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

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 2356 matching lines @@
     DCHECK(send == SEND_NOTHING);
     if (engine()->voe()->base()->StopSend(channel) == -1) {
       LOG_RTCERR1(StopSend, channel);
       return false;
     }
   }
 
   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(
           channel, *this) == -1) {
     LOG_RTCERR2(RegisterExternalTransport, channel, this);
   }
 
   // Enable RTCP (for quality stats and feedback messages)
   EnableRtcp(channel);
 
@@ skipping 329 matching lines @@
   }
 
   if (renderer)
     it->second->Start(renderer);
   else
     it->second->Stop();
 
   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
   // |receive_channels_|.
   actives->clear();
   for (const auto& ch : receive_channels_) {
     int level = GetOutputLevel(ch.second->channel());
     if (level > 0) {
       actives->push_back(std::make_pair(ch.first, level));
     }
@@ skipping 267 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) {
-  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) {
   LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetMaxSendBandwidth.";
 
   return SetSendBitrateInternal(bps);
 }
 
 bool WebRtcVoiceMediaChannel::SetSendBitrateInternal(int bps) {
   LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetSendBitrateInternal.";
@@ skipping 537 matching lines @@
 
 int WebRtcSoundclipStream::Rewind() {
   mem_.Rewind();
   // Return -1 to keep VoiceEngine from looping.
   return (loop_) ? 0 : -1;
 }
 
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE