Chromium Code Reviews| Index: talk/media/webrtc/webrtcvoiceengine.cc | 
| diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc | 
| index 9742564985bca6c0437865dafbe76803608e384a..79c807564a504e71939687582447e8fb749ded89 100644 | 
| --- a/talk/media/webrtc/webrtcvoiceengine.cc | 
| +++ b/talk/media/webrtc/webrtcvoiceengine.cc | 
| @@ -132,6 +132,16 @@ const int kMaxTelephoneEventCode = 255; | 
| const int kMinTelephoneEventDuration = 100; | 
| const int kMaxTelephoneEventDuration = 60000; // Actual limit is 2^16 | 
| +class ProxySink : public webrtc::AudioSinkInterface { | 
| + public: | 
| + ProxySink(AudioSinkInterface* sink) : sink_(sink) {} | 
| + | 
| + void OnData(const Data& audio) override { sink_->OnData(audio); } | 
| + | 
| + private: | 
| + webrtc::AudioSinkInterface* sink_; | 
| +}; | 
| + | 
| bool ValidateStreamParams(const StreamParams& sp) { | 
| if (sp.ssrcs.empty()) { | 
| LOG(LS_ERROR) << "No SSRCs in stream parameters: " << sp.ToString(); | 
| @@ -2186,6 +2196,9 @@ void WebRtcVoiceMediaChannel::OnPacketReceived( | 
| } | 
| default_recv_ssrc_ = ssrc; | 
| SetOutputVolume(default_recv_ssrc_, default_recv_volume_); | 
| + rtc::scoped_ptr<webrtc::AudioSinkInterface> proxy_sink( | 
| + default_sink_ ? new ProxySink(default_sink_.get()) : nullptr); | 
| + SetRawAudioSink(default_recv_ssrc_, std::move(proxy_sink)); | 
| 
 
tommi
2016/01/13 23:14:24
Do we actually want to make this call if default_s
 
Taylor Brandstetter
2016/01/13 23:55:12
I had "if (default_sink_)" originally, but I was s
 
the sun
2016/01/14 09:24:39
Yes, now that we can avoid allocating an object it
 
 | 
| } | 
| // Forward packet to Call. If the SSRC is unknown we'll return after this. | 
| @@ -2414,7 +2427,17 @@ void WebRtcVoiceMediaChannel::SetRawAudioSink( | 
| uint32_t ssrc, | 
| rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) { | 
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); | 
| - LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::SetRawAudioSink"; | 
| + LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::SetRawAudioSink: ssrc:" << ssrc | 
| + << " " << (sink ? "(ptr)" : "NULL"); | 
| + if (ssrc == 0) { | 
| + if (default_recv_ssrc_ != -1) { | 
| + rtc::scoped_ptr<webrtc::AudioSinkInterface> proxy_sink( | 
| + sink ? new ProxySink(sink.get()) : nullptr); | 
| + SetRawAudioSink(default_recv_ssrc_, std::move(proxy_sink)); | 
| + } | 
| + default_sink_ = std::move(sink); | 
| 
 
tommi
2016/01/13 23:14:24
nit: do you mind moving this above the check for -
 
Taylor Brandstetter
2016/01/13 23:55:12
Is the sink accessed on any other threads? If so I
 
the sun
2016/01/14 09:24:39
I'd still like you to avoid the duplicate log line
 
tommi (sloooow) - chröme
2016/01/14 12:34:13
ah of course
 
Taylor Brandstetter
2016/01/14 15:48:47
My reasoning is that SetRenderer does the same thi
 
the sun
2016/01/14 15:51:44
Acknowledged.
 
 | 
| + return; | 
| + } | 
| const auto it = recv_streams_.find(ssrc); | 
| if (it == recv_streams_.end()) { | 
| LOG(LS_WARNING) << "SetRawAudioSink: no recv stream" << ssrc; |