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
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
|
/*
* Copyright (c) 2014 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 MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_
#define MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_
#include <string>
#include "modules/include/module_common_types.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "rtc_base/constructormagic.h"
namespace webrtc {
class RtpPacketToSend;
class RtpPacketizer {
public:
static RtpPacketizer* Create(RtpVideoCodecTypes type,
size_t max_payload_len,
size_t last_packet_reduction_len,
const RTPVideoTypeHeader* rtp_type_header,
FrameType frame_type);
virtual ~RtpPacketizer() {}
// Returns total number of packets which would be produced by the packetizer.
virtual size_t SetPayloadData(
const uint8_t* payload_data,
size_t payload_size,
const RTPFragmentationHeader* fragmentation) = 0;
// Get the next payload with payload header.
// Write payload and set marker bit of the |packet|.
// Returns true on success, false otherwise.
virtual bool NextPacket(RtpPacketToSend* packet) = 0;
virtual std::string ToString() = 0;
};
// TODO(sprang): Update the depacketizer to return a std::unqie_ptr with a copy
// of the parsed payload, rather than just a pointer into the incoming buffer.
// This way we can move some parsing out from the jitter buffer into here, and
// the jitter buffer can just store that pointer rather than doing a copy there.
class RtpDepacketizer {
public:
struct ParsedPayload {
const uint8_t* payload;
size_t payload_length;
FrameType frame_type;
RTPTypeHeader type;
};
static RtpDepacketizer* Create(RtpVideoCodecTypes type);
virtual ~RtpDepacketizer() {}
// Parses the RTP payload, parsed result will be saved in |parsed_payload|.
virtual bool Parse(ParsedPayload* parsed_payload,
const uint8_t* payload_data,
size_t payload_data_length) = 0;
};
} // namespace webrtc
#endif // MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_
|
