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

talk/media/webrtc/webrtcvoiceengine.cc

 /*
  * libjingle
  * Copyright 2004 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 568 matching lines @@
   return AUDIO_SEND | AUDIO_RECV;
 }
 
 VoiceMediaChannel* WebRtcVoiceEngine::CreateChannel(
     const AudioOptions& options) {
   WebRtcVoiceMediaChannel* ch = new WebRtcVoiceMediaChannel(this);
   if (!ch->valid()) {
     delete ch;
     return nullptr;
   }
-  if (!ch->SetOptions(options)) {
-    LOG(LS_WARNING) << "Failed to set options while creating channel.";
-  }
+  // TODO(solenberg): !!!!! SetParameters with only options?
+  // if (!ch->SetOptions(options)) {
+  //   LOG(LS_WARNING) << "Failed to set options while creating channel.";
+  // }

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

Same here: they just need to be remembered for later.  I'm guessing it would
make sense to pass them into the WebRtcVoiceMediaChannel constructor.  Either
that or you'll need to go refactor WebRtcSession to keep them around and then
pass them in along with SetRemote/SetLocalDescription.  I'm guessing the first
option will be easier.

[the sun, 2015/09/11 15:03:03]

Done.

   return ch;
 }
 
 bool WebRtcVoiceEngine::SetOptions(const AudioOptions& options) {
   if (!ApplyOptions(options)) {
     return false;
   }
   options_ = options;
   return true;
 }
@@ skipping 2456 matching lines @@
     // Make sure not sending the same packet to default channel more than once.
     if (IsDefaultChannel(ch.second->channel()) &&
         has_sent_to_default_channel)
       continue;
 
     engine()->voe()->network()->ReceivedRTCPPacket(
         ch.second->channel(), packet->data(), packet->size());
   }
 }
 
-bool WebRtcVoiceMediaChannel::MuteStream(uint32 ssrc, bool muted) {
+bool WebRtcVoiceMediaChannel::MuteStream(uint32 ssrc, bool mute,
+                                         const AudioOptions* options) {
   int channel = (ssrc == 0) ? voe_channel() : GetSendChannelNum(ssrc);
   if (channel == -1) {
     LOG(LS_WARNING) << "The specified ssrc " << ssrc << " is not in use.";
     return false;
   }
-  if (engine()->voe()->volume()->SetInputMute(channel, muted) == -1) {
-    LOG_RTCERR2(SetInputMute, channel, muted);
+  if (engine()->voe()->volume()->SetInputMute(channel, mute) == -1) {
+    LOG_RTCERR2(SetInputMute, channel, mute);
     return false;
   }
   // We set the AGC to mute state only when all the channels are muted.
   // This implementation is not ideal, instead we should signal the AGC when
   // the mic channel is muted/unmuted. We can't do it today because there
   // is no good way to know which stream is mapping to the mic channel.
-  bool all_muted = muted;
+  bool all_muted = mute;
   for (const auto& ch : send_channels_) {
     if (!all_muted) {
       break;
     }
     if (engine()->voe()->volume()->GetInputMute(ch.second->channel(),
                                                 all_muted)) {
       LOG_RTCERR1(GetInputMute, ch.second->channel());
       return false;
     }
   }
 
   webrtc::AudioProcessing* ap = engine()->voe()->base()->audio_processing();
   if (ap)
     ap->set_output_will_be_muted(all_muted);
-  return true;
+
+  if (!mute && options) {
+    return SetOptions(*options);
+  } else {
+    return true;
+  }
 }
 
 // TODO(minyue): SetMaxSendBandwidth() is subject to be renamed to
 // SetMaxSendBitrate() in future.
 bool WebRtcVoiceMediaChannel::SetMaxSendBandwidth(int bps) {
   LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetMaxSendBandwidth.";
 
   return SetSendBitrateInternal(bps);
 }
 
@@ skipping 539 matching lines @@
 
 int WebRtcSoundclipStream::Rewind() {
   mem_.Rewind();
   // Return -1 to keep VoiceEngine from looping.
   return (loop_) ? 0 : -1;
 }
 
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE