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

webrtc/modules/audio_processing/vad/vad_audio_proc.h

 /*
  *  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.
  */
 
@@ skipping 32 matching lines @@
                            size_t subframe_index);
   void GetLpcPolynomials(double* lpc, size_t length_lpc);
   void FindFirstSpectralPeaks(double* f_peak, size_t length_f_peak);
   void Rms(double* rms, size_t length_rms);
   void ResetBuffer();
 
   // To compute spectral peak we perform LPC analysis to get spectral envelope.
   // For every 30 ms we compute 3 spectral peak there for 3 LPC analysis.
   // LPC is computed over 15 ms of windowed audio. For every 10 ms sub-frame
   // we need 5 ms of past signal to create the input of LPC analysis.
-  static const size_t kNumPastSignalSamples =
-      static_cast<size_t>(kSampleRateHz / 200);
+  enum : size_t {
+    kNumPastSignalSamples = static_cast<size_t>(kSampleRateHz / 200)
+  };
 
   // TODO(turajs): maybe defining this at a higher level (maybe enum) so that
   // all the code recognize it as "no-error."
-  static const int kNoError = 0;
+  enum : int { kNoError = 0 };
 
-  static const size_t kNum10msSubframes = 3;
-  static const size_t kNumSubframeSamples =
-      static_cast<size_t>(kSampleRateHz / 100);
-  static const size_t kNumSamplesToProcess =
-      kNum10msSubframes *
-      kNumSubframeSamples;  // Samples in 30 ms @ given sampling rate.
-  static const size_t kBufferLength =
-      kNumPastSignalSamples + kNumSamplesToProcess;
-  static const size_t kIpLength = kDftSize >> 1;
-  static const size_t kWLength = kDftSize >> 1;
-
-  static const size_t kLpcOrder = 16;
+  enum : size_t { kNum10msSubframes = 3 };
+  enum : size_t {
+    kNumSubframeSamples = static_cast<size_t>(kSampleRateHz / 100)
+  };
+  enum : size_t {
+    // Samples in 30 ms @ given sampling rate.
+    kNumSamplesToProcess = kNum10msSubframes * kNumSubframeSamples
+  };
+  enum : size_t {
+    kBufferLength = kNumPastSignalSamples + kNumSamplesToProcess
+  };
+  enum : size_t { kIpLength = kDftSize >> 1 };
+  enum : size_t { kWLength = kDftSize >> 1 };
+  enum : size_t { kLpcOrder = 16 };
 
   size_t ip_[kIpLength];
   float w_fft_[kWLength];
 
   // A buffer of 5 ms (past audio) + 30 ms (one iSAC frame ).
   float audio_buffer_[kBufferLength];
   size_t num_buffer_samples_;
 
   double log_old_gain_;
   double old_lag_;
 
   std::unique_ptr<PitchAnalysisStruct> pitch_analysis_handle_;
   std::unique_ptr<PreFiltBankstr> pre_filter_handle_;
   std::unique_ptr<PoleZeroFilter> high_pass_filter_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_VAD_VAD_AUDIO_PROC_H_