Adding codecs to the RtpParameters returned by an RtpSender.

webrtc/media/engine/webrtcvoiceengine.cc

Index: webrtc/media/engine/webrtcvoiceengine.cc
diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
index 21f12105f1db6745f8c70c6167e8210e023e305e..924c109230f7add1d0ad08005f15db94ea9de235 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -1418,7 +1418,18 @@ webrtc::RtpParameters WebRtcVoiceMediaChannel::GetRtpParameters(
     return webrtc::RtpParameters();
   }
 
-  return it->second->rtp_parameters();
+  webrtc::RtpParameters rtp_params = it->second->rtp_parameters();
+  // Need to add the common list of codecs to the send stream-specific
+  // RTP parameters.
+  for (const AudioCodec& codec : send_codecs_) {
+    webrtc::RtpCodecParameters codec_params;
+    codec_params.payload_type = codec.id;
+    codec_params.mime_type = codec.name;
+    codec_params.clock_rate = codec.clockrate;
+    codec_params.channels = codec.channels;
+    rtp_params.codecs.push_back(codec_params);
+  }
+  return rtp_params;
 }
 
 bool WebRtcVoiceMediaChannel::SetRtpParameters(
@@ -1670,6 +1681,7 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
     }
   }
 
+  send_codecs_ = codecs;
   return true;
 }
 
@@ -2389,6 +2401,8 @@ bool WebRtcVoiceMediaChannel::SetChannelParameters(
     int channel,
     const webrtc::RtpParameters& parameters) {
   RTC_CHECK_EQ(1UL, parameters.encodings.size());
+  // TODO(deadbeef): Handle setting parameters with a list of codecs in a
+  // different order (which should change the send codec).
   return SetSendBitrate(
       channel,
       MinPositive(send_bitrate_bps_, parameters.encodings[0].max_bitrate_bps));