Index: talk/media/webrtc/webrtcvoiceengine.cc |
diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc |
index 9eee2af20219d710bfad79104c047ad905f33c76..ade7274fd09f2632ac55d4bad8442e60b2285e23 100644 |
--- a/talk/media/webrtc/webrtcvoiceengine.cc |
+++ b/talk/media/webrtc/webrtcvoiceengine.cc |
@@ -2186,6 +2186,9 @@ void WebRtcVoiceMediaChannel::OnPacketReceived( |
} |
default_recv_ssrc_ = ssrc; |
SetOutputVolume(default_recv_ssrc_, default_recv_volume_); |
+ if (default_sink_) { |
+ SetRawAudioSink(default_recv_ssrc_, std::move(default_sink_)); |
pthatcher1
2015/12/30 16:31:50
This looks different than default_recv_volume_ in
Taylor Brandstetter
2016/01/05 18:47:21
The first solution seems a little fragile. And it
|
+ } |
} |
// Forward packet to Call. If the SSRC is unknown we'll return after this. |
@@ -2415,6 +2418,14 @@ void WebRtcVoiceMediaChannel::SetRawAudioSink( |
rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) { |
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::SetRawAudioSink"; |
+ if (ssrc == 0) { |
+ if (default_recv_ssrc_ == -1) { |
+ default_sink_ = std::move(sink); |
+ } else { |
+ SetRawAudioSink(default_recv_ssrc_, std::move(sink)); |
+ } |
+ return; |
+ } |
const auto it = recv_streams_.find(ssrc); |
if (it == recv_streams_.end()) { |
LOG(LS_WARNING) << "SetRawAudioSink: no recv stream" << ssrc; |