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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 MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_ | 11 #ifndef MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_ |
12 #define MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_ | 12 #define MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_ |
13 | 13 |
14 #include <string> | 14 #include <string> |
15 | 15 |
16 #include "modules/include/module_common_types.h" | 16 #include "modules/include/module_common_types.h" |
17 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" | 17 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
18 #include "rtc_base/constructormagic.h" | 18 #include "rtc_base/constructormagic.h" |
19 | 19 |
20 namespace webrtc { | 20 namespace webrtc { |
21 class RtpPacketToSend; | 21 class RtpPacketToSend; |
22 | 22 |
23 class RtpPacketizer { | 23 class RtpPacketizer { |
24 public: | 24 public: |
25 static RtpPacketizer* Create(RtpVideoCodecTypes type, | 25 static RtpPacketizer* Create(RtpVideoCodecTypes type, |
26 size_t max_payload_len, | 26 size_t max_payload_len, |
27 size_t last_packet_reduction_len, | 27 size_t last_packet_reduction_len, |
28 const RTPVideoTypeHeader* rtp_type_header, | 28 const RTPVideoTypeHeader* rtp_type_header, |
| 29 const RTPVideoStereoInfo* stereoInfo, |
29 FrameType frame_type); | 30 FrameType frame_type); |
30 | 31 |
31 virtual ~RtpPacketizer() {} | 32 virtual ~RtpPacketizer() {} |
32 | 33 |
33 // Returns total number of packets which would be produced by the packetizer. | 34 // Returns total number of packets which would be produced by the packetizer. |
34 virtual size_t SetPayloadData( | 35 virtual size_t SetPayloadData( |
35 const uint8_t* payload_data, | 36 const uint8_t* payload_data, |
36 size_t payload_size, | 37 size_t payload_size, |
37 const RTPFragmentationHeader* fragmentation) = 0; | 38 const RTPFragmentationHeader* fragmentation) = 0; |
38 | 39 |
(...skipping 22 matching lines...) Expand all Loading... |
61 | 62 |
62 virtual ~RtpDepacketizer() {} | 63 virtual ~RtpDepacketizer() {} |
63 | 64 |
64 // Parses the RTP payload, parsed result will be saved in |parsed_payload|. | 65 // Parses the RTP payload, parsed result will be saved in |parsed_payload|. |
65 virtual bool Parse(ParsedPayload* parsed_payload, | 66 virtual bool Parse(ParsedPayload* parsed_payload, |
66 const uint8_t* payload_data, | 67 const uint8_t* payload_data, |
67 size_t payload_data_length) = 0; | 68 size_t payload_data_length) = 0; |
68 }; | 69 }; |
69 } // namespace webrtc | 70 } // namespace webrtc |
70 #endif // MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_ | 71 #endif // MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_ |
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