55 lines
1.6 KiB
C++
55 lines
1.6 KiB
C++
/*
|
|
* 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 "api/ref_counted_base.h"
|
|
#include "api/scoped_refptr.h"
|
|
|
|
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.
|
|
rtc::scoped_refptr<rtc::RefCountedBase> additional_data;
|
|
// Whether this is a retransmission of an earlier packet.
|
|
bool is_retransmit = false;
|
|
bool included_in_feedback = false;
|
|
bool included_in_allocation = 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_
|