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Side by Side Diff: webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc

Issue 1413333002: system_wrappers: rename interface -> include (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebased again! 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 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_video.h" 11 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_video.h"
12 12
13 #include <stdlib.h> 13 #include <stdlib.h>
14 #include <string.h> 14 #include <string.h>
15 15
16 #include <vector> 16 #include <vector>
17 17
18 #include "webrtc/base/checks.h" 18 #include "webrtc/base/checks.h"
19 #include "webrtc/base/logging.h" 19 #include "webrtc/base/logging.h"
20 #include "webrtc/base/trace_event.h" 20 #include "webrtc/base/trace_event.h"
21 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" 21 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
22 #include "webrtc/modules/rtp_rtcp/source/byte_io.h" 22 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
23 #include "webrtc/modules/rtp_rtcp/source/producer_fec.h" 23 #include "webrtc/modules/rtp_rtcp/source/producer_fec.h"
24 #include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h" 24 #include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h"
25 #include "webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h" 25 #include "webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h"
26 #include "webrtc/modules/rtp_rtcp/source/rtp_format_vp9.h" 26 #include "webrtc/modules/rtp_rtcp/source/rtp_format_vp9.h"
27 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" 27 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
28 28
29 namespace webrtc { 29 namespace webrtc {
30 enum { REDForFECHeaderLength = 1 }; 30 enum { REDForFECHeaderLength = 1 };
31 31
32 struct RtpPacket { 32 struct RtpPacket {
33 uint16_t rtpHeaderLength; 33 uint16_t rtpHeaderLength;
34 ForwardErrorCorrection::Packet* pkt; 34 ForwardErrorCorrection::Packet* pkt;
35 }; 35 };
36 36
37 RTPSenderVideo::RTPSenderVideo(Clock* clock, RTPSenderInterface* rtpSender) 37 RTPSenderVideo::RTPSenderVideo(Clock* clock, RTPSenderInterface* rtpSender)
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350 CriticalSectionScoped cs(crit_.get()); 350 CriticalSectionScoped cs(crit_.get());
351 return _retransmissionSettings; 351 return _retransmissionSettings;
352 } 352 }
353 353
354 void RTPSenderVideo::SetSelectiveRetransmissions(uint8_t settings) { 354 void RTPSenderVideo::SetSelectiveRetransmissions(uint8_t settings) {
355 CriticalSectionScoped cs(crit_.get()); 355 CriticalSectionScoped cs(crit_.get());
356 _retransmissionSettings = settings; 356 _retransmissionSettings = settings;
357 } 357 }
358 358
359 } // namespace webrtc 359 } // namespace webrtc
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