Chromium Code Reviews| 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_; |
| }; |