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      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 
     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 
     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