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

webrtc/modules/audio_processing/audio_processing_impl.h

 /*
  *  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 36 matching lines @@
 }  // namespace audioproc
 #endif
 
 class AudioRate {
  public:
   explicit AudioRate(int sample_rate_hz) { set(sample_rate_hz); }
   virtual ~AudioRate() {}
 
   void set(int rate) {
     rate_ = rate;
-    samples_per_channel_ = AudioProcessing::kChunkSizeMs * rate_ / 1000;
+    samples_per_channel_ =
+        static_cast<size_t>(AudioProcessing::kChunkSizeMs * rate_ / 1000);
   }
 
   int rate() const { return rate_; }
-  int samples_per_channel() const { return samples_per_channel_; }
+  size_t samples_per_channel() const { return samples_per_channel_; }
 
  private:
   int rate_;
-  int samples_per_channel_;
+  size_t samples_per_channel_;
 };
 
 class AudioFormat : public AudioRate {
  public:
   AudioFormat(int sample_rate_hz, int num_channels)
       : AudioRate(sample_rate_hz),
         num_channels_(num_channels) {}
   virtual ~AudioFormat() {}
 
   void set(int rate, int num_channels) {
@@ skipping 29 matching lines @@
   int sample_rate_hz() const override;
   int proc_sample_rate_hz() const override;
   int proc_split_sample_rate_hz() const override;
   int num_input_channels() const override;
   int num_output_channels() const override;
   int num_reverse_channels() const override;
   void set_output_will_be_muted(bool muted) override;
   bool output_will_be_muted() const override;
   int ProcessStream(AudioFrame* frame) override;
   int 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) override;
   int AnalyzeReverseStream(AudioFrame* frame) override;
   int AnalyzeReverseStream(const float* const* data,
-                           int samples_per_channel,
+                           size_t samples_per_channel,
                            int sample_rate_hz,
                            ChannelLayout layout) override;
   int set_stream_delay_ms(int delay) override;
   int stream_delay_ms() const override;
   bool was_stream_delay_set() const override;
   void set_delay_offset_ms(int offset) override;
   int delay_offset_ms() const override;
   void set_stream_key_pressed(bool key_pressed) override;
   bool stream_key_pressed() const override;
   int StartDebugRecording(const char filename[kMaxFilenameSize]) override;
@@ skipping 94 matching lines @@
   const bool beamformer_enabled_;
   rtc::scoped_ptr<Beamformer<float>> beamformer_;
   const std::vector<Point> array_geometry_;
 
   const bool supports_48kHz_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_