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92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
45 lines
1.7 KiB
C++
45 lines
1.7 KiB
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 { NACK_BYTECOUNT_SIZE = 60 }; // size of our NACK history
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// A sanity for the NACK list parsing at the send-side.
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enum { kSendSideNackListSizeSanity = 20000 };
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enum { kDefaultMaxReorderingThreshold = 50 }; // In sequence numbers.
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enum { kRtcpMaxNackFields = 253 };
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enum { RTCP_INTERVAL_VIDEO_MS = 1000 };
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enum { RTCP_INTERVAL_AUDIO_MS = 5000 };
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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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enum {
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kRtcpAppCode_DATA_SIZE = 32 * 4
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}; // multiple of 4, this is not a limitation of the size
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enum { RTCP_NUMBER_OF_SR = 60 };
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enum { MAX_NUMBER_OF_TEMPORAL_ID = 8 }; // RFC
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enum { MAX_NUMBER_OF_DEPENDENCY_QUALITY_ID = 128 }; // RFC
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enum { MAX_NUMBER_OF_REMB_FEEDBACK_SSRCS = 255 };
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enum { BW_HISTORY_SIZE = 35 };
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#define MIN_AUDIO_BW_MANAGEMENT_BITRATE 6
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#define MIN_VIDEO_BW_MANAGEMENT_BITRATE 30
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enum { RTP_MAX_BURST_SLEEP_TIME = 500 };
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enum { RTP_AUDIO_LEVEL_UNIQUE_ID = 0xbede };
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enum { RTP_MAX_PACKETS_PER_FRAME = 512 }; // must be multiple of 32
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} // namespace webrtc
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#endif // MODULES_RTP_RTCP_SOURCE_RTP_RTCP_CONFIG_H_
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