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