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

webrtc/modules/audio_processing/audio_processing_impl.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 492 matching lines @@
     agc_manager_->SetCaptureMuted(output_will_be_muted_);
   }
 }
 
 bool AudioProcessingImpl::output_will_be_muted() const {
   CriticalSectionScoped lock(crit_);
   return output_will_be_muted_;
 }
 
 int AudioProcessingImpl::ProcessStream(const float* const* src,
-                                       int samples_per_channel,
+                                       size_t samples_per_channel,
                                        int input_sample_rate_hz,
                                        ChannelLayout input_layout,
                                        int output_sample_rate_hz,
                                        ChannelLayout output_layout,
                                        float* const* dest) {
   CriticalSectionScoped crit_scoped(crit_);
   StreamConfig input_stream = api_format_.input_stream();
   input_stream.set_sample_rate_hz(input_sample_rate_hz);
   input_stream.set_num_channels(ChannelsFromLayout(input_layout));
   input_stream.set_has_keyboard(LayoutHasKeyboard(input_layout));
@@ skipping 185 matching lines @@
   }
 
   // The level estimator operates on the recombined data.
   RETURN_ON_ERR(level_estimator_->ProcessStream(ca));
 
   was_stream_delay_set_ = false;
   return kNoError;
 }
 
 int AudioProcessingImpl::AnalyzeReverseStream(const float* const* data,
-                                              int samples_per_channel,
+                                              size_t samples_per_channel,
                                               int rev_sample_rate_hz,
                                               ChannelLayout layout) {
   const StreamConfig reverse_config = {
       rev_sample_rate_hz, ChannelsFromLayout(layout), LayoutHasKeyboard(layout),
   };
   if (samples_per_channel != reverse_config.num_frames()) {
     return kBadDataLengthError;
   }
   return AnalyzeReverseStream(data, reverse_config, reverse_config);
 }
@@ skipping 512 matching lines @@
   int err = WriteMessageToDebugFile();
   if (err != kNoError) {
     return err;
   }
 
   return kNoError;
 }
 #endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
 
 }  // namespace webrtc