| Index: talk/app/webrtc/webrtcsession.h
|
| diff --git a/talk/app/webrtc/webrtcsession.h b/talk/app/webrtc/webrtcsession.h
|
| index efb280fc48dc8d632342bf8b5ff648a9d7be11f3..12f9abc7fa47c8850bd602c86c8ed539f127078e 100644
|
| --- a/talk/app/webrtc/webrtcsession.h
|
| +++ b/talk/app/webrtc/webrtcsession.h
|
| @@ -310,6 +310,7 @@ 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;
|
| @@ -334,6 +335,9 @@ 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.
|
| @@ -517,6 +521,8 @@ 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
|
|
|