Full impl of NnChannel::SetSendParameters and NnChannel::SetRecvParameters

talk/media/webrtc/webrtcvideoengine2.cc

 /*
  * libjingle
  * Copyright 2014 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 593 matching lines @@
     const VideoOptions& options,
     VoiceMediaChannel* voice_channel) {
   DCHECK(initialized_);
   LOG(LS_INFO) << "CreateChannel: "
                << (voice_channel != NULL ? "With" : "Without")
                << " voice channel. Options: " << options.ToString();
   WebRtcVideoChannel2* channel =
       new WebRtcVideoChannel2(call_factory_, voice_engine_,
           static_cast<WebRtcVoiceMediaChannel*>(voice_channel), options,
           external_encoder_factory_, external_decoder_factory_);
-  if (!channel->Init()) {
+  if (!channel->Init(video_codecs_)) {
     delete channel;
     return NULL;
   }
-  channel->SetRecvCodecs(video_codecs_);
   return channel;
 }
 
 const std::vector<VideoCodec>& WebRtcVideoEngine2::codecs() const {
   return video_codecs_;
 }
 
 const std::vector<RtpHeaderExtension>&
 WebRtcVideoEngine2::rtp_header_extensions() const {
   return rtp_header_extensions_;
@@ skipping 185 matching lines @@
 }
 
 WebRtcVideoChannel2::~WebRtcVideoChannel2() {
   DetachVoiceChannel();
   for (auto& kv : send_streams_)
     delete kv.second;
   for (auto& kv : receive_streams_)
     delete kv.second;
 }
 
-bool WebRtcVideoChannel2::Init() { return true; }
+bool WebRtcVideoChannel2::Init(const std::vector<VideoCodec>& recv_codecs) {
+  SetRecvCodecs(recv_codecs);
+  return true;
+}
 
 void WebRtcVideoChannel2::DetachVoiceChannel() {
   DCHECK(thread_checker_.CalledOnValidThread());
   if (voice_channel_) {
     voice_channel_->SetCall(nullptr);
     voice_channel_ = nullptr;
   }
 }
 
 bool WebRtcVideoChannel2::CodecIsExternallySupported(
@@ skipping 658 matching lines @@
       webrtc::PacketReceiver::DELIVERY_OK) {
     LOG(LS_WARNING) << "Failed to deliver RTCP packet.";
   }
 }
 
 void WebRtcVideoChannel2::OnReadyToSend(bool ready) {
   LOG(LS_VERBOSE) << "OnReadyToSend: " << (ready ? "Ready." : "Not ready.");
   call_->SignalNetworkState(ready ? webrtc::kNetworkUp : webrtc::kNetworkDown);
 }
 
-bool WebRtcVideoChannel2::MuteStream(uint32 ssrc, bool mute) {
+bool WebRtcVideoChannel2::MuteStream(uint32 ssrc, bool mute,
+                                     const VideoOptions* options) {
   LOG(LS_VERBOSE) << "MuteStream: " << ssrc << " -> "
                   << (mute ? "mute" : "unmute");
   DCHECK(ssrc != 0);
   rtc::CritScope stream_lock(&stream_crit_);
   if (send_streams_.find(ssrc) == send_streams_.end()) {
     LOG(LS_ERROR) << "No sending stream on ssrc " << ssrc;
     return false;
   }
 
   send_streams_[ssrc]->MuteStream(mute);
-  return true;
+
+  if (!mute && options) {
+    return SetOptions(*options);
+  } else {
+    return true;
+  }
 }
 
 bool WebRtcVideoChannel2::SetRecvRtpHeaderExtensions(
     const std::vector<RtpHeaderExtension>& extensions) {
   TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetRecvRtpHeaderExtensions");
   LOG(LS_INFO) << "SetRecvRtpHeaderExtensions: "
                << RtpExtensionsToString(extensions);
   if (!ValidateRtpHeaderExtensionIds(extensions))
     return false;
 
@@ skipping 1224 matching lines @@
       video_codecs[i].rtx_payload_type = rtx_mapping[video_codecs[i].codec.id];
     }
   }
 
   return video_codecs;
 }
 
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VIDEO