Support multiple rtx codecs.

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 476 matching lines @@
     vie_channel_->SetSSRC(config_.rtp.rtx.ssrcs[i], kViEStreamTypeRtx,
                           static_cast<unsigned char>(i));
     RtpStateMap::iterator it = suspended_ssrcs_.find(ssrc);
     if (it != suspended_ssrcs_.end())
       vie_channel_->SetRtpStateForSsrc(ssrc, it->second);
   }
 
   RTC_DCHECK_GE(config_.rtp.rtx.payload_type, 0);
   vie_channel_->SetRtxSendPayloadType(config_.rtp.rtx.payload_type,
                                       config_.encoder_settings.payload_type);
+  if (config_.rtp.fec.red_payload_type != -1 &&
+      config_.rtp.fec.red_rtx_payload_type != -1) {
+    vie_channel_->SetRtxSendPayloadType(config_.rtp.fec.red_rtx_payload_type,
+                                        config_.rtp.fec.red_payload_type);
+  }
 }
 
 std::map<uint32_t, RtpState> VideoSendStream::GetRtpStates() const {
   std::map<uint32_t, RtpState> rtp_states;
   for (size_t i = 0; i < config_.rtp.ssrcs.size(); ++i) {
     uint32_t ssrc = config_.rtp.ssrcs[i];
     rtp_states[ssrc] = vie_channel_->GetRtpStateForSsrc(ssrc);
   }
 
   for (size_t i = 0; i < config_.rtp.rtx.ssrcs.size(); ++i) {
@@ skipping 68 matching lines @@
   vie_encoder_->SetSsrcs(used_ssrcs);
 
   // Restart the media flow
   vie_encoder_->Restart();
 
   return true;
 }
 
 }  // namespace internal
 }  // namespace webrtc