1/*
2 *  Copyright (c) 2014 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 WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_
12#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_
13
14#include <string>
15
16#include "webrtc/base/constructormagic.h"
17#include "webrtc/modules/include/module_common_types.h"
18#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
19
20namespace webrtc {
21
22class RtpPacketizer {
23 public:
24  static RtpPacketizer* Create(RtpVideoCodecTypes type,
25                               size_t max_payload_len,
26                               const RTPVideoTypeHeader* rtp_type_header,
27                               FrameType frame_type);
28
29  virtual ~RtpPacketizer() {}
30
31  virtual void SetPayloadData(const uint8_t* payload_data,
32                              size_t payload_size,
33                              const RTPFragmentationHeader* fragmentation) = 0;
34
35  // Get the next payload with payload header.
36  // buffer is a pointer to where the output will be written.
37  // bytes_to_send is an output variable that will contain number of bytes
38  // written to buffer. The parameter last_packet is true for the last packet of
39  // the frame, false otherwise (i.e., call the function again to get the
40  // next packet).
41  // Returns true on success or false if there was no payload to packetize.
42  virtual bool NextPacket(uint8_t* buffer,
43                          size_t* bytes_to_send,
44                          bool* last_packet) = 0;
45
46  virtual ProtectionType GetProtectionType() = 0;
47
48  virtual StorageType GetStorageType(uint32_t retransmission_settings) = 0;
49
50  virtual std::string ToString() = 0;
51};
52
53class RtpDepacketizer {
54 public:
55  struct ParsedPayload {
56    const uint8_t* payload;
57    size_t payload_length;
58    FrameType frame_type;
59    RTPTypeHeader type;
60  };
61
62  static RtpDepacketizer* Create(RtpVideoCodecTypes type);
63
64  virtual ~RtpDepacketizer() {}
65
66  // Parses the RTP payload, parsed result will be saved in |parsed_payload|.
67  virtual bool Parse(ParsedPayload* parsed_payload,
68                     const uint8_t* payload_data,
69                     size_t payload_data_length) = 0;
70};
71}  // namespace webrtc
72#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_
73