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Side by Side Diff: webrtc/modules/rtp_rtcp/source/rtp_format_vp9.h

Issue 1417683006: modules: more interface -> include renames (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Fix last incorrectly wrapped pragma message Created 5 years, 1 month ago
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1 /* 1 /*
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2015 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 // 11 //
12 // This file contains the declaration of the VP9 packetizer class. 12 // This file contains the declaration of the VP9 packetizer class.
13 // A packetizer object is created for each encoded video frame. The 13 // A packetizer object is created for each encoded video frame. The
14 // constructor is called with the payload data and size. 14 // constructor is called with the payload data and size.
15 // 15 //
16 // After creating the packetizer, the method NextPacket is called 16 // After creating the packetizer, the method NextPacket is called
17 // repeatedly to get all packets for the frame. The method returns 17 // repeatedly to get all packets for the frame. The method returns
18 // false as long as there are more packets left to fetch. 18 // false as long as there are more packets left to fetch.
19 // 19 //
20 20
21 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP9_H_ 21 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP9_H_
22 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP9_H_ 22 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP9_H_
23 23
24 #include <queue> 24 #include <queue>
25 #include <string> 25 #include <string>
26 26
27 #include "webrtc/base/constructormagic.h" 27 #include "webrtc/base/constructormagic.h"
28 #include "webrtc/modules/interface/module_common_types.h" 28 #include "webrtc/modules/include/module_common_types.h"
29 #include "webrtc/modules/rtp_rtcp/source/rtp_format.h" 29 #include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
30 #include "webrtc/typedefs.h" 30 #include "webrtc/typedefs.h"
31 31
32 namespace webrtc { 32 namespace webrtc {
33 33
34 class RtpPacketizerVp9 : public RtpPacketizer { 34 class RtpPacketizerVp9 : public RtpPacketizer {
35 public: 35 public:
36 RtpPacketizerVp9(const RTPVideoHeaderVP9& hdr, size_t max_payload_length); 36 RtpPacketizerVp9(const RTPVideoHeaderVP9& hdr, size_t max_payload_length);
37 37
38 virtual ~RtpPacketizerVp9(); 38 virtual ~RtpPacketizerVp9();
(...skipping 60 matching lines...) Expand 10 before | Expand all | Expand 10 after
99 public: 99 public:
100 virtual ~RtpDepacketizerVp9() {} 100 virtual ~RtpDepacketizerVp9() {}
101 101
102 bool Parse(ParsedPayload* parsed_payload, 102 bool Parse(ParsedPayload* parsed_payload,
103 const uint8_t* payload, 103 const uint8_t* payload,
104 size_t payload_length) override; 104 size_t payload_length) override;
105 }; 105 };
106 106
107 } // namespace webrtc 107 } // namespace webrtc
108 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP9_H_ 108 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP9_H_
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