1 /* 2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 3 * 4 * Use of this source code is governed by a BSD-style license 5 * that can be found in the LICENSE file in the root of the source 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11 #ifndef API_CALL_TRANSPORT_H_ 12 #define API_CALL_TRANSPORT_H_ 13 14 #include <stddef.h> 15 #include <stdint.h> 16 17 #include "api/ref_counted_base.h" 18 #include "api/scoped_refptr.h" 19 20 namespace webrtc { 21 22 // TODO(holmer): Look into unifying this with the PacketOptions in 23 // asyncpacketsocket.h. 24 struct PacketOptions { 25 PacketOptions(); 26 PacketOptions(const PacketOptions&); 27 ~PacketOptions(); 28 29 // A 16 bits positive id. Negative ids are invalid and should be interpreted 30 // as packet_id not being set. 31 int packet_id = -1; 32 // Additional data bound to the RTP packet for use in application code, 33 // outside of WebRTC. 34 rtc::scoped_refptr<rtc::RefCountedBase> additional_data; 35 // Whether this is a retransmission of an earlier packet. 36 bool is_retransmit = false; 37 bool included_in_feedback = false; 38 bool included_in_allocation = false; 39 }; 40 41 class Transport { 42 public: 43 virtual bool SendRtp(const uint8_t* packet, 44 size_t length, 45 const PacketOptions& options) = 0; 46 virtual bool SendRtcp(const uint8_t* packet, size_t length) = 0; 47 48 protected: ~Transport()49 virtual ~Transport() {} 50 }; 51 52 } // namespace webrtc 53 54 #endif // API_CALL_TRANSPORT_H_ 55