| /* |
| * Copyright (c) 2013 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 API_CALL_TRANSPORT_H_ |
| #define API_CALL_TRANSPORT_H_ |
| |
| #include <stddef.h> |
| #include <stdint.h> |
| #include <vector> |
| |
| namespace webrtc { |
| |
| // TODO(holmer): Look into unifying this with the PacketOptions in |
| // asyncpacketsocket.h. |
| struct PacketOptions { |
| PacketOptions(); |
| PacketOptions(const PacketOptions&); |
| ~PacketOptions(); |
| |
| // A 16 bits positive id. Negative ids are invalid and should be interpreted |
| // as packet_id not being set. |
| int packet_id = -1; |
| // Additional data bound to the RTP packet for use in application code, |
| // outside of WebRTC. |
| std::vector<uint8_t> application_data; |
| // Whether this is a retransmission of an earlier packet. |
| bool is_retransmit = false; |
| }; |
| |
| class Transport { |
| public: |
| virtual bool SendRtp(const uint8_t* packet, |
| size_t length, |
| const PacketOptions& options) = 0; |
| virtual bool SendRtcp(const uint8_t* packet, size_t length) = 0; |
| |
| protected: |
| virtual ~Transport() {} |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // API_CALL_TRANSPORT_H_ |