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Side by Side Diff: webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h

Issue 2742383004: Delete support for receiving RTCP RPSI and SLI messages. (Closed)
Patch Set: Rebased. Created 3 years, 9 months ago
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1 /* 1 /*
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
11 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_CONFIG_H_ 11 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_CONFIG_H_
12 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_CONFIG_H_ 12 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_CONFIG_H_
13 13
14 // Configuration file for RTP utilities (RTPSender, RTPReceiver ...) 14 // Configuration file for RTP utilities (RTPSender, RTPReceiver ...)
15 namespace webrtc { 15 namespace webrtc {
16 enum { NACK_BYTECOUNT_SIZE = 60 }; // size of our NACK history 16 enum { NACK_BYTECOUNT_SIZE = 60 }; // size of our NACK history
17 // A sanity for the NACK list parsing at the send-side. 17 // A sanity for the NACK list parsing at the send-side.
18 enum { kSendSideNackListSizeSanity = 20000 }; 18 enum { kSendSideNackListSizeSanity = 20000 };
19 enum { kDefaultMaxReorderingThreshold = 50 }; // In sequence numbers. 19 enum { kDefaultMaxReorderingThreshold = 50 }; // In sequence numbers.
20 enum { kRtcpMaxNackFields = 253 }; 20 enum { kRtcpMaxNackFields = 253 };
21 21
22 enum { RTCP_INTERVAL_VIDEO_MS = 1000 }; 22 enum { RTCP_INTERVAL_VIDEO_MS = 1000 };
23 enum { RTCP_INTERVAL_AUDIO_MS = 5000 }; 23 enum { RTCP_INTERVAL_AUDIO_MS = 5000 };
24 enum { RTCP_SEND_BEFORE_KEY_FRAME_MS = 100 }; 24 enum { RTCP_SEND_BEFORE_KEY_FRAME_MS = 100 };
25 enum { RTCP_MAX_REPORT_BLOCKS = 31 }; // RFC 3550 page 37 25 enum { RTCP_MAX_REPORT_BLOCKS = 31 }; // RFC 3550 page 37
26 enum { 26 enum {
27 kRtcpAppCode_DATA_SIZE = 32 * 4 27 kRtcpAppCode_DATA_SIZE = 32 * 4
28 }; // multiple of 4, this is not a limitation of the size 28 }; // multiple of 4, this is not a limitation of the size
29 enum { RTCP_RPSI_DATA_SIZE = 30 };
30 enum { RTCP_NUMBER_OF_SR = 60 }; 29 enum { RTCP_NUMBER_OF_SR = 60 };
31 30
32 enum { MAX_NUMBER_OF_TEMPORAL_ID = 8 }; // RFC 31 enum { MAX_NUMBER_OF_TEMPORAL_ID = 8 }; // RFC
33 enum { MAX_NUMBER_OF_DEPENDENCY_QUALITY_ID = 128 }; // RFC 32 enum { MAX_NUMBER_OF_DEPENDENCY_QUALITY_ID = 128 }; // RFC
34 enum { MAX_NUMBER_OF_REMB_FEEDBACK_SSRCS = 255 }; 33 enum { MAX_NUMBER_OF_REMB_FEEDBACK_SSRCS = 255 };
35 34
36 enum { BW_HISTORY_SIZE = 35 }; 35 enum { BW_HISTORY_SIZE = 35 };
37 36
38 #define MIN_AUDIO_BW_MANAGEMENT_BITRATE 6 37 #define MIN_AUDIO_BW_MANAGEMENT_BITRATE 6
39 #define MIN_VIDEO_BW_MANAGEMENT_BITRATE 30 38 #define MIN_VIDEO_BW_MANAGEMENT_BITRATE 30
40 39
41 enum { RTP_MAX_BURST_SLEEP_TIME = 500 }; 40 enum { RTP_MAX_BURST_SLEEP_TIME = 500 };
42 enum { RTP_AUDIO_LEVEL_UNIQUE_ID = 0xbede }; 41 enum { RTP_AUDIO_LEVEL_UNIQUE_ID = 0xbede };
43 enum { RTP_MAX_PACKETS_PER_FRAME = 512 }; // must be multiple of 32 42 enum { RTP_MAX_PACKETS_PER_FRAME = 512 }; // must be multiple of 32
44 } // namespace webrtc 43 } // namespace webrtc
45 44
46 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_CONFIG_H_ 45 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_CONFIG_H_
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