webrtc/modules/audio_processing: Use RTC_DCHECK() instead of assert()

webrtc/modules/audio_processing/vad/standalone_vad.cc

 /*
  *  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.
  */
 
 #include "webrtc/modules/audio_processing/vad/standalone_vad.h"
 
-#include <assert.h>
-
+#include "webrtc/base/checks.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/utility/include/audio_frame_operations.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 static const int kDefaultStandaloneVadMode = 3;
 
 StandaloneVad::StandaloneVad(VadInst* vad)
     : vad_(vad), buffer_(), index_(0), mode_(kDefaultStandaloneVadMode) {
@@ skipping 32 matching lines @@
   return 0;
 }
 
 int StandaloneVad::GetActivity(double* p, size_t length_p) {
   if (index_ == 0)
     return -1;
 
   const size_t num_frames = index_ / kLength10Ms;
   if (num_frames > length_p)
     return -1;
-  assert(WebRtcVad_ValidRateAndFrameLength(kSampleRateHz, index_) == 0);
+  RTC_DCHECK_EQ(0, WebRtcVad_ValidRateAndFrameLength(kSampleRateHz, index_));
 
   int activity = WebRtcVad_Process(vad_, kSampleRateHz, buffer_, index_);
   if (activity < 0)
     return -1;
   else if (activity == 0)
     p[0] = 0.01;  // Arbitrary but small and non-zero.
   else
     p[0] = 0.5;  // 0.5 is neutral values when combinned by other probabilities.
   for (size_t n = 1; n < num_frames; n++)
     p[n] = p[0];
   // Reset the buffer to start from the beginning.
   index_ = 0;
   return activity;
 }
 
 int StandaloneVad::set_mode(int mode) {
   if (mode < 0 || mode > 3)
     return -1;
   if (WebRtcVad_set_mode(vad_, mode) != 0)
     return -1;
 
   mode_ = mode;
   return 0;
 }
 
 }  // namespace webrtc