text: /*
   *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
   *
   *  Use of this source code is governed by a BSD-style license
   *  that can be found in the LICENSE file in the root of the source
   *  tree. An additional intellectual property rights grant can be found
   *  in the file PATENTS.  All contributing project authors may
   *  be found in the AUTHORS file in the root of the source tree.
   */
  
  #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_
  #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_
  
  #include <string>
  
  #include "webrtc/base/constructormagic.h"
  #include "webrtc/modules/include/module_common_types.h"
  #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
  
  namespace webrtc {
  
  class RtpPacketizer {
   public:
    static RtpPacketizer* Create(RtpVideoCodecTypes type,
                                 size_t max_payload_len,
                                 const RTPVideoTypeHeader* rtp_type_header,
                                 FrameType frame_type);
  
    virtual ~RtpPacketizer() {}
  
    virtual void SetPayloadData(const uint8_t* payload_data,
                                size_t payload_size,
                                const RTPFragmentationHeader* fragmentation) = 0;
  
    // Get the next payload with payload header.
    // buffer is a pointer to where the output will be written.
    // bytes_to_send is an output variable that will contain number of bytes
    // written to buffer. The parameter last_packet is true for the last packet of
    // the frame, false otherwise (i.e., call the function again to get the
    // next packet).
    // Returns true on success or false if there was no payload to packetize.
    virtual bool NextPacket(uint8_t* buffer,
                            size_t* bytes_to_send,
                            bool* last_packet) = 0;
  
    virtual ProtectionType GetProtectionType() = 0;
  
    virtual StorageType GetStorageType(uint32_t retransmission_settings) = 0;
  
    virtual std::string ToString() = 0;
  };
  
  class RtpDepacketizer {
   public:
    struct ParsedPayload {
      const uint8_t* payload;
      size_t payload_length;
      FrameType frame_type;
      RTPTypeHeader type;
    };
  
    static RtpDepacketizer* Create(RtpVideoCodecTypes type);
  
    virtual ~RtpDepacketizer() {}
  
    // Parses the RTP payload, parsed result will be saved in |parsed_payload|.
    virtual bool Parse(ParsedPayload* parsed_payload,
                       const uint8_t* payload_data,
                       size_t payload_data_length) = 0;
  };
  }  // namespace webrtc
  #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_
  