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

Issue 1888593004: Delete all use of tick_util.h. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebase. Created 4 years, 7 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/logging.h" 15 #include "webrtc/base/logging.h"
16 #include "webrtc/common_types.h" 16 #include "webrtc/common_types.h"
17 #include "webrtc/config.h" 17 #include "webrtc/config.h"
18 #include "webrtc/modules/pacing/packet_router.h" 18 #include "webrtc/modules/pacing/packet_router.h"
19 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimat or.h" 19 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimat or.h"
20 #include "webrtc/modules/rtp_rtcp/include/fec_receiver.h" 20 #include "webrtc/modules/rtp_rtcp/include/fec_receiver.h"
21 #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h" 21 #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
22 #include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h" 22 #include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h"
23 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" 23 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
24 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h" 24 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
25 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" 25 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
26 #include "webrtc/modules/video_coding/video_coding_impl.h" 26 #include "webrtc/modules/video_coding/video_coding_impl.h"
27 #include "webrtc/system_wrappers/include/metrics.h" 27 #include "webrtc/system_wrappers/include/metrics.h"
28 #include "webrtc/system_wrappers/include/tick_util.h"
29 #include "webrtc/system_wrappers/include/timestamp_extrapolator.h" 28 #include "webrtc/system_wrappers/include/timestamp_extrapolator.h"
30 #include "webrtc/system_wrappers/include/trace.h" 29 #include "webrtc/system_wrappers/include/trace.h"
31 #include "webrtc/video/receive_statistics_proxy.h" 30 #include "webrtc/video/receive_statistics_proxy.h"
32 #include "webrtc/video/vie_remb.h" 31 #include "webrtc/video/vie_remb.h"
33 32
34 namespace webrtc { 33 namespace webrtc {
35 34
36 std::unique_ptr<RtpRtcp> CreateRtpRtcpModule( 35 std::unique_ptr<RtpRtcp> CreateRtpRtcpModule(
37 ReceiveStatistics* receive_statistics, 36 ReceiveStatistics* receive_statistics,
38 Transport* outgoing_transport, 37 Transport* outgoing_transport,
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537 const std::string& extension, int id) { 536 const std::string& extension, int id) {
538 // One-byte-extension local identifiers are in the range 1-14 inclusive. 537 // One-byte-extension local identifiers are in the range 1-14 inclusive.
539 RTC_DCHECK_GE(id, 1); 538 RTC_DCHECK_GE(id, 1);
540 RTC_DCHECK_LE(id, 14); 539 RTC_DCHECK_LE(id, 14);
541 RTC_DCHECK(RtpExtension::IsSupportedForVideo(extension)); 540 RTC_DCHECK(RtpExtension::IsSupportedForVideo(extension));
542 RTC_CHECK(rtp_header_parser_->RegisterRtpHeaderExtension( 541 RTC_CHECK(rtp_header_parser_->RegisterRtpHeaderExtension(
543 StringToRtpExtensionType(extension), id)); 542 StringToRtpExtensionType(extension), id));
544 } 543 }
545 544
546 } // namespace webrtc 545 } // namespace webrtc
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