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Side by Side Diff: webrtc/video/rtp_stream_receiver.cc

Issue 2216533002: Move RTP for synchroninzation and rename classes, files and variables. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Indentation + rebase. Created 4 years, 4 months 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/video/rtp_stream_receiver.h" 11 #include "webrtc/video/rtp_stream_receiver.h"
12 12
13 #include <vector> 13 #include <vector>
14 14
15 #include "webrtc/base/checks.h"
15 #include "webrtc/base/logging.h" 16 #include "webrtc/base/logging.h"
16 #include "webrtc/common_types.h" 17 #include "webrtc/common_types.h"
17 #include "webrtc/config.h" 18 #include "webrtc/config.h"
18 #include "webrtc/modules/pacing/packet_router.h" 19 #include "webrtc/modules/pacing/packet_router.h"
19 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimat or.h" 20 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimat or.h"
20 #include "webrtc/modules/rtp_rtcp/include/fec_receiver.h" 21 #include "webrtc/modules/rtp_rtcp/include/fec_receiver.h"
21 #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h" 22 #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
22 #include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h" 23 #include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h"
23 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" 24 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
24 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h" 25 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
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538 const std::string& extension, int id) { 539 const std::string& extension, int id) {
539 // One-byte-extension local identifiers are in the range 1-14 inclusive. 540 // One-byte-extension local identifiers are in the range 1-14 inclusive.
540 RTC_DCHECK_GE(id, 1); 541 RTC_DCHECK_GE(id, 1);
541 RTC_DCHECK_LE(id, 14); 542 RTC_DCHECK_LE(id, 14);
542 RTC_DCHECK(RtpExtension::IsSupportedForVideo(extension)); 543 RTC_DCHECK(RtpExtension::IsSupportedForVideo(extension));
543 RTC_CHECK(rtp_header_parser_->RegisterRtpHeaderExtension( 544 RTC_CHECK(rtp_header_parser_->RegisterRtpHeaderExtension(
544 StringToRtpExtensionType(extension), id)); 545 StringToRtpExtensionType(extension), id));
545 } 546 }
546 547
547 } // namespace webrtc 548 } // namespace webrtc
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