| /* |
| * Copyright 2017 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #ifndef WEBRTC_PC_RTPTRANSPORTINTERNAL_H_ |
| #define WEBRTC_PC_RTPTRANSPORTINTERNAL_H_ |
| |
| #include "webrtc/api/ortc/rtptransportinterface.h" |
| #include "webrtc/rtc_base/sigslot.h" |
| |
| namespace rtc { |
| class CopyOnWriteBuffer; |
| struct PacketOptions; |
| struct PacketTime; |
| } // namespace rtc |
| |
| namespace webrtc { |
| |
| // This represents the internal interface beneath RtpTransportInterface; |
| // it is not accessible to API consumers but is accessible to internal classes |
| // in order to send and receive RTP and RTCP packets belonging to a single RTP |
| // session. Additional convenience and configuration methods are also provided. |
| class RtpTransportInternal : public RtpTransportInterface, |
| public sigslot::has_slots<> { |
| public: |
| virtual void SetRtcpMuxEnabled(bool enable) = 0; |
| |
| // TODO(zstein): Remove PacketTransport setters. Clients should pass these |
| // in to constructors instead and construct a new RtpTransportInternal instead |
| // of updating them. |
| |
| virtual rtc::PacketTransportInternal* rtp_packet_transport() const = 0; |
| virtual void SetRtpPacketTransport(rtc::PacketTransportInternal* rtp) = 0; |
| |
| virtual rtc::PacketTransportInternal* rtcp_packet_transport() const = 0; |
| virtual void SetRtcpPacketTransport(rtc::PacketTransportInternal* rtcp) = 0; |
| |
| // Called whenever a transport's ready-to-send state changes. The argument |
| // is true if all used transports are ready to send. This is more specific |
| // than just "writable"; it means the last send didn't return ENOTCONN. |
| sigslot::signal1<bool> SignalReadyToSend; |
| |
| // TODO(zstein): Consider having two signals - RtpPacketReceived and |
| // RtcpPacketReceived. |
| // The first argument is true for RTCP packets and false for RTP packets. |
| sigslot::signal3<bool, rtc::CopyOnWriteBuffer*, const rtc::PacketTime&> |
| SignalPacketReceived; |
| |
| virtual bool IsWritable(bool rtcp) const = 0; |
| |
| virtual bool SendPacket(bool rtcp, |
| rtc::CopyOnWriteBuffer* packet, |
| const rtc::PacketOptions& options, |
| int flags) = 0; |
| |
| virtual bool HandlesPayloadType(int payload_type) const = 0; |
| |
| virtual void AddHandledPayloadType(int payload_type) = 0; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // WEBRTC_PC_RTPTRANSPORTINTERNAL_H_ |