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      1 /*
      2  *  Copyright (c) 2013 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 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VIDEO_GENERIC_H_
     11 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VIDEO_GENERIC_H_
     12 
     13 #include "webrtc/common_types.h"
     14 #include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
     15 #include "webrtc/typedefs.h"
     16 
     17 namespace webrtc {
     18 namespace RtpFormatVideoGeneric {
     19 static const uint8_t kKeyFrameBit = 0x01;
     20 static const uint8_t kFirstPacketBit = 0x02;
     21 }  // namespace RtpFormatVideoGeneric
     22 
     23 class RtpPacketizerGeneric : public RtpPacketizer {
     24  public:
     25   // Initialize with payload from encoder.
     26   // The payload_data must be exactly one encoded generic frame.
     27   RtpPacketizerGeneric(FrameType frametype, size_t max_payload_len);
     28 
     29   virtual ~RtpPacketizerGeneric();
     30 
     31   virtual void SetPayloadData(
     32       const uint8_t* payload_data,
     33       size_t payload_size,
     34       const RTPFragmentationHeader* fragmentation) OVERRIDE;
     35 
     36   // Get the next payload with generic payload header.
     37   // buffer is a pointer to where the output will be written.
     38   // bytes_to_send is an output variable that will contain number of bytes
     39   // written to buffer. The parameter last_packet is true for the last packet of
     40   // the frame, false otherwise (i.e., call the function again to get the
     41   // next packet).
     42   // Returns true on success or false if there was no payload to packetize.
     43   virtual bool NextPacket(uint8_t* buffer,
     44                           size_t* bytes_to_send,
     45                           bool* last_packet) OVERRIDE;
     46 
     47   virtual ProtectionType GetProtectionType() OVERRIDE;
     48 
     49   virtual StorageType GetStorageType(uint32_t retransmission_settings) OVERRIDE;
     50 
     51   virtual std::string ToString() OVERRIDE;
     52 
     53  private:
     54   const uint8_t* payload_data_;
     55   size_t payload_size_;
     56   const size_t max_payload_len_;
     57   FrameType frame_type_;
     58   uint32_t payload_length_;
     59   uint8_t generic_header_;
     60 
     61   DISALLOW_COPY_AND_ASSIGN(RtpPacketizerGeneric);
     62 };
     63 
     64 // Depacketizer for generic codec.
     65 class RtpDepacketizerGeneric : public RtpDepacketizer {
     66  public:
     67   explicit RtpDepacketizerGeneric(RtpData* const callback);
     68 
     69   virtual ~RtpDepacketizerGeneric() {}
     70 
     71   virtual bool Parse(WebRtcRTPHeader* rtp_header,
     72                      const uint8_t* payload_data,
     73                      size_t payload_data_length) OVERRIDE;
     74 
     75  private:
     76   RtpData* const callback_;
     77 
     78   DISALLOW_COPY_AND_ASSIGN(RtpDepacketizerGeneric);
     79 };
     80 }  // namespace webrtc
     81 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VIDEO_GENERIC_H_
     82