCHECK_GT(streams[0].max_framerate, 0);
   video_codec.maxFramerate = streams[0].max_framerate;
 
-  if (!SetSendCodec(video_codec))
+  if (!SetSendCodec(video_codec)) {
+    LOG(LS_WARNING) << "(Re)configureVideoEncoder: SetSendCodec failed "
+                       "for config: "
+                    << config.ToString();
     return false;
-
+  }
   // Clear stats for disabled layers.
   for (size_t i = video_codec.numberOfSimulcastStreams;
        i < config_.rtp.ssrcs.size(); ++i) {
     stats_proxy_.OnInactiveSsrc(config_.rtp.ssrcs[i]);
   }
 
   stats_proxy_.SetContentType(config.content_type);
 
   RTC_DCHECK_GE(config.min_transmit_bitrate_bps, 0);
   vie_encoder_.SetMinTransmitBitrate(config.min_transmit_bitrate_bps / 1000);
@@ skipping 134 matching lines @@
   used_ssrcs.resize(static_cast<size_t>(video_codec.numberOfSimulcastStreams));
   vie_encoder_.SetSsrcs(used_ssrcs);
 
   // Restart the media flow
   vie_encoder_.Restart();
 
   return true;
 }
 }  // namespace internal
 }  // namespace webrtc