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

webrtc/modules/audio_processing/audio_buffer.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.
  */
 
 #include "webrtc/modules/audio_processing/audio_buffer.h"
 
 #include "webrtc/common_audio/include/audio_util.h"
 #include "webrtc/common_audio/resampler/push_sinc_resampler.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/common_audio/channel_buffer.h"
 #include "webrtc/modules/audio_processing/common.h"
 
 namespace webrtc {
 namespace {
 
-const int kSamplesPer16kHzChannel = 160;
-const int kSamplesPer32kHzChannel = 320;
-const int kSamplesPer48kHzChannel = 480;
+const size_t kSamplesPer16kHzChannel = 160;
+const size_t kSamplesPer32kHzChannel = 320;
+const size_t kSamplesPer48kHzChannel = 480;
 
 int KeyboardChannelIndex(const StreamConfig& stream_config) {
   if (!stream_config.has_keyboard()) {
     assert(false);
     return -1;
   }
 
   return stream_config.num_channels();
 }
 
-int NumBandsFromSamplesPerChannel(int num_frames) {
-  int num_bands = 1;
+size_t NumBandsFromSamplesPerChannel(size_t num_frames) {
+  size_t num_bands = 1;
   if (num_frames == kSamplesPer32kHzChannel ||
       num_frames == kSamplesPer48kHzChannel) {
-    num_bands = rtc::CheckedDivExact(num_frames,
-                                     static_cast<int>(kSamplesPer16kHzChannel));
+    num_bands = rtc::CheckedDivExact(num_frames, kSamplesPer16kHzChannel);
   }
   return num_bands;
 }
 
 }  // namespace
 
-AudioBuffer::AudioBuffer(int input_num_frames,
+AudioBuffer::AudioBuffer(size_t input_num_frames,
                          int num_input_channels,
-                         int process_num_frames,
+                         size_t process_num_frames,
                          int num_process_channels,
-                         int output_num_frames)
+                         size_t output_num_frames)
   : input_num_frames_(input_num_frames),
     num_input_channels_(num_input_channels),
     proc_num_frames_(process_num_frames),
     num_proc_channels_(num_process_channels),
     output_num_frames_(output_num_frames),
     num_channels_(num_process_channels),
     num_bands_(NumBandsFromSamplesPerChannel(proc_num_frames_)),
     num_split_frames_(rtc::CheckedDivExact(proc_num_frames_, num_bands_)),
     mixed_low_pass_valid_(false),
     reference_copied_(false),
@@ skipping 276 matching lines @@
 }
 
 int AudioBuffer::num_channels() const {
   return num_channels_;
 }
 
 void AudioBuffer::set_num_channels(int num_channels) {
   num_channels_ = num_channels;
 }
 
-int AudioBuffer::num_frames() const {
+size_t AudioBuffer::num_frames() const {
   return proc_num_frames_;
 }
 
-int AudioBuffer::num_frames_per_band() const {
+size_t AudioBuffer::num_frames_per_band() const {
   return num_split_frames_;
 }
 
-int AudioBuffer::num_keyboard_frames() const {
+size_t AudioBuffer::num_keyboard_frames() const {
   // We don't resample the keyboard channel.
   return input_num_frames_;
 }
 
-int AudioBuffer::num_bands() const {
+size_t AudioBuffer::num_bands() const {
   return num_bands_;
 }
 
 // The resampler is only for supporting 48kHz to 16kHz in the reverse stream.
 void AudioBuffer::DeinterleaveFrom(AudioFrame* frame) {
   assert(frame->num_channels_ == num_input_channels_);
   assert(frame->samples_per_channel_ == input_num_frames_);
   InitForNewData();
   // Initialized lazily because there's a different condition in CopyFrom.
   if ((input_num_frames_ != proc_num_frames_) && !input_buffer_) {
@@ skipping 66 matching lines @@
 
 void AudioBuffer::SplitIntoFrequencyBands() {
   splitting_filter_->Analysis(data_.get(), split_data_.get());
 }
 
 void AudioBuffer::MergeFrequencyBands() {
   splitting_filter_->Synthesis(split_data_.get(), data_.get());
 }
 
 }  // namespace webrtc