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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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