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

talk/media/webrtc/webrtcvoiceengine.cc

Index: talk/media/webrtc/webrtcvoiceengine.cc
diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc
index 93f4b97b14ad38cecf9c0aeedfd732606c5b3624..843cb60761a5f98e26486efbab9e3179b816d669 100644
--- a/talk/media/webrtc/webrtcvoiceengine.cc
+++ b/talk/media/webrtc/webrtcvoiceengine.cc
@@ -2374,6 +2374,67 @@ bool WebRtcVoiceMediaChannel::ChangeSend(int channel, SendFlags send) {
   return true;
 }
 
+bool WebRtcVoiceMediaChannel::SetAudioSend(uint32 ssrc, bool mute,
+                                           const AudioOptions* options,
+                                           AudioRenderer* renderer) {
+  // SetLocalRenderer
+  {
+    ChannelMap::iterator it = send_channels_.find(ssrc);
+    if (it == send_channels_.end()) {
+      if (renderer) {
+        // Return error if trying to set a valid renderer with an invalid ssrc.
+        LOG(LS_ERROR) << "SetLocalRenderer failed with ssrc "<< ssrc;
+        return false;
+      }
+      // The channel likely has gone away, do nothing.
+    } else {
+      if (renderer) {
+        it->second->Start(renderer);
+      } else {
+        it->second->Stop();
+      }
+    }
+  }
+
+  // MuteStream
+  {
+    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, 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 = 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);
+
+    if (!mute && options) {
+      return SetOptions(*options);
+    }
+  }
+
+  return true;

[pthatcher1, 2015/09/09 07:16:12]

Similarly, would it make more sense for this CL to keep them as private methods
and call them instead of moving the code?  It would make the CL more obvious to
read.

[the sun, 2015/09/09 09:50:50]

Done.

+}
+
 // TODO(ronghuawu): Change this method to return bool.
 void WebRtcVoiceMediaChannel::ConfigureSendChannel(int channel) {
   if (engine()->voe()->network()->RegisterExternalTransport(
@@ -2723,28 +2784,6 @@ bool WebRtcVoiceMediaChannel::SetRemoteRenderer(uint32 ssrc,
   return true;
 }
 
-bool WebRtcVoiceMediaChannel::SetLocalRenderer(uint32 ssrc,
-                                               AudioRenderer* renderer) {
-  ChannelMap::iterator it = send_channels_.find(ssrc);
-  if (it == send_channels_.end()) {
-    if (renderer) {
-      // Return an error if trying to set a valid renderer with an invalid ssrc.
-      LOG(LS_ERROR) << "SetLocalRenderer failed with ssrc "<< ssrc;
-      return false;
-    }
-
-    // The channel likely has gone away, do nothing.
-    return true;
-  }
-
-  if (renderer)
-    it->second->Start(renderer);
-  else
-    it->second->Stop();
-
-  return true;
-}
-
 bool WebRtcVoiceMediaChannel::GetActiveStreams(
     AudioInfo::StreamList* actives) {
   // In conference mode, the default channel should not be in
@@ -3032,38 +3071,6 @@ void WebRtcVoiceMediaChannel::OnRtcpReceived(
   }
 }
 
-bool WebRtcVoiceMediaChannel::MuteStream(uint32 ssrc, bool muted) {
-  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);
-    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;
-  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;
-}
-
 // TODO(minyue): SetMaxSendBandwidth() is subject to be renamed to
 // SetMaxSendBitrate() in future.
 bool WebRtcVoiceMediaChannel::SetMaxSendBandwidth(int bps) {