Line data    Source code

       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             : class RtpPacketToSend;
      22             : 
      23           0 : class RtpPacketizer {
      24             :  public:
      25             :   static RtpPacketizer* Create(RtpVideoCodecTypes type,
      26             :                                size_t max_payload_len,
      27             :                                const RTPVideoTypeHeader* rtp_type_header,
      28             :                                FrameType frame_type);
      29             : 
      30           0 :   virtual ~RtpPacketizer() {}
      31             : 
      32             :   virtual void SetPayloadData(const uint8_t* payload_data,
      33             :                               size_t payload_size,
      34             :                               const RTPFragmentationHeader* fragmentation) = 0;
      35             : 
      36             :   // Get the next payload with payload header.
      37             :   // Write payload and set marker bit of the |packet|.
      38             :   // The parameter |last_packet| is true for the last packet of the frame, false
      39             :   // otherwise (i.e., call the function again to get the next packet).
      40             :   // Returns true on success, false otherwise.
      41             :   virtual bool NextPacket(RtpPacketToSend* packet, bool* last_packet) = 0;
      42             : 
      43             :   virtual ProtectionType GetProtectionType() = 0;
      44             : 
      45             :   virtual StorageType GetStorageType(uint32_t retransmission_settings) = 0;
      46             : 
      47             :   virtual std::string ToString() = 0;
      48             : };
      49             : 
      50             : // TODO(sprang): Update the depacketizer to return a std::unqie_ptr with a copy
      51             : // of the parsed payload, rather than just a pointer into the incoming buffer.
      52             : // This way we can move some parsing out from the jitter buffer into here, and
      53             : // the jitter buffer can just store that pointer rather than doing a copy there.
      54           0 : class RtpDepacketizer {
      55             :  public:
      56             :   struct ParsedPayload {
      57             :     const uint8_t* payload;
      58             :     size_t payload_length;
      59             :     FrameType frame_type;
      60             :     RTPTypeHeader type;
      61             :   };
      62             : 
      63             :   static RtpDepacketizer* Create(RtpVideoCodecTypes type);
      64             : 
      65           0 :   virtual ~RtpDepacketizer() {}
      66             : 
      67             :   // Parses the RTP payload, parsed result will be saved in |parsed_payload|.
      68             :   virtual bool Parse(ParsedPayload* parsed_payload,
      69             :                      const uint8_t* payload_data,
      70             :                      size_t payload_data_length) = 0;
      71             : };
      72             : }  // namespace webrtc
      73             : #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_