OLD | NEW |
1 /* | 1 /* |
2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 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 | 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 | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
11 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_ | 11 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_ |
12 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_ | 12 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_ |
13 | 13 |
14 #include <string> | 14 #include <string> |
15 | 15 |
16 #include "webrtc/base/constructormagic.h" | 16 #include "webrtc/base/constructormagic.h" |
17 #include "webrtc/modules/include/module_common_types.h" | 17 #include "webrtc/modules/include/module_common_types.h" |
18 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" | 18 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
19 | 19 |
20 namespace webrtc { | 20 namespace webrtc { |
| 21 class RtpPacketToSend; |
21 | 22 |
22 class RtpPacketizer { | 23 class RtpPacketizer { |
23 public: | 24 public: |
24 static RtpPacketizer* Create(RtpVideoCodecTypes type, | 25 static RtpPacketizer* Create(RtpVideoCodecTypes type, |
25 size_t max_payload_len, | 26 size_t max_payload_len, |
26 const RTPVideoTypeHeader* rtp_type_header, | 27 const RTPVideoTypeHeader* rtp_type_header, |
27 FrameType frame_type); | 28 FrameType frame_type); |
28 | 29 |
29 virtual ~RtpPacketizer() {} | 30 virtual ~RtpPacketizer() {} |
30 | 31 |
31 virtual void SetPayloadData(const uint8_t* payload_data, | 32 virtual void SetPayloadData(const uint8_t* payload_data, |
32 size_t payload_size, | 33 size_t payload_size, |
33 const RTPFragmentationHeader* fragmentation) = 0; | 34 const RTPFragmentationHeader* fragmentation) = 0; |
34 | 35 |
35 // Get the next payload with payload header. | 36 // Get the next payload with payload header. |
36 // buffer is a pointer to where the output will be written. | 37 // Write payload and set marker bit of the |packet|. |
37 // bytes_to_send is an output variable that will contain number of bytes | 38 // The parameter |last_packet| is true for the last packet of the frame, false |
38 // written to buffer. The parameter last_packet is true for the last packet of | 39 // otherwise (i.e., call the function again to get the next packet). |
39 // the frame, false otherwise (i.e., call the function again to get the | 40 // Returns true on success, false otherwise. |
40 // next packet). | 41 virtual bool NextPacket(RtpPacketToSend* packet, bool* last_packet) = 0; |
41 // Returns true on success or false if there was no payload to packetize. | |
42 virtual bool NextPacket(uint8_t* buffer, | |
43 size_t* bytes_to_send, | |
44 bool* last_packet) = 0; | |
45 | 42 |
46 virtual ProtectionType GetProtectionType() = 0; | 43 virtual ProtectionType GetProtectionType() = 0; |
47 | 44 |
48 virtual StorageType GetStorageType(uint32_t retransmission_settings) = 0; | 45 virtual StorageType GetStorageType(uint32_t retransmission_settings) = 0; |
49 | 46 |
50 virtual std::string ToString() = 0; | 47 virtual std::string ToString() = 0; |
51 }; | 48 }; |
52 | 49 |
53 // TODO(sprang): Update the depacketizer to return a std::unqie_ptr with a copy | 50 // TODO(sprang): Update the depacketizer to return a std::unqie_ptr with a copy |
54 // of the parsed payload, rather than just a pointer into the incoming buffer. | 51 // of the parsed payload, rather than just a pointer into the incoming buffer. |
(...skipping 12 matching lines...) Expand all Loading... |
67 | 64 |
68 virtual ~RtpDepacketizer() {} | 65 virtual ~RtpDepacketizer() {} |
69 | 66 |
70 // Parses the RTP payload, parsed result will be saved in |parsed_payload|. | 67 // Parses the RTP payload, parsed result will be saved in |parsed_payload|. |
71 virtual bool Parse(ParsedPayload* parsed_payload, | 68 virtual bool Parse(ParsedPayload* parsed_payload, |
72 const uint8_t* payload_data, | 69 const uint8_t* payload_data, |
73 size_t payload_data_length) = 0; | 70 size_t payload_data_length) = 0; |
74 }; | 71 }; |
75 } // namespace webrtc | 72 } // namespace webrtc |
76 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_ | 73 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_ |
OLD | NEW |