| Index: talk/app/webrtc/webrtcsession.h
|
| diff --git a/talk/app/webrtc/webrtcsession.h b/talk/app/webrtc/webrtcsession.h
|
| index cd3f8967261232faee479eb26d370907b23566e9..384801454559c89c2442f89c302619b77aa1a59d 100644
|
| --- a/talk/app/webrtc/webrtcsession.h
|
| +++ b/talk/app/webrtc/webrtcsession.h
|
| @@ -309,6 +309,7 @@ class WebRtcSession : public AudioProviderInterface,
|
| void OnCertificateReady(
|
| const rtc::scoped_refptr<rtc::RTCCertificate>& certificate);
|
| void OnDtlsSetupFailure(cricket::BaseChannel*, bool rtcp);
|
| + void OnChannelFirstPacketReceived(cricket::BaseChannel*);
|
|
|
| // For unit test.
|
| bool waiting_for_certificate_for_testing() const;
|
| @@ -335,6 +336,9 @@ class WebRtcSession : public AudioProviderInterface,
|
| sigslot::signal2<const std::string&, const InternalDataChannelInit&>
|
| SignalDataChannelOpenMessage;
|
|
|
| + // Called when the first RTP packet is received.
|
| + sigslot::signal0<> SignalFirstMediaPacketReceived;
|
| +
|
| private:
|
| // Indicates the type of SessionDescription in a call to SetLocalDescription
|
| // and SetRemoteDescription.
|
| @@ -518,6 +522,8 @@ class WebRtcSession : public AudioProviderInterface,
|
| // Declares the RTCP mux policy for the WebRTCSession.
|
| PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy_;
|
|
|
| + bool has_received_media_packet_ = false;
|
| +
|
| RTC_DISALLOW_COPY_AND_ASSIGN(WebRtcSession);
|
| };
|
| } // namespace webrtc
|
|
|