Revert of Storing raw audio sink for default audio track.

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 a69c824fdf738af39380ce6195bd66a4af635a1c..9eee2af20219d710bfad79104c047ad905f33c76 100644
--- a/talk/media/webrtc/webrtcvoiceengine.cc
+++ b/talk/media/webrtc/webrtcvoiceengine.cc
@@ -1244,10 +1244,9 @@
     return config_.voe_channel_id;
   }
 
-  void SetRawAudioSink(
-      const rtc::scoped_refptr<webrtc::AudioSinkInterface>& sink) {
+  void SetRawAudioSink(rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) {
     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
-    stream_->SetSink(sink);
+    stream_->SetSink(std::move(sink));
   }
 
  private:
@@ -2187,7 +2186,6 @@
     }
     default_recv_ssrc_ = ssrc;
     SetOutputVolume(default_recv_ssrc_, default_recv_volume_);
-    SetRawAudioSink(default_recv_ssrc_, default_sink_);
   }
 
   // Forward packet to Call. If the SSRC is unknown we'll return after this.
@@ -2414,22 +2412,15 @@
 
 void WebRtcVoiceMediaChannel::SetRawAudioSink(
     uint32_t ssrc,
-    const rtc::scoped_refptr<webrtc::AudioSinkInterface>& sink) {
+    rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::SetRawAudioSink";
-  if (ssrc == 0) {
-    default_sink_ = sink;
-    if (default_recv_ssrc_ == -1) {
-      return;
-    }
-    ssrc = static_cast<uint32_t>(default_recv_ssrc_);
-  }
   const auto it = recv_streams_.find(ssrc);
   if (it == recv_streams_.end()) {
     LOG(LS_WARNING) << "SetRawAudioSink: no recv stream" << ssrc;
     return;
   }
-  it->second->SetRawAudioSink(sink);
+  it->second->SetRawAudioSink(std::move(sink));
 }
 
 int WebRtcVoiceMediaChannel::GetOutputLevel(int channel) {