Adding codecs to the RtpParameters returned by an RtpSender.

webrtc/media/engine/webrtcvideoengine2.cc

 /*
  *  Copyright (c) 2014 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 859 matching lines @@
             HasNack(send_codec_->codec), HasRemb(send_codec_->codec),
             HasTransportCc(send_codec_->codec),
             params.rtcp.reduced_size ? webrtc::RtcpMode::kReducedSize
                                      : webrtc::RtcpMode::kCompound);
       }
     }
   }
   send_params_ = params;
   return true;
 }
+
 webrtc::RtpParameters WebRtcVideoChannel2::GetRtpParameters(
     uint32_t ssrc) const {
   rtc::CritScope stream_lock(&stream_crit_);
   auto it = send_streams_.find(ssrc);
   if (it == send_streams_.end()) {
     LOG(LS_WARNING) << "Attempting to get RTP parameters for stream with ssrc "
                     << ssrc << " which doesn't exist.";
     return webrtc::RtpParameters();
   }
 
-  return it->second->GetRtpParameters();
+  webrtc::RtpParameters rtp_params = it->second->GetRtpParameters();
+  // Need to add the common list of codecs to the send stream-specific
+  // RTP parameters.
+  for (const VideoCodec& codec : send_params_.codecs) {
+    webrtc::RtpCodecParameters codec_params;
+    codec_params.payload_type = codec.id;
+    codec_params.mime_type = codec.name;
+    codec_params.clock_rate = codec.clockrate;
+    rtp_params.codecs.push_back(codec_params);

[skvlad, 2016/04/14 01:03:56]

Could this be moved into a function common for both audio and video? It seems to
only be using fields from the base Codec struct.

[Taylor Brandstetter, 2016/04/14 01:43:45]

Yeah, I can move this to codec.h/codec.cc.

+  }
+  return rtp_params;
 }
 
 bool WebRtcVideoChannel2::SetRtpParameters(
     uint32_t ssrc,
     const webrtc::RtpParameters& parameters) {
   TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetRtpParameters");
   rtc::CritScope stream_lock(&stream_crit_);
   auto it = send_streams_.find(ssrc);
   if (it == send_streams_.end()) {
     LOG(LS_ERROR) << "Attempting to set RTP parameters for stream with ssrc "
                   << ssrc << " which doesn't exist.";
     return false;
   }
 
+  // TODO(deadbeef): Handle setting parameters with a list of codecs in a
+  // different order (which should change the send codec).
   return it->second->SetRtpParameters(parameters);
 }
 
 bool WebRtcVideoChannel2::GetChangedRecvParameters(
     const VideoRecvParameters& params,
     ChangedRecvParameters* changed_params) const {
   if (!ValidateCodecFormats(params.codecs) ||
       !ValidateRtpExtensions(params.extensions)) {
     return false;
   }
@@ skipping 1662 matching lines @@
         rtx_mapping[video_codecs[i].codec.id] !=
             fec_settings.red_payload_type) {
       video_codecs[i].rtx_payload_type = rtx_mapping[video_codecs[i].codec.id];
     }
   }
 
   return video_codecs;
 }
 
 }  // namespace cricket