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Side by Side Diff: webrtc/base/virtualsocketserver.h

Issue 2325623002: webrtc/base: Use RTC_DCHECK() instead of assert() (Closed)
Patch Set: Created 4 years, 3 months ago
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1 /* 1 /*
2 * Copyright 2004 The WebRTC Project Authors. All rights reserved. 2 * Copyright 2004 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_BASE_VIRTUALSOCKETSERVER_H_ 11 #ifndef WEBRTC_BASE_VIRTUALSOCKETSERVER_H_
12 #define WEBRTC_BASE_VIRTUALSOCKETSERVER_H_ 12 #define WEBRTC_BASE_VIRTUALSOCKETSERVER_H_
13 13
14 #include <assert.h>
15
16 #include <deque> 14 #include <deque>
17 #include <map> 15 #include <map>
18 16
17 #include "webrtc/base/checks.h"
19 #include "webrtc/base/constructormagic.h" 18 #include "webrtc/base/constructormagic.h"
20 #include "webrtc/base/messagequeue.h" 19 #include "webrtc/base/messagequeue.h"
21 #include "webrtc/base/socketserver.h" 20 #include "webrtc/base/socketserver.h"
22 21
23 namespace rtc { 22 namespace rtc {
24 23
25 class Packet; 24 class Packet;
26 class VirtualSocket; 25 class VirtualSocket;
27 class SocketAddressPair; 26 class SocketAddressPair;
28 27
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79 } 78 }
80 79
81 // If the (transit) delay parameters are modified, this method should be 80 // If the (transit) delay parameters are modified, this method should be
82 // called to recompute the new distribution. 81 // called to recompute the new distribution.
83 void UpdateDelayDistribution(); 82 void UpdateDelayDistribution();
84 83
85 // Controls the (uniform) probability that any sent packet is dropped. This 84 // Controls the (uniform) probability that any sent packet is dropped. This
86 // is separate from calculations to drop based on queue size. 85 // is separate from calculations to drop based on queue size.
87 double drop_probability() { return drop_prob_; } 86 double drop_probability() { return drop_prob_; }
88 void set_drop_probability(double drop_prob) { 87 void set_drop_probability(double drop_prob) {
89 assert((0 <= drop_prob) && (drop_prob <= 1)); 88 RTC_DCHECK_GE(drop_prob, 0.0);
89 RTC_DCHECK_LE(drop_prob, 1.0);
90 drop_prob_ = drop_prob; 90 drop_prob_ = drop_prob;
91 } 91 }
92 92
93 // If |blocked| is true, subsequent attempts to send will result in -1 being 93 // If |blocked| is true, subsequent attempts to send will result in -1 being
94 // returned, with the socket error set to EWOULDBLOCK. 94 // returned, with the socket error set to EWOULDBLOCK.
95 // 95 //
96 // If this method is later called with |blocked| set to false, any sockets 96 // If this method is later called with |blocked| set to false, any sockets
97 // that previously failed to send with EWOULDBLOCK will emit SignalWriteEvent. 97 // that previously failed to send with EWOULDBLOCK will emit SignalWriteEvent.
98 // 98 //
99 // This can be used to simulate the send buffer on a network interface being 99 // This can be used to simulate the send buffer on a network interface being
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375 375
376 // Store the options that are set 376 // Store the options that are set
377 OptionsMap options_map_; 377 OptionsMap options_map_;
378 378
379 friend class VirtualSocketServer; 379 friend class VirtualSocketServer;
380 }; 380 };
381 381
382 } // namespace rtc 382 } // namespace rtc
383 383
384 #endif // WEBRTC_BASE_VIRTUALSOCKETSERVER_H_ 384 #endif // WEBRTC_BASE_VIRTUALSOCKETSERVER_H_
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