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c6b9ac782a
This change migrates RTCPSender to use webrtc::Timestamp, preparing for later improvements regarding bugs.webrtc.org/11581. Fixed: webrtc:12873 Change-Id: I1159701dc373883367d9b2c86823f8fb59904d55 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222324 Commit-Queue: Markus Handell <handellm@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34346}
25 lines
971 B
C++
25 lines
971 B
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_RTP_RTCP_SOURCE_RTP_RTCP_CONFIG_H_
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#define MODULES_RTP_RTCP_SOURCE_RTP_RTCP_CONFIG_H_
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#include "api/units/time_delta.h"
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// Configuration file for RTP utilities (RTPSender, RTPReceiver ...)
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namespace webrtc {
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constexpr int kDefaultMaxReorderingThreshold = 5; // In sequence numbers.
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constexpr int kRtcpMaxNackFields = 253;
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constexpr TimeDelta RTCP_SEND_BEFORE_KEY_FRAME = TimeDelta::Millis(100);
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constexpr int RTCP_MAX_REPORT_BLOCKS = 31; // RFC 3550 page 37
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} // namespace webrtc
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#endif // MODULES_RTP_RTCP_SOURCE_RTP_RTCP_CONFIG_H_
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