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Issue 2380683005: Moved RtcEventLog files from call/ to logging/ (new top level dir) (Closed)
Patch Set: Rebase to master Created 4 years, 2 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/modules/rtp_rtcp/source/rtp_sender.h" 11 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
12 12
13 #include <algorithm> 13 #include <algorithm>
14 #include <utility> 14 #include <utility>
15 15
16 #include "webrtc/base/checks.h" 16 #include "webrtc/base/checks.h"
17 #include "webrtc/base/logging.h" 17 #include "webrtc/base/logging.h"
18 #include "webrtc/base/rate_limiter.h" 18 #include "webrtc/base/rate_limiter.h"
19 #include "webrtc/base/trace_event.h" 19 #include "webrtc/base/trace_event.h"
20 #include "webrtc/base/timeutils.h" 20 #include "webrtc/base/timeutils.h"
21 #include "webrtc/call.h" 21 #include "webrtc/call.h"
22 #include "webrtc/call/rtc_event_log.h" 22 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
23 #include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h" 23 #include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h"
24 #include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h" 24 #include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h"
25 #include "webrtc/modules/rtp_rtcp/source/byte_io.h" 25 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
26 #include "webrtc/modules/rtp_rtcp/source/playout_delay_oracle.h" 26 #include "webrtc/modules/rtp_rtcp/source/playout_delay_oracle.h"
27 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h" 27 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h"
28 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h" 28 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h"
29 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h" 29 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h"
30 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_video.h" 30 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_video.h"
31 #include "webrtc/modules/rtp_rtcp/source/time_util.h" 31 #include "webrtc/modules/rtp_rtcp/source/time_util.h"
32 32
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1763 rtc::CritScope lock(&send_critsect_); 1763 rtc::CritScope lock(&send_critsect_);
1764 1764
1765 RtpState state; 1765 RtpState state;
1766 state.sequence_number = sequence_number_rtx_; 1766 state.sequence_number = sequence_number_rtx_;
1767 state.start_timestamp = timestamp_offset_; 1767 state.start_timestamp = timestamp_offset_;
1768 1768
1769 return state; 1769 return state;
1770 } 1770 }
1771 1771
1772 } // namespace webrtc 1772 } // namespace webrtc
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