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 
20 namespace webrtc {
21 
22 class 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 
~RtpPacketizer()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 
53 class 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 
~RtpDepacketizer()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