mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-15 06:40:43 +01:00

Bug: webrtc:8789 Change-Id: I79c9132123c946b030ed79c647b4329e81d6e6ae Reviewed-on: https://webrtc-review.googlesource.com/43201 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Stefan Holmer <stefan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21837}
43 lines
1.6 KiB
C++
43 lines
1.6 KiB
C++
/*
|
|
* 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 { 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_SEND_BEFORE_KEY_FRAME_MS = 100 };
|
|
enum { RTCP_MAX_REPORT_BLOCKS = 31 }; // RFC 3550 page 37
|
|
enum {
|
|
kRtcpAppCode_DATA_SIZE = 32 * 4
|
|
}; // multiple of 4, this is not a limitation of the size
|
|
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 // MODULES_RTP_RTCP_SOURCE_RTP_RTCP_CONFIG_H_
|