Replace ASSERT(false) by RTC_NOTREACHED().

webrtc/base/physicalsocketserver.cc

 /*
  *  Copyright 2004 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.
  */
 #include "webrtc/base/physicalsocketserver.h"
@@ skipping 24 matching lines @@
 #include <ws2tcpip.h>
 #undef SetPort
 #endif
 
 #include <algorithm>
 #include <map>
 
 #include "webrtc/base/arraysize.h"
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/byteorder.h"
+#include "webrtc/base/checks.h"
 #include "webrtc/base/common.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/networkmonitor.h"
 #include "webrtc/base/nullsocketserver.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/base/winping.h"
 #include "webrtc/base/win32socketinit.h"
 
 #if defined(WEBRTC_POSIX)
 #include <netinet/tcp.h>  // for TCP_NODELAY
@@ skipping 417 matching lines @@
                                            1, false);
     if (result == WinPing::PING_FAIL) {
       SetError(EINVAL);  // can't think of a better error ID
       return -1;
     } else if (result != WinPing::PING_TOO_LARGE) {
       *mtu = PACKET_MAXIMUMS[level];
       return 0;
     }
   }
 
-  ASSERT(false);
+  RTC_NOTREACHED();
   return -1;
 #elif defined(WEBRTC_MAC)
   // No simple way to do this on Mac OS X.
   // SIOCGIFMTU would work if we knew which interface would be used, but
   // figuring that out is pretty complicated. For now we'll return an error
   // and let the caller pick a default MTU.
   SetError(EINVAL);
   return -1;
 #elif defined(WEBRTC_LINUX)
   // Gets the path MTU.
@@ skipping 95 matching lines @@
     case OPT_NODELAY:
       *slevel = IPPROTO_TCP;
       *sopt = TCP_NODELAY;
       break;
     case OPT_DSCP:
       LOG(LS_WARNING) << "Socket::OPT_DSCP not supported.";
       return -1;
     case OPT_RTP_SENDTIME_EXTN_ID:
       return -1;  // No logging is necessary as this not a OS socket option.
     default:
-      ASSERT(false);
+      RTC_NOTREACHED();
       return -1;
   }
   return 0;
 }
 
 SocketDispatcher::SocketDispatcher(PhysicalSocketServer *ss)
 #if defined(WEBRTC_WIN)
   : PhysicalSocket(ss), id_(0), signal_close_(false)
 #else
   : PhysicalSocket(ss)
@@ skipping 238 matching lines @@
     // pipes.  This simulates it by resetting before the event is handled.
 
     CritScope cs(&crit_);
     if (fSignaled_) {
       uint8_t b[4];  // Allow for reading more than 1 byte, but expect 1.
       VERIFY(1 == read(afd_[0], b, sizeof(b)));
       fSignaled_ = false;
     }
   }
 
-  void OnEvent(uint32_t ff, int err) override { ASSERT(false); }
+  void OnEvent(uint32_t ff, int err) override { RTC_NOTREACHED(); }
 
   int GetDescriptor() override { return afd_[0]; }
 
   bool IsDescriptorClosed() override { return false; }
 
  private:
   PhysicalSocketServer *ss_;
   int afd_[2];
   bool fSignaled_;
   CriticalSection crit_;
@@ skipping 624 matching lines @@
     DWORD dw = WSAWaitForMultipleEvents(static_cast<DWORD>(events.size()),
                                         &events[0],
                                         false,
                                         static_cast<DWORD>(cmsNext),
                                         false);
 
     if (dw == WSA_WAIT_FAILED) {
       // Failed?
       // TODO(pthatcher): need a better strategy than this!
       WSAGetLastError();
-      ASSERT(false);
+      RTC_NOTREACHED();
       return false;
     } else if (dw == WSA_WAIT_TIMEOUT) {
       // Timeout?
       return true;
     } else {
       // Figure out which one it is and call it
       CritScope cr(&crit_);
       int index = dw - WSA_WAIT_EVENT_0;
       if (index > 0) {
         --index; // The first event is the socket event
@@ skipping 83 matching lines @@
        break;
     }
   }
 
   // Done
   return true;
 }
 #endif  // WEBRTC_WIN
 
 }  // namespace rtc