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24e704f148
Move that calculation into dedicated function, move comment why it is calculated the way it is into the same function. Cleanup that comment - remove parts unused by current code, in particular remove description of code that was deleted a while ago Use more strict types for the calculation to make it clearer. Replace DCHECK result can't be zero with a clamp to ensure it can't be zero, because with large bitrates it may. Reland of https://webrtc-review.googlesource.com/c/src/+/315143 Bug: None Change-Id: I41ce383a2f19d489e4cae0b1bf1f720e0ffdd17a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315460 Reviewed-by: Åsa Persson <asapersson@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40538}
22 lines
864 B
C++
22 lines
864 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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constexpr int kDefaultMaxReorderingThreshold = 50; // In sequence numbers.
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constexpr int kRtcpMaxNackFields = 253;
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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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