Storing raw audio sink for default audio track.

talk/session/media/channel.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 27 matching lines @@
 #include "webrtc/base/byteorder.h"
 #include "webrtc/base/common.h"
 #include "webrtc/base/dscp.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/p2p/base/transportchannel.h"
 
 namespace cricket {
 using rtc::Bind;
 
-namespace {
-// See comment below for why we need to use a pointer to a scoped_ptr.
-bool SetRawAudioSink_w(VoiceMediaChannel* channel,
-                       uint32_t ssrc,
-                       rtc::scoped_ptr<webrtc::AudioSinkInterface>* sink) {
-  channel->SetRawAudioSink(ssrc, std::move(*sink));
-  return true;
-}
-}  // namespace
-
 enum {
   MSG_EARLYMEDIATIMEOUT = 1,
   MSG_SCREENCASTWINDOWEVENT,
   MSG_RTPPACKET,
   MSG_RTCPPACKET,
   MSG_CHANNEL_ERROR,
   MSG_READYTOSENDDATA,
   MSG_DATARECEIVED,
   MSG_FIRSTPACKETRECEIVED,
   MSG_STREAMCLOSEDREMOTELY,
@@ skipping 1324 matching lines @@
                              ssrc, event_code, duration));
 }
 
 bool VoiceChannel::SetOutputVolume(uint32_t ssrc, double volume) {
   return InvokeOnWorker(Bind(&VoiceMediaChannel::SetOutputVolume,
                              media_channel(), ssrc, volume));
 }
 
 void VoiceChannel::SetRawAudioSink(
     uint32_t ssrc,
-    rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) {
-  // We need to work around Bind's lack of support for scoped_ptr and ownership
-  // passing.  So we invoke to our own little routine that gets a pointer to
-  // our local variable.  This is OK since we're synchronously invoking.
-  InvokeOnWorker(Bind(&SetRawAudioSink_w, media_channel(), ssrc, &sink));
+    const rtc::scoped_refptr<webrtc::AudioSinkInterface>& sink) {
+  worker_thread()->Invoke<void>(
+      Bind(&VoiceMediaChannel::SetRawAudioSink, media_channel(), ssrc, sink));
 }
 
 bool VoiceChannel::GetStats(VoiceMediaInfo* stats) {
   return InvokeOnWorker(Bind(&VoiceMediaChannel::GetStats,
                              media_channel(), stats));
 }
 
 void VoiceChannel::StartMediaMonitor(int cms) {
   media_monitor_.reset(new VoiceMediaMonitor(media_channel(), worker_thread(),
       rtc::Thread::Current()));
@@ skipping 901 matching lines @@
   return (data_channel_type_ == DCT_RTP) && BaseChannel::ShouldSetupDtlsSrtp();
 }
 
 void DataChannel::OnStreamClosedRemotely(uint32_t sid) {
   rtc::TypedMessageData<uint32_t>* message =
       new rtc::TypedMessageData<uint32_t>(sid);
   signaling_thread()->Post(this, MSG_STREAMCLOSEDREMOTELY, message);
 }
 
 }  // namespace cricket