Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(160)

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

Issue 2964773002: Revert "Update includes for webrtc/{base => rtc_base} rename (1/3)" (Closed)
Patch Set: Created 3 years, 5 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
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_RTCP_PACKET_H_ 11 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_H_
12 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_H_ 12 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_H_
13 13
14 #include "webrtc/rtc_base/basictypes.h" 14 #include "webrtc/base/basictypes.h"
15 #include "webrtc/rtc_base/buffer.h" 15 #include "webrtc/base/buffer.h"
16 16
17 namespace webrtc { 17 namespace webrtc {
18 namespace rtcp { 18 namespace rtcp {
19 // Class for building RTCP packets. 19 // Class for building RTCP packets.
20 // 20 //
21 // Example: 21 // Example:
22 // ReportBlock report_block; 22 // ReportBlock report_block;
23 // report_block.SetMediaSsrc(234); 23 // report_block.SetMediaSsrc(234);
24 // report_block.SetFractionLost(10); 24 // report_block.SetFractionLost(10);
25 // 25 //
(...skipping 68 matching lines...) Expand 10 before | Expand all | Expand 10 after
94 94
95 bool OnBufferFull(uint8_t* packet, 95 bool OnBufferFull(uint8_t* packet,
96 size_t* index, 96 size_t* index,
97 PacketReadyCallback* callback) const; 97 PacketReadyCallback* callback) const;
98 // Size of the rtcp packet as written in header. 98 // Size of the rtcp packet as written in header.
99 size_t HeaderLength() const; 99 size_t HeaderLength() const;
100 }; 100 };
101 } // namespace rtcp 101 } // namespace rtcp
102 } // namespace webrtc 102 } // namespace webrtc
103 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_H_ 103 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_H_
OLDNEW
« no previous file with comments | « webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator.cc ('k') | webrtc/modules/rtp_rtcp/source/rtcp_packet.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698