Forward the SignalFirstPacketReceived to RtpReceiver.

webrtc/api/webrtcsession.cc

 /*
  *  Copyright 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 1749 matching lines @@
       bundle_transport, create_rtcp_transport_channel, audio_options_));
   if (!voice_channel_) {
     return false;
   }
   if (require_rtcp_mux) {
     voice_channel_->ActivateRtcpMux();
   }
 
   voice_channel_->SignalDtlsSetupFailure.connect(
       this, &WebRtcSession::OnDtlsSetupFailure);
+  voice_channel_->SignalFirstPacketReceived.connect(
+      this, &WebRtcSession::OnChannelFirstPacketReceived);
 
   SignalVoiceChannelCreated();
   voice_channel_->SignalSentPacket.connect(this,
                                            &WebRtcSession::OnSentPacket_w);
   return true;
 }
 
 bool WebRtcSession::CreateVideoChannel(const cricket::ContentInfo* content,
                                        const std::string* bundle_transport) {
   bool require_rtcp_mux =
       rtcp_mux_policy_ == PeerConnectionInterface::kRtcpMuxPolicyRequire;
   bool create_rtcp_transport_channel = !require_rtcp_mux;
   video_channel_.reset(channel_manager_->CreateVideoChannel(
       media_controller_, transport_controller_.get(), content->name,
       bundle_transport, create_rtcp_transport_channel, video_options_));
   if (!video_channel_) {
     return false;
   }
   if (require_rtcp_mux) {
     video_channel_->ActivateRtcpMux();
   }
   video_channel_->SignalDtlsSetupFailure.connect(
       this, &WebRtcSession::OnDtlsSetupFailure);
+  video_channel_->SignalFirstPacketReceived.connect(
+      this, &WebRtcSession::OnChannelFirstPacketReceived);
 
   SignalVideoChannelCreated();
   video_channel_->SignalSentPacket.connect(this,
                                            &WebRtcSession::OnSentPacket_w);
   return true;
 }
 
 bool WebRtcSession::CreateDataChannel(const cricket::ContentInfo* content,
                                       const std::string* bundle_transport) {
   bool sctp = (data_channel_type_ == cricket::DCT_SCTP);
@@ skipping 21 matching lines @@
   SignalDataChannelCreated();
   data_channel_->SignalSentPacket.connect(this, &WebRtcSession::OnSentPacket_w);
   return true;
 }
 
 void WebRtcSession::OnDtlsSetupFailure(cricket::BaseChannel*, bool rtcp) {
   SetError(ERROR_TRANSPORT,
            rtcp ? kDtlsSetupFailureRtcp : kDtlsSetupFailureRtp);
 }
 
+void WebRtcSession::OnChannelFirstPacketReceived(
+    cricket::BaseChannel* channel) {
+  ASSERT(signaling_thread()->IsCurrent());
+
+  if (!received_first_audio_packet_ &&
+      channel->media_type() == cricket::MEDIA_TYPE_AUDIO) {
+    received_first_audio_packet_ = true;
+    SignalFirstAudioPacketReceived();
+  } else if (!received_first_video_packet_ &&
+             channel->media_type() == cricket::MEDIA_TYPE_VIDEO) {
+    received_first_video_packet_ = true;
+    SignalFirstVideoPacketReceived();
+  }
+}
+
 void WebRtcSession::OnDataChannelMessageReceived(
     cricket::DataChannel* channel,
     const cricket::ReceiveDataParams& params,
     const rtc::CopyOnWriteBuffer& payload) {
   RTC_DCHECK(data_channel_type_ == cricket::DCT_SCTP);
   if (params.type == cricket::DMT_CONTROL && IsOpenMessage(payload)) {
     // Received OPEN message; parse and signal that a new data channel should
     // be created.
     std::string label;
     InternalDataChannelInit config;
@@ skipping 298 matching lines @@
                                                   ssl_cipher_suite);
   }
 }
 
 void WebRtcSession::OnSentPacket_w(const rtc::SentPacket& sent_packet) {
   RTC_DCHECK(worker_thread()->IsCurrent());
   media_controller_->call_w()->OnSentPacket(sent_packet);
 }
 
 }  // namespace webrtc