Convert channel counts to size_t.

webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h

 /*
  *  Copyright (c) 2014 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 29 matching lines @@
         : sample_rate_hz(16000),
           num_capture_channels(1),
           num_render_channels(1),
           var_type(intelligibility::VarianceArray::kStepDecaying),
           var_decay_rate(0.9f),
           var_window_size(10),
           analysis_rate(800),
           gain_change_limit(0.1f),
           rho(0.02f) {}
     int sample_rate_hz;
-    int num_capture_channels;
-    int num_render_channels;
+    size_t num_capture_channels;
+    size_t num_render_channels;
     intelligibility::VarianceArray::StepType var_type;
     float var_decay_rate;
     size_t var_window_size;
     int analysis_rate;
     float gain_change_limit;
     float rho;
   };
 
   explicit IntelligibilityEnhancer(const Config& config);
   IntelligibilityEnhancer();  // Initialize with default config.
 
   // Reads and processes chunk of noise stream in time domain.
   void AnalyzeCaptureAudio(float* const* audio,
                            int sample_rate_hz,
-                           int num_channels);
+                           size_t num_channels);
 
   // Reads chunk of speech in time domain and updates with modified signal.
   void ProcessRenderAudio(float* const* audio,
                           int sample_rate_hz,
-                          int num_channels);
+                          size_t num_channels);
   bool active() const;
 
  private:
   enum AudioSource {
     kRenderStream = 0,  // Clear speech stream.
     kCaptureStream,  // Noise stream.
   };
 
   // Provides access point to the frequency domain.
   class TransformCallback : public LappedTransform::Callback {
    public:
     TransformCallback(IntelligibilityEnhancer* parent, AudioSource source);
 
     // All in frequency domain, receives input |in_block|, applies
     // intelligibility enhancement, and writes result to |out_block|.
     void ProcessAudioBlock(const std::complex<float>* const* in_block,
-                           int in_channels,
+                           size_t in_channels,
                            size_t frames,
-                           int out_channels,
+                           size_t out_channels,
                            std::complex<float>* const* out_block) override;
 
    private:
     IntelligibilityEnhancer* parent_;
     AudioSource source_;
   };
   friend class TransformCallback;
   FRIEND_TEST_ALL_PREFIXES(IntelligibilityEnhancerTest, TestErbCreation);
   FRIEND_TEST_ALL_PREFIXES(IntelligibilityEnhancerTest, TestSolveForGains);
 
@@ skipping 36 matching lines @@
 
   // Returns dot product of vectors specified by size |length| arrays |a|,|b|.
   static float DotProduct(const float* a, const float* b, size_t length);
 
   const size_t freqs_;         // Num frequencies in frequency domain.
   const size_t window_size_;   // Window size in samples; also the block size.
   const size_t chunk_length_;  // Chunk size in samples.
   const size_t bank_size_;     // Num ERB filters.
   const int sample_rate_hz_;
   const int erb_resolution_;
-  const int num_capture_channels_;
-  const int num_render_channels_;
+  const size_t num_capture_channels_;
+  const size_t num_render_channels_;
   const int analysis_rate_;    // Num blocks before gains recalculated.
 
   const bool active_;          // Whether render gains are being updated.
                                // TODO(ekm): Add logic for updating |active_|.
 
   intelligibility::VarianceArray clear_variance_;
   intelligibility::VarianceArray noise_variance_;
   rtc::scoped_ptr<float[]> filtered_clear_var_;
   rtc::scoped_ptr<float[]> filtered_noise_var_;
   std::vector<std::vector<float>> filter_bank_;
@@ skipping 14 matching lines @@
   TransformCallback capture_callback_;
   rtc::scoped_ptr<LappedTransform> render_mangler_;
   rtc::scoped_ptr<LappedTransform> capture_mangler_;
   int block_count_;
   int analysis_step_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_INTELLIGIBILITY_INTELLIGIBILITY_ENHANCER_H_