Line data    Source code

       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/base/constructormagic.h"
      16             : #include "webrtc/common_types.h"
      17             : #include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
      18             : #include "webrtc/typedefs.h"
      19             : 
      20             : namespace webrtc {
      21             : namespace RtpFormatVideoGeneric {
      22             : static const uint8_t kKeyFrameBit = 0x01;
      23             : static const uint8_t kFirstPacketBit = 0x02;
      24             : }  // namespace RtpFormatVideoGeneric
      25             : 
      26             : class RtpPacketizerGeneric : public RtpPacketizer {
      27             :  public:
      28             :   // Initialize with payload from encoder.
      29             :   // The payload_data must be exactly one encoded generic frame.
      30             :   RtpPacketizerGeneric(FrameType frametype, size_t max_payload_len);
      31             : 
      32             :   virtual ~RtpPacketizerGeneric();
      33             : 
      34             :   void SetPayloadData(const uint8_t* payload_data,
      35             :                       size_t payload_size,
      36             :                       const RTPFragmentationHeader* fragmentation) override;
      37             : 
      38             :   // Get the next payload with generic payload header.
      39             :   // Write payload and set marker bit of the |packet|.
      40             :   // The parameter |last_packet| is true for the last packet of the frame, false
      41             :   // otherwise (i.e., call the function again to get the next packet).
      42             :   // Returns true on success, false otherwise.
      43             :   bool NextPacket(RtpPacketToSend* packet, bool* last_packet) override;
      44             : 
      45             :   ProtectionType GetProtectionType() override;
      46             : 
      47             :   StorageType GetStorageType(uint32_t retransmission_settings) override;
      48             : 
      49             :   std::string ToString() override;
      50             : 
      51             :  private:
      52             :   const uint8_t* payload_data_;
      53             :   size_t payload_size_;
      54             :   const size_t max_payload_len_;
      55             :   FrameType frame_type_;
      56             :   size_t payload_length_;
      57             :   uint8_t generic_header_;
      58             : 
      59             :   RTC_DISALLOW_COPY_AND_ASSIGN(RtpPacketizerGeneric);
      60             : };
      61             : 
      62             : // Depacketizer for generic codec.
      63           0 : class RtpDepacketizerGeneric : public RtpDepacketizer {
      64             :  public:
      65           0 :   virtual ~RtpDepacketizerGeneric() {}
      66             : 
      67             :   bool Parse(ParsedPayload* parsed_payload,
      68             :              const uint8_t* payload_data,
      69             :              size_t payload_data_length) override;
      70             : };
      71             : }  // namespace webrtc
      72             : #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VIDEO_GENERIC_H_