Adding codecs to the RtpParameters returned by an RtpSender.

webrtc/media/engine/webrtcvoiceengine.cc

 /*
  *  Copyright (c) 2004 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 1400 matching lines @@
 webrtc::RtpParameters WebRtcVoiceMediaChannel::GetRtpParameters(
     uint32_t ssrc) const {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   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->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_) {
+    rtp_params.codecs.push_back(codec.ToCodecParameters());
+  }
+  return rtp_params;
 }
 
 bool WebRtcVoiceMediaChannel::SetRtpParameters(
     uint32_t ssrc,
     const webrtc::RtpParameters& parameters) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   if (!ValidateRtpParameters(parameters)) {
     return false;
   }
   auto it = send_streams_.find(ssrc);
   if (it == send_streams_.end()) {
     LOG(LS_WARNING) << "Attempting to set RTP parameters for stream with ssrc "
                     << ssrc << " which doesn't exist.";
     return false;
   }
 
   if (!SetChannelParameters(it->second->channel(), parameters)) {
     LOG(LS_WARNING) << "Failed to set RtpParameters.";
     return false;
   }
-  it->second->set_rtp_parameters(parameters);
+  // Codecs are handled at the WebRtcVoiceMediaChannel level.
+  webrtc::RtpParameters reduced_params = parameters;
+  reduced_params.codecs.clear();
+  it->second->set_rtp_parameters(reduced_params);
   return true;
 }
 
 bool WebRtcVoiceMediaChannel::ValidateRtpParameters(
     const webrtc::RtpParameters& rtp_parameters) {
   if (rtp_parameters.encodings.size() != 1) {
     LOG(LS_ERROR)
         << "Attempted to set RtpParameters without exactly one encoding";
     return false;
   }
@@ skipping 210 matching lines @@
   // and in that case reconfigure all receive streams.
   if (recv_transport_cc_enabled_ != send_codec_spec_.transport_cc_enabled) {
     LOG(LS_INFO) << "Recreate all the receive streams because the send "
                     "codec has changed.";
     recv_transport_cc_enabled_ = send_codec_spec_.transport_cc_enabled;
     for (auto& kv : recv_streams_) {
       kv.second->RecreateAudioReceiveStream(recv_transport_cc_enabled_);
     }
   }
 
+  send_codecs_ = codecs;
   return true;
 }
 
 // Apply current codec settings to a single voe::Channel used for sending.
 bool WebRtcVoiceMediaChannel::SetSendCodecs(
     int channel,
     const webrtc::RtpParameters& rtp_parameters) {
   // Disable VAD, FEC, and RED unless we know the other side wants them.
   engine()->voe()->codec()->SetVADStatus(channel, false);
   engine()->voe()->rtp()->SetNACKStatus(channel, false, 0);
@@ skipping 699 matching lines @@
       return false;
     }
   }
   return true;
 }
 
 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));
 }
 
 bool WebRtcVoiceMediaChannel::SetSendBitrate(int channel, int bps) {
   // Bitrate is auto by default.
   // TODO(bemasc): Fix this so that if SetMaxSendBandwidth(50) is followed by
   // SetMaxSendBandwith(0), the second call removes the previous limit.
   if (bps <= 0)
@@ skipping 160 matching lines @@
     }
   } else {
     LOG(LS_INFO) << "Stopping playout for channel #" << channel;
     engine()->voe()->base()->StopPlayout(channel);
   }
   return true;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE