Replace some asserts with DCHECKs

webrtc/modules/audio_device/dummy/file_audio_device.cc

 /*
  *  Copyright (c) 2014 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/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/platform_thread.h"
 #include "webrtc/modules/audio_device/dummy/file_audio_device.h"
 #include "webrtc/system_wrappers/include/sleep.h"
 
 namespace webrtc {
 
 const int kRecordingFixedSampleRate = 48000;
 const size_t kRecordingNumChannels = 2;
 const int kPlayoutFixedSampleRate = 48000;
@@ skipping 465 matching lines @@
     int64_t currentTime = rtc::TimeMillis();
     _critSect.Enter();
 
     if (_lastCallPlayoutMillis == 0 ||
         currentTime - _lastCallPlayoutMillis >= 10) {
         _critSect.Leave();
         _ptrAudioBuffer->RequestPlayoutData(_playoutFramesIn10MS);
         _critSect.Enter();
 
         _playoutFramesLeft = _ptrAudioBuffer->GetPlayoutData(_playoutBuffer);
-        assert(_playoutFramesLeft == _playoutFramesIn10MS);
+        RTC_DCHECK_EQ(_playoutFramesIn10MS, _playoutFramesLeft);
         if (_outputFile.is_open()) {
           _outputFile.Write(_playoutBuffer, kPlayoutBufferSize);
         }
         _lastCallPlayoutMillis = currentTime;
     }
     _playoutFramesLeft = 0;
     _critSect.Leave();
 
     int64_t deltaTimeMillis = rtc::TimeMillis() - currentTime;
     if (deltaTimeMillis < 10) {
@@ skipping 32 matching lines @@
 
     int64_t deltaTimeMillis = rtc::TimeMillis() - currentTime;
     if (deltaTimeMillis < 10) {
       SleepMs(10 - deltaTimeMillis);
     }
 
     return true;
 }
 
 }  // namespace webrtc