Update a ton of audio code to use size_t more correctly and in general reduce

webrtc/modules/audio_processing/agc/agc_manager_direct.cc

 /*
  *  Copyright (c) 2013 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 77 matching lines @@
 class DebugFile {
 #ifdef WEBRTC_AGC_DEBUG_DUMP
  public:
   explicit DebugFile(const char* filename)
       : file_(fopen(filename, "wb")) {
     assert(file_);
   }
   ~DebugFile() {
     fclose(file_);
   }
-  void Write(const int16_t* data, int length_samples) {
+  void Write(const int16_t* data, size_t length_samples) {
     fwrite(data, 1, length_samples * sizeof(int16_t), file_);
   }
  private:
   FILE* file_;
 #else
  public:
   explicit DebugFile(const char* filename) {
   }
   ~DebugFile() {
   }
-  void Write(const int16_t* data, int length_samples) {
+  void Write(const int16_t* data, size_t length_samples) {
   }
 #endif  // WEBRTC_AGC_DEBUG_DUMP
 };
 
 AgcManagerDirect::AgcManagerDirect(GainControl* gctrl,
                                    VolumeCallbacks* volume_callbacks,
                                    int startup_min_level)
     : agc_(new Agc()),
       gctrl_(gctrl),
       volume_callbacks_(volume_callbacks),
@@ skipping 61 matching lines @@
   }
   if (gctrl_->enable_limiter(true) != 0) {
     LOG_FERR1(LS_ERROR, enable_limiter, true);
     return -1;
   }
   return 0;
 }
 
 void AgcManagerDirect::AnalyzePreProcess(int16_t* audio,
                                          int num_channels,
-                                         int samples_per_channel) {
-  int length = num_channels * samples_per_channel;
+                                         size_t samples_per_channel) {
+  size_t length = num_channels * samples_per_channel;
   if (capture_muted_) {
     return;
   }
 
   file_preproc_->Write(audio, length);
 
   if (frames_since_clipped_ < kClippedWaitFrames) {
     ++frames_since_clipped_;
     return;
   }
@@ skipping 20 matching lines @@
       // will still not react until the postproc updates the level.
       SetLevel(std::max(kClippedLevelMin, level_ - kClippedLevelStep));
       // Reset the AGC since the level has changed.
       agc_->Reset();
     }
     frames_since_clipped_ = 0;
   }
 }
 
 void AgcManagerDirect::Process(const int16_t* audio,
-                               int length,
+                               size_t length,
                                int sample_rate_hz) {
   if (capture_muted_) {
     return;
   }
 
   if (check_volume_on_next_process_) {
     check_volume_on_next_process_ = false;
     // We have to wait until the first process call to check the volume,
     // because Chromium doesn't guarantee it to be valid any earlier.
     CheckVolumeAndReset();
@@ skipping 189 matching lines @@
   if (new_compression != compression_) {
     compression_ = new_compression;
     compression_accumulator_ = new_compression;
     if (gctrl_->set_compression_gain_db(compression_) != 0) {
       LOG_FERR1(LS_ERROR, set_compression_gain_db, compression_);
     }
   }
 }
 
 }  // namespace webrtc