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/interface/module_common_types.h" 18 #include "webrtc/modules/rtp_rtcp/interface/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 static RtpDepacketizer* Create(RtpVideoCodecTypes type, 56 RtpData* const callback); 57 ~RtpDepacketizer()58 virtual ~RtpDepacketizer() {} 59 60 virtual bool Parse(WebRtcRTPHeader* rtp_header, 61 const uint8_t* payload_data, 62 size_t payload_data_length) = 0; 63 }; 64 } // namespace webrtc 65 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_ 66