2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
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.
11 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_
12 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_
16 #include "webrtc/base/constructormagic.h"
17 #include "webrtc/modules/interface/module_common_types.h"
18 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
24 static RtpPacketizer* Create(RtpVideoCodecTypes type,
25 size_t max_payload_len,
26 const RTPVideoTypeHeader* rtp_type_header,
27 FrameType frame_type);
29 virtual ~RtpPacketizer() {}
31 virtual void SetPayloadData(const uint8_t* payload_data,
33 const RTPFragmentationHeader* fragmentation) = 0;
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
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;
46 virtual ProtectionType GetProtectionType() = 0;
48 virtual StorageType GetStorageType(uint32_t retransmission_settings) = 0;
50 virtual std::string ToString() = 0;
53 class RtpDepacketizer {
55 struct ParsedPayload {
56 const uint8_t* payload;
57 size_t payload_length;
62 static RtpDepacketizer* Create(RtpVideoCodecTypes type);
64 virtual ~RtpDepacketizer() {}
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;
72 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_