becf9c897c
This is a second version of http://review.webrtc.org/1065006/ which passes the parameters via methods instead of via constructors. BUG=1289 Review URL: https://webrtc-codereview.appspot.com/1065007 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3456 4adac7df-926f-26a2-2b94-8c16560cd09d
53 lines
2.0 KiB
C++
53 lines
2.0 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 WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_CONFIG_H_
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#define WEBRTC_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 { kRtpRtcpMaxIdleTimeProcess = 5,
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kRtpRtcpBitrateProcessTimeMs = 10,
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kRtpRtcpPacketTimeoutProcessTimeMs = 100,
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kRtpRtcpRttProcessTimeMs = 1000 };
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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 { 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 { RTCP_MIN_FRAME_LENGTH_MS = 17};
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enum { kRtcpAppCode_DATA_SIZE = 32*4}; // multiple of 4, this is not a limitation of the size
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enum { RTCP_RPSI_DATA_SIZE = 30};
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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 { DTMF_OUTBAND_MAX = 20};
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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 // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_CONFIG_H_
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