commit 2d110be77f14cab0bb51efe8b61d9c7a967d04cb
author deadbeef <deadbeef@webrtc.org> Wed Jan 13 12:00:26 2016 -0800
committer Commit bot <commit-bot@chromium.org> Wed Jan 13 20:00:29 2016 +0000
tree 9c9d51bc720ca8975e9e41a8f50f31828e1dd700
parent 8432e1f4b84f79c4eea3a0820f4c4a83c267ef80

Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ )

Reason for revert:
tommi pointed out that using a refptr for the sink may cause issues. Will reland with a slightly different approach.

Original issue's description:
> Storing raw audio sink for default audio track.
>
> BUG=webrtc:5250
>
> Committed: https://crrev.com/e591f9377f33f3f725a30faecd1bef1a71fa6b99
> Cr-Commit-Position: refs/heads/master@{#11230}

TBR=pthatcher@webrtc.org,solenberg@webrtc.org,pbos@webrtc.org,tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5250

Review URL: https://codereview.webrtc.org/1588693002

Cr-Commit-Position: refs/heads/master@{#11241}

talk/media/webrtc/webrtcvoiceengine.cc

diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc
index 962cb54..9742564 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) {