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

webrtc/modules/audio_processing/agc/agc.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.
  */
 
@@ skipping 21 matching lines @@
 
 Agc::Agc()
     : target_level_loudness_(Dbfs2Loudness(kDefaultLevelDbfs)),
       target_level_dbfs_(kDefaultLevelDbfs),
       histogram_(LoudnessHistogram::Create(kNumAnalysisFrames)),
       inactive_histogram_(LoudnessHistogram::Create()) {}
 
 Agc::~Agc() {}
 
 float Agc::AnalyzePreproc(const int16_t* audio, size_t length) {
-  assert(length > 0);
+  RTC_DCHECK_GT(length, 0u);
   size_t num_clipped = 0;
   for (size_t i = 0; i < length; ++i) {
     if (audio[i] == 32767 || audio[i] == -32768)
       ++num_clipped;
   }
   return 1.0f * num_clipped / length;
 }
 
 int Agc::Process(const int16_t* audio, size_t length, int sample_rate_hz) {
   vad_.ProcessChunk(audio, length, sample_rate_hz);
   const std::vector<double>& rms = vad_.chunkwise_rms();
   const std::vector<double>& probabilities =
       vad_.chunkwise_voice_probabilities();
   RTC_DCHECK_EQ(rms.size(), probabilities.size());
   for (size_t i = 0; i < rms.size(); ++i) {
     histogram_->Update(rms[i], probabilities[i]);
   }
   return 0;
 }
 
 bool Agc::GetRmsErrorDb(int* error) {
   if (!error) {
-    assert(false);
+    RTC_NOTREACHED();
     return false;
   }
 
   if (histogram_->num_updates() < kNumAnalysisFrames) {
     // We haven't yet received enough frames.
     return false;
   }
 
   if (histogram_->AudioContent() < kNumAnalysisFrames * kActivityThreshold) {
     // We are likely in an inactive segment.
@@ skipping 23 matching lines @@
 
 int Agc::target_level_dbfs() const {
   return target_level_dbfs_;
 }
 
 float Agc::voice_probability() const {
   return vad_.last_voice_probability();
 }
 
 }  // namespace webrtc