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 |