Update audio code to use size_t more correctly, webrtc/modules/audio_processing/

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_(Histogram::Create(kNumAnalysisFrames)),
       inactive_histogram_(Histogram::Create()) {
   }
 
 Agc::~Agc() {}
 
-float Agc::AnalyzePreproc(const int16_t* audio, int length) {
+float Agc::AnalyzePreproc(const int16_t* audio, size_t length) {
   assert(length > 0);
-  int num_clipped = 0;
-  for (int i = 0; i < length; ++i) {
+  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, int length, int sample_rate_hz) {
+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();
   DCHECK_EQ(rms.size(), probabilities.size());
   for (size_t i = 0; i < rms.size(); ++i) {
     histogram_->Update(rms[i], probabilities[i]);
   }
   return 0;
 }
@@ skipping 29 matching lines @@
   // limits. The upper limit should be chosen such that the risk of clipping is
   // low. The lower limit should not result in a too quiet signal.
   if (level >= 0 || level <= -100)
     return -1;
   target_level_dbfs_ = level;
   target_level_loudness_ = Dbfs2Loudness(level);
   return 0;
 }
 
 }  // namespace webrtc