Forward the SignalFirstPacketReceived to RtpReceiver.

webrtc/api/mediastreamprovider.h

 /*
  *  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 54 matching lines @@
 
   virtual RtpParameters GetAudioRtpSendParameters(uint32_t ssrc) const = 0;
   virtual bool SetAudioRtpSendParameters(uint32_t ssrc,
                                          const RtpParameters& parameters) = 0;
 
   virtual RtpParameters GetAudioRtpReceiveParameters(uint32_t ssrc) const = 0;
   virtual bool SetAudioRtpReceiveParameters(
       uint32_t ssrc,
       const RtpParameters& parameters) = 0;
 
+  // Called when the first audio packet is received.
+  sigslot::signal0<> SignalFirstAudioPacketReceived;
+
  protected:
   virtual ~AudioProviderInterface() {}
 };
 
 // This interface is called by VideoRtpSender/Receivers to change the settings
 // of a video track connected to a certain PeerConnection.
 class VideoProviderInterface {
  public:
   virtual bool SetSource(
       uint32_t ssrc,
@@ skipping 10 matching lines @@
 
   virtual RtpParameters GetVideoRtpSendParameters(uint32_t ssrc) const = 0;
   virtual bool SetVideoRtpSendParameters(uint32_t ssrc,
                                          const RtpParameters& parameters) = 0;
 
   virtual RtpParameters GetVideoRtpReceiveParameters(uint32_t ssrc) const = 0;
   virtual bool SetVideoRtpReceiveParameters(
       uint32_t ssrc,
       const RtpParameters& parameters) = 0;
 
+  // Called when the first video packet is received.
+  sigslot::signal0<> SignalFirstVideoPacketReceived;
+
  protected:
   virtual ~VideoProviderInterface() {}
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_API_MEDIASTREAMPROVIDER_H_