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Bug: None Change-Id: Iced341f0574e26ac3be3292870fb7d7522b75ce1 Reviewed-on: https://webrtc-review.googlesource.com/93280 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24238}
23 lines
895 B
C++
23 lines
895 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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// Configuration file for RTP utilities (RTPSender, RTPReceiver ...)
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namespace webrtc {
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enum { kDefaultMaxReorderingThreshold = 50 }; // In sequence numbers.
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enum { kRtcpMaxNackFields = 253 };
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enum { RTCP_SEND_BEFORE_KEY_FRAME_MS = 100 };
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enum { 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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