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 114 matching lines @@
 #else
 const char kAecDumpByAudioOptionFilename[] = "audio.aecdump";
 #endif
 
 // Constants from voice_engine_defines.h.
 const int kMinTelephoneEventCode = 0;           // RFC4733 (Section 2.3.1)
 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) { RTC_DCHECK(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();
     return false;
   }
   if (sp.ssrcs.size() > 1) {
     LOG(LS_ERROR) << "Multiple SSRCs in stream parameters: " << sp.ToString();
     return false;
   }
   return true;
@@ skipping 2034 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_);
+    if (default_sink_) {
+      rtc::scoped_ptr<webrtc::AudioSinkInterface> proxy_sink(
+          new ProxySink(default_sink_.get()));
+      SetRawAudioSink(default_recv_ssrc_, std::move(proxy_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 208 matching lines @@
     info->receivers.push_back(rinfo);
   }
 
   return true;
 }
 
 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);
+    return;
+  }
   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));
 }
 
 int WebRtcVoiceMediaChannel::GetOutputLevel(int channel) {
   unsigned int ulevel = 0;
@@ skipping 72 matching lines @@
     }
   } else {
     LOG(LS_INFO) << "Stopping playout for channel #" << channel;
     engine()->voe()->base()->StopPlayout(channel);
   }
   return true;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE