Fix PayloadRouter::OnEncodedImage() to handle errors properly.

webrtc/video/payload_router.cc

 /*
  *  Copyright (c) 2015 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 109 matching lines @@
 
 EncodedImageCallback::Result PayloadRouter::OnEncodedImage(
     const EncodedImage& encoded_image,
     const CodecSpecificInfo* codec_specific_info,
     const RTPFragmentationHeader* fragmentation) {
   rtc::CritScope lock(&crit_);
   RTC_DCHECK(!rtp_modules_.empty());
   if (!active_)
     return Result(Result::ERROR_SEND_FAILED);
 
-  int stream_index = 0;
-
   RTPVideoHeader rtp_video_header;
   memset(&rtp_video_header, 0, sizeof(RTPVideoHeader));
   if (codec_specific_info)
     CopyCodecSpecific(codec_specific_info, &rtp_video_header);
   rtp_video_header.rotation = encoded_image.rotation_;
   rtp_video_header.playout_delay = encoded_image.playout_delay_;
-  stream_index = rtp_video_header.simulcastIdx;
 
+  int stream_index = rtp_video_header.simulcastIdx;
+  RTC_DCHECK_LT(stream_index, rtp_modules_.size());
   uint32_t frame_id;
-  int send_result = rtp_modules_[stream_index]->SendOutgoingData(
+  bool send_result = rtp_modules_[stream_index]->SendOutgoingData(
       encoded_image._frameType, payload_type_, encoded_image._timeStamp,
       encoded_image.capture_time_ms_, encoded_image._buffer,
       encoded_image._length, fragmentation, &rtp_video_header, &frame_id);
-
-  RTC_DCHECK_LT(rtp_video_header.simulcastIdx, rtp_modules_.size());
-  if (send_result < 0)
+  if (!send_result)
     return Result(Result::ERROR_SEND_FAILED);
 
   return Result(Result::OK, frame_id);
 }
 
 size_t PayloadRouter::MaxPayloadLength() const {
   size_t min_payload_length = DefaultMaxPayloadLength();
   rtc::CritScope lock(&crit_);
   for (size_t i = 0; i < rtp_modules_.size(); ++i) {
     size_t module_payload_length = rtp_modules_[i]->MaxDataPayloadLength();
     if (module_payload_length < min_payload_length)
       min_payload_length = module_payload_length;
   }
   return min_payload_length;
 }
 
 }  // namespace webrtc