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

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 455 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_);
   if (!src || !dest) {
     return kNullPointerError;
   }
 
@@ skipping 171 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 sample_rate_hz,
                                               ChannelLayout layout) {
   CriticalSectionScoped crit_scoped(crit_);
   if (data == NULL) {
     return kNullPointerError;
   }
 
   const int num_channels = ChannelsFromLayout(layout);
   RETURN_ON_ERR(MaybeInitializeLocked(fwd_in_format_.rate(),
                                       fwd_out_format_.rate(),
@@ skipping 427 matching lines @@
   int err = WriteMessageToDebugFile();
   if (err != kNoError) {
     return err;
   }
 
   return kNoError;
 }
 #endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
 
 }  // namespace webrtc