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

webrtc/modules/audio_processing/gain_control_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 46 matching lines @@
     return apm_->kNoError;
   }
 
   assert(audio->num_frames_per_band() <= 160);
 
   for (int i = 0; i < num_handles(); i++) {
     Handle* my_handle = static_cast<Handle*>(handle(i));
     int err = WebRtcAgc_AddFarend(
         my_handle,
         audio->mixed_low_pass_data(),
-        static_cast<int16_t>(audio->num_frames_per_band()));
+        audio->num_frames_per_band());
 
     if (err != apm_->kNoError) {
       return GetHandleError(my_handle);
     }
   }
 
   return apm_->kNoError;
 }
 
 int GainControlImpl::AnalyzeCaptureAudio(AudioBuffer* audio) {
   if (!is_component_enabled()) {
     return apm_->kNoError;
   }
 
   assert(audio->num_frames_per_band() <= 160);
   assert(audio->num_channels() == num_handles());
 
   int err = apm_->kNoError;
 
   if (mode_ == kAdaptiveAnalog) {
     capture_levels_.assign(num_handles(), analog_capture_level_);
     for (int i = 0; i < num_handles(); i++) {
       Handle* my_handle = static_cast<Handle*>(handle(i));
       err = WebRtcAgc_AddMic(
           my_handle,
           audio->split_bands(i),
           audio->num_bands(),
-          static_cast<int16_t>(audio->num_frames_per_band()));
+          audio->num_frames_per_band());
 
       if (err != apm_->kNoError) {
         return GetHandleError(my_handle);
       }
     }
   } else if (mode_ == kAdaptiveDigital) {
 
     for (int i = 0; i < num_handles(); i++) {
       Handle* my_handle = static_cast<Handle*>(handle(i));
       int32_t capture_level_out = 0;
 
       err = WebRtcAgc_VirtualMic(
           my_handle,
           audio->split_bands(i),
           audio->num_bands(),
-          static_cast<int16_t>(audio->num_frames_per_band()),
+          audio->num_frames_per_band(),
           analog_capture_level_,
           &capture_level_out);
 
       capture_levels_[i] = capture_level_out;
 
       if (err != apm_->kNoError) {
         return GetHandleError(my_handle);
       }
 
     }
@@ skipping 17 matching lines @@
   stream_is_saturated_ = false;
   for (int i = 0; i < num_handles(); i++) {
     Handle* my_handle = static_cast<Handle*>(handle(i));
     int32_t capture_level_out = 0;
     uint8_t saturation_warning = 0;
 
     int err = WebRtcAgc_Process(
         my_handle,
         audio->split_bands_const(i),
         audio->num_bands(),
-        static_cast<int16_t>(audio->num_frames_per_band()),
+        audio->num_frames_per_band(),
         audio->split_bands(i),
         capture_levels_[i],
         &capture_level_out,
         apm_->echo_cancellation()->stream_has_echo(),
         &saturation_warning);
 
     if (err != apm_->kNoError) {
       return GetHandleError(my_handle);
     }
 
@@ skipping 174 matching lines @@
   return apm_->num_output_channels();
 }
 
 int GainControlImpl::GetHandleError(void* handle) const {
   // The AGC has no get_error() function.
   // (Despite listing errors in its interface...)
   assert(handle != NULL);
   return apm_->kUnspecifiedError;
 }
 }  // namespace webrtc