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Side by Side Diff: webrtc/modules/rtp_rtcp/source/rtp_sender.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) 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2012 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_SENDER_H_ 11 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_
12 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_ 12 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_
13 #include <assert.h> 13 #include <assert.h>
14 #include <math.h> 14 #include <math.h>
15 15
16 #include <map> 16 #include <map>
17 17
18 #include "webrtc/base/thread_annotations.h" 18 #include "webrtc/base/thread_annotations.h"
19 #include "webrtc/common_types.h" 19 #include "webrtc/common_types.h"
20 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" 20 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
21 #include "webrtc/modules/rtp_rtcp/source/bitrate.h" 21 #include "webrtc/modules/rtp_rtcp/source/bitrate.h"
22 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h" 22 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
23 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_history.h" 23 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_history.h"
24 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h" 24 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
25 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" 25 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
26 #include "webrtc/modules/rtp_rtcp/source/ssrc_database.h" 26 #include "webrtc/modules/rtp_rtcp/source/ssrc_database.h"
27 #include "webrtc/transport.h" 27 #include "webrtc/transport.h"
28 28
29 #define MAX_INIT_RTP_SEQ_NUMBER 32767 // 2^15 -1. 29 #define MAX_INIT_RTP_SEQ_NUMBER 32767 // 2^15 -1.
30 30
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461 // SetTargetBitrateKbps or GetTargetBitrateKbps. Also remember 461 // SetTargetBitrateKbps or GetTargetBitrateKbps. Also remember
462 // that by the time the function returns there is no guarantee 462 // that by the time the function returns there is no guarantee
463 // that the target bitrate is still valid. 463 // that the target bitrate is still valid.
464 rtc::scoped_ptr<CriticalSectionWrapper> target_bitrate_critsect_; 464 rtc::scoped_ptr<CriticalSectionWrapper> target_bitrate_critsect_;
465 uint32_t target_bitrate_ GUARDED_BY(target_bitrate_critsect_); 465 uint32_t target_bitrate_ GUARDED_BY(target_bitrate_critsect_);
466 }; 466 };
467 467
468 } // namespace webrtc 468 } // namespace webrtc
469 469
470 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_ 470 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_
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