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

webrtc/modules/audio_processing/gain_control_impl.cc

Index: webrtc/modules/audio_processing/gain_control_impl.cc
diff --git a/webrtc/modules/audio_processing/gain_control_impl.cc b/webrtc/modules/audio_processing/gain_control_impl.cc
index 398cf5c5fd71ead8c17d91c9e68de88ee1730c5e..8a3612dce5a7b7ea9c2b5a6c128329a3168b2da9 100644
--- a/webrtc/modules/audio_processing/gain_control_impl.cc
+++ b/webrtc/modules/audio_processing/gain_control_impl.cc
@@ -64,7 +64,7 @@ int GainControlImpl::ProcessRenderAudio(AudioBuffer* audio) {
     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);
@@ -92,7 +92,7 @@ int GainControlImpl::AnalyzeCaptureAudio(AudioBuffer* audio) {
           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);
@@ -108,7 +108,7 @@ int GainControlImpl::AnalyzeCaptureAudio(AudioBuffer* audio) {
           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);
 
@@ -146,7 +146,7 @@ int GainControlImpl::ProcessCaptureAudio(AudioBuffer* audio) {
         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,