|  | /* | 
|  | *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 
|  | * | 
|  | *  Use of this source code is governed by a BSD-style license | 
|  | *  that can be found in the LICENSE file in the root of the source | 
|  | *  tree. An additional intellectual property rights grant can be found | 
|  | *  in the file PATENTS.  All contributing project authors may | 
|  | *  be found in the AUTHORS file in the root of the source tree. | 
|  | */ | 
|  |  | 
|  | #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_CONFIG_H_ | 
|  | #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_CONFIG_H_ | 
|  |  | 
|  | // Configuration file for RTP utilities (RTPSender, RTPReceiver ...) | 
|  | namespace webrtc { | 
|  | enum { NACK_BYTECOUNT_SIZE = 60 };  // size of our NACK history | 
|  | // A sanity for the NACK list parsing at the send-side. | 
|  | enum { kSendSideNackListSizeSanity = 20000 }; | 
|  | enum { kDefaultMaxReorderingThreshold = 50 };  // In sequence numbers. | 
|  | enum { kRtcpMaxNackFields = 253 }; | 
|  |  | 
|  | enum { RTCP_INTERVAL_VIDEO_MS = 1000 }; | 
|  | enum { RTCP_INTERVAL_AUDIO_MS = 5000 }; | 
|  | enum { RTCP_SEND_BEFORE_KEY_FRAME_MS = 100 }; | 
|  | enum { RTCP_MAX_REPORT_BLOCKS = 31 };  // RFC 3550 page 37 | 
|  | enum { RTCP_MIN_FRAME_LENGTH_MS = 17 }; | 
|  | enum { | 
|  | kRtcpAppCode_DATA_SIZE = 32 * 4 | 
|  | };  // multiple of 4, this is not a limitation of the size | 
|  | enum { RTCP_RPSI_DATA_SIZE = 30 }; | 
|  | enum { RTCP_NUMBER_OF_SR = 60 }; | 
|  |  | 
|  | enum { MAX_NUMBER_OF_TEMPORAL_ID = 8 };              // RFC | 
|  | enum { MAX_NUMBER_OF_DEPENDENCY_QUALITY_ID = 128 };  // RFC | 
|  | enum { MAX_NUMBER_OF_REMB_FEEDBACK_SSRCS = 255 }; | 
|  |  | 
|  | enum { BW_HISTORY_SIZE = 35 }; | 
|  |  | 
|  | #define MIN_AUDIO_BW_MANAGEMENT_BITRATE 6 | 
|  | #define MIN_VIDEO_BW_MANAGEMENT_BITRATE 30 | 
|  |  | 
|  | enum { RTP_MAX_BURST_SLEEP_TIME = 500 }; | 
|  | enum { RTP_AUDIO_LEVEL_UNIQUE_ID = 0xbede }; | 
|  | enum { RTP_MAX_PACKETS_PER_FRAME = 512 };  // must be multiple of 32 | 
|  | }  // namespace webrtc | 
|  |  | 
|  | #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_CONFIG_H_ |