Update audio code to use size_t more correctly, webrtc/ portion.

webrtc/tools/agc/agc_manager.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 48 matching lines @@
   MediaCallback(AgcManagerDirect* direct, AudioProcessing* audioproc,
                 CriticalSectionWrapper* crit)
       : direct_(direct),
         audioproc_(audioproc),
         crit_(crit),
         frame_() {
   }
 
  protected:
   virtual void Process(const int channel, const ProcessingTypes type,
-                       int16_t audio[], const int samples_per_channel,
+                       int16_t audio[], const size_t samples_per_channel,
                        const int sample_rate_hz, const bool is_stereo) {
     CriticalSectionScoped cs(crit_);
     if (direct_->capture_muted()) {
       return;
     }
 
     // Extract the first channel.
     const int kMaxSampleRateHz = 48000;
     const int kMaxSamplesPerChannel = kMaxSampleRateHz / 100;
     assert(samples_per_channel < kMaxSamplesPerChannel &&
            sample_rate_hz < kMaxSampleRateHz);
     int16_t mono[kMaxSamplesPerChannel];
     int16_t* mono_ptr = audio;
     if (is_stereo) {
-      for (int n = 0; n < samples_per_channel; n++) {
+      for (size_t n = 0; n < samples_per_channel; n++) {
         mono[n] = audio[n * 2];
       }
       mono_ptr = mono;
     }
 
     direct_->Process(mono_ptr, samples_per_channel, sample_rate_hz);
 
     // TODO(ajm): It's unfortunate we have to memcpy to this frame here, but
     // it's needed for use with AudioProcessing.
     frame_.num_channels_ = is_stereo ? 2 : 1;
     frame_.samples_per_channel_ = samples_per_channel;
     frame_.sample_rate_hz_ = sample_rate_hz;
-    const int length_samples = frame_.num_channels_ * samples_per_channel;
+    const size_t length_samples = frame_.num_channels_ * samples_per_channel;
     memcpy(frame_.data_, audio, length_samples * sizeof(int16_t));
 
     // Apply compression to the audio.
     if (audioproc_->ProcessStream(&frame_) != 0) {
       LOG_FERR0(LS_ERROR, ProcessStream);
     }
 
     // Copy the compressed audio back to voice engine's array.
     memcpy(audio, frame_.data_, length_samples * sizeof(int16_t));
   }
 
  private:
   AgcManagerDirect* direct_;
   AudioProcessing* audioproc_;
   CriticalSectionWrapper* crit_;
   AudioFrame frame_;
 };
 
 class PreprocCallback : public VoEMediaProcess {
  public:
   PreprocCallback(AgcManagerDirect* direct, CriticalSectionWrapper* crit)
       : direct_(direct),
         crit_(crit) {
   }
 
  protected:
   virtual void Process(const int channel, const ProcessingTypes type,
-                       int16_t audio[], const int samples_per_channel,
+                       int16_t audio[], const size_t samples_per_channel,
                        const int sample_rate_hz, const bool is_stereo) {
     CriticalSectionScoped cs(crit_);
     if (direct_->capture_muted()) {
       return;
     }
     direct_->AnalyzePreProcess(audio, is_stereo ? 2 : 1, samples_per_channel);
   }
 
  private:
   AgcManagerDirect* direct_;
@@ skipping 110 matching lines @@
   }
   if (media_->DeRegisterExternalMediaProcessing(0,
           kRecordingPreprocessing) != 0) {
     LOG(LS_ERROR) << "Failed to deregister preproc callback";
     err = -1;
   }
   return err;
 }
 
 }  // namespace webrtc