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_H264_H_ 12 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H264_H_ 13 14 #include <queue> 15 16 #include "webrtc/modules/rtp_rtcp/source/rtp_format.h" 17 18 namespace webrtc { 19 20 class RtpPacketizerH264 : public RtpPacketizer { 21 public: 22 // Initialize with payload from encoder. 23 // The payload_data must be exactly one encoded H264 frame. 24 RtpPacketizerH264(FrameType frame_type, size_t max_payload_len); 25 26 virtual ~RtpPacketizerH264(); 27 28 virtual void SetPayloadData( 29 const uint8_t* payload_data, 30 size_t payload_size, 31 const RTPFragmentationHeader* fragmentation) OVERRIDE; 32 33 // Get the next payload with H264 payload header. 34 // buffer is a pointer to where the output will be written. 35 // bytes_to_send is an output variable that will contain number of bytes 36 // written to buffer. The parameter last_packet is true for the last packet of 37 // the frame, false otherwise (i.e., call the function again to get the 38 // next packet). 39 // Returns true on success or false if there was no payload to packetize. 40 virtual bool NextPacket(uint8_t* buffer, 41 size_t* bytes_to_send, 42 bool* last_packet) OVERRIDE; 43 44 virtual ProtectionType GetProtectionType() OVERRIDE; 45 46 virtual StorageType GetStorageType(uint32_t retransmission_settings) OVERRIDE; 47 48 virtual std::string ToString() OVERRIDE; 49 50 private: 51 struct Packet { 52 Packet(size_t offset, 53 size_t size, 54 bool first_fragment, 55 bool last_fragment, 56 bool aggregated, 57 uint8_t header) 58 : offset(offset), 59 size(size), 60 first_fragment(first_fragment), 61 last_fragment(last_fragment), 62 aggregated(aggregated), 63 header(header) {} 64 65 size_t offset; 66 size_t size; 67 bool first_fragment; 68 bool last_fragment; 69 bool aggregated; 70 uint8_t header; 71 }; 72 typedef std::queue<Packet> PacketQueue; 73 74 void GeneratePackets(); 75 void PacketizeFuA(size_t fragment_offset, size_t fragment_length); 76 int PacketizeStapA(size_t fragment_index, 77 size_t fragment_offset, 78 size_t fragment_length); 79 void NextAggregatePacket(uint8_t* buffer, size_t* bytes_to_send); 80 void NextFragmentPacket(uint8_t* buffer, size_t* bytes_to_send); 81 82 const uint8_t* payload_data_; 83 size_t payload_size_; 84 const size_t max_payload_len_; 85 RTPFragmentationHeader fragmentation_; 86 PacketQueue packets_; 87 FrameType frame_type_; 88 89 DISALLOW_COPY_AND_ASSIGN(RtpPacketizerH264); 90 }; 91 92 // Depacketizer for H264. 93 class RtpDepacketizerH264 : public RtpDepacketizer { 94 public: 95 explicit RtpDepacketizerH264(RtpData* const callback); 96 97 virtual ~RtpDepacketizerH264() {} 98 99 virtual bool Parse(WebRtcRTPHeader* rtp_header, 100 const uint8_t* payload_data, 101 size_t payload_data_length) OVERRIDE; 102 103 private: 104 RtpData* const callback_; 105 106 DISALLOW_COPY_AND_ASSIGN(RtpDepacketizerH264); 107 }; 108 } // namespace webrtc 109 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H264_H_ 110