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

Issue 1765443002: Added log messages for important call setup events: first packet sent/received (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Moved first packet logging from rtp_receiver_impl.cc to rtp_receiver_audio/video.cc Created 4 years, 9 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
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109 int8_t fec_payload_type_ GUARDED_BY(crit_); 109 int8_t fec_payload_type_ GUARDED_BY(crit_);
110 FecProtectionParams delta_fec_params_ GUARDED_BY(crit_); 110 FecProtectionParams delta_fec_params_ GUARDED_BY(crit_);
111 FecProtectionParams key_fec_params_ GUARDED_BY(crit_); 111 FecProtectionParams key_fec_params_ GUARDED_BY(crit_);
112 ProducerFec producer_fec_ GUARDED_BY(crit_); 112 ProducerFec producer_fec_ GUARDED_BY(crit_);
113 113
114 // Bitrate used for FEC payload, RED headers, RTP headers for FEC packets 114 // Bitrate used for FEC payload, RED headers, RTP headers for FEC packets
115 // and any padding overhead. 115 // and any padding overhead.
116 Bitrate _fecOverheadRate; 116 Bitrate _fecOverheadRate;
117 // Bitrate used for video payload and RTP headers 117 // Bitrate used for video payload and RTP headers
118 Bitrate _videoBitrate; 118 Bitrate _videoBitrate;
119 bool sent_first_frame_ GUARDED_BY(crit_);
119 }; 120 };
120 } // namespace webrtc 121 } // namespace webrtc
121 122
122 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_VIDEO_H_ 123 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_VIDEO_H_
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