1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
|
/*
* 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 <stdint.h>
#include "api/array_view.h"
namespace webrtc {
// TODO(holmer): Look into unifying this with the PacketOptions in
// asyncpacketsocket.h.
struct PacketOptions {
PacketOptions();
PacketOptions(const PacketOptions&);
~PacketOptions();
// Negative ids are invalid and should be interpreted
// as packet_id not being set.
int64_t packet_id = -1;
// Whether this is an audio or video packet, excluding retransmissions.
// Defaults to `false` which is the more common case.
bool is_media = false;
bool included_in_feedback = false;
bool included_in_allocation = false;
bool send_as_ect1 = false;
// Whether this packet can be part of a packet batch at lower levels.
bool batchable = false;
// Whether this packet is the last of a batch.
bool last_packet_in_batch = false;
};
class Transport {
public:
virtual bool SendRtp(ArrayView<const uint8_t> packet,
const PacketOptions& options) = 0;
virtual bool SendRtcp(ArrayView<const uint8_t> packet,
const PacketOptions& options) = 0;
protected:
virtual ~Transport() = default;
};
} // namespace webrtc
#endif // API_CALL_TRANSPORT_H_
|
