Storing raw audio sink for default audio track.

talk/media/webrtc/webrtcvoiceengine.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 1226 matching lines @@
     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
     RTC_DCHECK(stream_);
     return stream_->GetStats();
   }
 
   int channel() const {
     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
     return config_.voe_channel_id;
   }
 
-  void SetRawAudioSink(rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) {
+  void SetRawAudioSink(
+      const rtc::scoped_refptr<webrtc::AudioSinkInterface>& sink) {
     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
-    stream_->SetSink(std::move(sink));
+    stream_->SetSink(sink);
   }
 
  private:
   void RecreateAudioReceiveStream(bool use_combined_bwe,
       const std::vector<webrtc::RtpExtension>& extensions) {
     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
     if (stream_) {
       call_->DestroyAudioReceiveStream(stream_);
       stream_ = nullptr;
     }
@@ skipping 919 matching lines @@
   if (default_recv_ssrc_ == -1 && GetReceiveChannelId(ssrc) == -1) {
     StreamParams sp;
     sp.ssrcs.push_back(ssrc);
     LOG(LS_INFO) << "Creating default receive stream for SSRC=" << ssrc << ".";
     if (!AddRecvStream(sp)) {
       LOG(LS_WARNING) << "Could not create default receive stream.";
       return;
     }
     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.
   const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp,
                                               packet_time.not_before);
   webrtc::PacketReceiver::DeliveryStatus delivery_result =
       call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO,
           reinterpret_cast<const uint8_t*>(packet->data()), packet->size(),
           webrtc_packet_time);
   if (webrtc::PacketReceiver::DELIVERY_OK != delivery_result) {
@@ skipping 206 matching lines @@
     rinfo.decoding_plc_cng = stats.decoding_plc_cng;
     rinfo.capture_start_ntp_time_ms = stats.capture_start_ntp_time_ms;
     info->receivers.push_back(rinfo);
   }
 
   return true;
 }
 
 void WebRtcVoiceMediaChannel::SetRawAudioSink(
     uint32_t ssrc,
-    rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) {
+    const rtc::scoped_refptr<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(std::move(sink));
+  it->second->SetRawAudioSink(sink);
 }
 
 int WebRtcVoiceMediaChannel::GetOutputLevel(int channel) {
   unsigned int ulevel = 0;
   int ret = engine()->voe()->volume()->GetSpeechOutputLevel(channel, ulevel);
   return (ret == 0) ? static_cast<int>(ulevel) : -1;
 }
 
 int WebRtcVoiceMediaChannel::GetReceiveChannelId(uint32_t ssrc) const {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
@@ skipping 66 matching lines @@
     }
   } else {
     LOG(LS_INFO) << "Stopping playout for channel #" << channel;
     engine()->voe()->base()->StopPlayout(channel);
   }
   return true;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE