[NOT FOR REVIEW] 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 27 matching lines @@
   // |analysis_rate| sets the number of input blocks (containing speech!)
   // to elapse before a new gain computation is made. |variance_rate| specifies
   // the number of gain recomputations after which the variances are reset.
   // |cv_*| are parameters for the VarianceArray constructor for the
   // clear speech stream.
   // TODO(bercic): the |cv_*|, |*_rate| and |gain_limit| parameters should
   // probably go away once fine tuning is done. They override the internal
   // constants in the class (kGainChangeLimit, kAnalyzeRate, kVarianceRate).
   IntelligibilityEnhancer(size_t erb_resolution,
                           int sample_rate_hz,
-                          int channels,
+                          size_t channels,
                           int cv_type,
                           float cv_alpha,
                           size_t cv_win,
                           int analysis_rate,
                           int variance_rate,
                           float gain_limit);
   ~IntelligibilityEnhancer();
 
   // Reads and processes chunk of noise stream in time domain.
   void ProcessCaptureAudio(float* const* audio);
 
   // Reads chunk of speech in time domain and updates with modified signal.
   void ProcessRenderAudio(float* const* audio);
 
  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 channels_;       // Num channels.
+  const size_t channels_;    // Num channels.
   const int analysis_rate_;  // Num blocks before gains recalculated.
   const int variance_rate_;  // Num recalculations before history is cleared.
 
   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_;
   rtc::scoped_ptr<float[]> center_freqs_;
   size_t start_freq_;
@@ skipping 21 matching lines @@
   // Note: VAD currently does not affect anything in IntelligibilityEnhancer.
   VadInst* vad_high_;
   VadInst* vad_low_;
   rtc::scoped_ptr<int16_t[]> vad_tmp_buffer_;
   bool has_voice_low_;  // Whether voice detected in speech stream.
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_INTELLIGIBILITY_INTELLIGIBILITY_ENHANCER_H_