- Rename VoiceChannel::MuteStream() -> SetAudioSend() (incl. media channel)

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 1077 matching lines @@
   }
   if (send) {
     StartAllSendStreams();
   } else {
     StopAllSendStreams();
   }
   sending_ = send;
   return true;
 }
 
+bool WebRtcVideoChannel2::SetVideoSend(uint32 ssrc, bool mute,
+                                       const VideoOptions* options) {
+  LOG(LS_VERBOSE) << "SetVideoSend: " << 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);
+
+  if (!mute && options) {
+    return SetOptions(*options);
+  } else {
+    return true;
+  }
+}
+
 bool WebRtcVideoChannel2::ValidateSendSsrcAvailability(
     const StreamParams& sp) const {
   for (uint32_t ssrc: sp.ssrcs) {
     if (send_ssrcs_.find(ssrc) != send_ssrcs_.end()) {
       LOG(LS_ERROR) << "Send stream with SSRC '" << ssrc << "' already exists.";
       return false;
     }
   }
   return true;
 }
@@ skipping 399 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) {
-  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;
-}
-
 bool WebRtcVideoChannel2::SetRecvRtpHeaderExtensions(
     const std::vector<RtpHeaderExtension>& extensions) {
   TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetRecvRtpHeaderExtensions");
   LOG(LS_INFO) << "SetRecvRtpHeaderExtensions: "
                << RtpExtensionsToString(extensions);
   if (!ValidateRtpHeaderExtensionIds(extensions))
     return false;
 
   std::vector<webrtc::RtpExtension> filtered_extensions =
       FilterRtpExtensions(extensions);
@@ skipping 1222 matching lines @@
       video_codecs[i].rtx_payload_type = rtx_mapping[video_codecs[i].codec.id];
     }
   }
 
   return video_codecs;
 }
 
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VIDEO