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

talk/app/webrtc/webrtcsession.cc

 /*
  * libjingle
  * Copyright 2012 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 1154 matching lines @@
   ASSERT(signaling_thread()->IsCurrent());
   if (!voice_channel_) {
     LOG(LS_ERROR) << "SetAudioSend: No audio channel exists.";
     return;
   }
   if (!voice_channel_->SetLocalRenderer(ssrc, renderer)) {
     // SetRenderer() can fail if the ssrc does not match any send channel.
     LOG(LS_ERROR) << "SetAudioSend: ssrc is incorrect: " << ssrc;
     return;
   }
-  if (!voice_channel_->MuteStream(ssrc, !enable)) {
+  if (!voice_channel_->MuteStream(ssrc, !enable, &options)) {

[pthatcher1, 2015/09/04 23:21:16]

I like this.  We should take in one step further rename this to something like
Send(ssrc, active, options).  Because RtpSender::Send(...) is what it will be
called later.

[the sun, 2015/09/08 11:31:12]

I decided to take it even further and collapse SetLocalRenderer() into the same
function. Turns out to be a change big enough to warrant a separate CL:
https://codereview.chromium.org/1311533009/

     // Allow that MuteStream fail if |enable| is false but assert otherwise.
     // This in the normal case when the underlying media channel has already
     // been deleted.
     ASSERT(enable == false);
-    return;
   }
-  if (enable)
-    voice_channel_->SetChannelOptions(options);
 }
 
 void WebRtcSession::SetAudioPlayoutVolume(uint32 ssrc, double volume) {
   ASSERT(signaling_thread()->IsCurrent());
   ASSERT(volume >= 0 && volume <= 10);
   if (!voice_channel_) {
     LOG(LS_ERROR) << "SetAudioPlayoutVolume: No audio channel exists.";
     return;
   }
 
@@ skipping 37 matching lines @@
   }
 }
 
 void WebRtcSession::SetVideoSend(uint32 ssrc, bool enable,
                                  const cricket::VideoOptions* options) {
   ASSERT(signaling_thread()->IsCurrent());
   if (!video_channel_) {
     LOG(LS_WARNING) << "SetVideoSend: No video channel exists.";
     return;
   }
-  if (!video_channel_->MuteStream(ssrc, !enable)) {
+  if (!video_channel_->MuteStream(ssrc, !enable, options)) {
     // Allow that MuteStream fail if |enable| is false but assert otherwise.
     // This in the normal case when the underlying media channel has already
     // been deleted.
     ASSERT(enable == false);
-    return;
   }
-  if (enable && options)
-    video_channel_->SetChannelOptions(*options);
 }
 
 bool WebRtcSession::CanInsertDtmf(const std::string& track_id) {
   ASSERT(signaling_thread()->IsCurrent());
   if (!voice_channel_) {
     LOG(LS_ERROR) << "CanInsertDtmf: No audio channel exists.";
     return false;
   }
   uint32 send_ssrc = 0;
   // The Dtmf is negotiated per channel not ssrc, so we only check if the ssrc
@@ skipping 821 matching lines @@
 
   if (!srtp_cipher.empty()) {
     metrics_observer_->AddHistogramSample(srtp_name, srtp_cipher);
   }
   if (!ssl_cipher.empty()) {
     metrics_observer_->AddHistogramSample(ssl_name, ssl_cipher);
   }
 }
 
 }  // namespace webrtc