Index: webrtc/pc/rtptransport.h |
diff --git a/webrtc/pc/rtptransport.h b/webrtc/pc/rtptransport.h |
index f5ffe3fe5f154aad0956d81a1877358f50ad6e48..f7793282b9420a8e9d784f0bf91fb633282f2c4c 100644 |
--- a/webrtc/pc/rtptransport.h |
+++ b/webrtc/pc/rtptransport.h |
@@ -12,6 +12,7 @@ |
#define WEBRTC_PC_RTPTRANSPORT_H_ |
#include "webrtc/api/ortc/rtptransportinterface.h" |
+#include "webrtc/base/sigslot.h" |
namespace rtc { |
@@ -21,7 +22,7 @@ class PacketTransportInternal; |
namespace webrtc { |
-class RtpTransport : public RtpTransportInterface { |
+class RtpTransport : public RtpTransportInterface, public sigslot::has_slots<> { |
public: |
RtpTransport(const RtpTransport&) = delete; |
RtpTransport& operator=(const RtpTransport&) = delete; |
@@ -41,6 +42,14 @@ class RtpTransport : public RtpTransportInterface { |
} |
void set_rtcp_packet_transport(rtc::PacketTransportInternal* rtcp); |
+ void Connect(bool rtcp); |
+ void Disconnect(bool rtcp); |
+ |
+ void OnReadyToSend(rtc::PacketTransportInternal* transport); |
Taylor Brandstetter
2017/04/14 18:05:33
This should be private.
Zach Stein
2017/04/18 23:39:29
Done.
|
+ sigslot::signal1<bool> SignalReadyToSend; |
+ |
+ void SetReadyToSend(bool rtcp, bool ready); |
+ |
PacketTransportInterface* GetRtpPacketTransport() const override; |
PacketTransportInterface* GetRtcpPacketTransport() const override; |
@@ -60,6 +69,9 @@ class RtpTransport : public RtpTransportInterface { |
rtc::PacketTransportInternal* rtp_packet_transport_ = nullptr; |
rtc::PacketTransportInternal* rtcp_packet_transport_ = nullptr; |
+ bool rtp_ready_to_send_ = false; |
+ bool rtcp_ready_to_send_ = false; |
+ |
RtcpParameters rtcp_parameters_; |
}; |