| /* |
| * 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 MODULES_RTP_RTCP_SOURCE_RTP_RTCP_CONFIG_H_ |
| #define MODULES_RTP_RTCP_SOURCE_RTP_RTCP_CONFIG_H_ |
| |
| // Configuration file for RTP utilities (RTPSender, RTPReceiver ...) |
| namespace webrtc { |
| enum { kDefaultMaxReorderingThreshold = 50 }; // In sequence numbers. |
| enum { kRtcpMaxNackFields = 253 }; |
| |
| enum { RTCP_SEND_BEFORE_KEY_FRAME_MS = 100 }; |
| enum { RTCP_MAX_REPORT_BLOCKS = 31 }; // RFC 3550 page 37 |
| } // namespace webrtc |
| |
| #endif // MODULES_RTP_RTCP_SOURCE_RTP_RTCP_CONFIG_H_ |