Convert channel counts to size_t.

webrtc/modules/audio_processing/noise_suppression_impl.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.
  */
 
@@ skipping 34 matching lines @@
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(Suppressor);
 };
 
 NoiseSuppressionImpl::NoiseSuppressionImpl(rtc::CriticalSection* crit)
     : crit_(crit) {
   RTC_DCHECK(crit);
 }
 
 NoiseSuppressionImpl::~NoiseSuppressionImpl() {}
 
-void NoiseSuppressionImpl::Initialize(int channels, int sample_rate_hz) {
-  RTC_DCHECK_LE(0, channels);
+void NoiseSuppressionImpl::Initialize(size_t channels, int sample_rate_hz) {
   rtc::CritScope cs(crit_);
   channels_ = channels;
   sample_rate_hz_ = sample_rate_hz;
   std::vector<rtc::scoped_ptr<Suppressor>> new_suppressors;
   if (enabled_) {
     new_suppressors.resize(channels);
-    for (int i = 0; i < channels; i++) {
+    for (size_t i = 0; i < channels; i++) {
       new_suppressors[i].reset(new Suppressor(sample_rate_hz));
     }
   }
   suppressors_.swap(new_suppressors);
   set_level(level_);
 }
 
 void NoiseSuppressionImpl::AnalyzeCaptureAudio(AudioBuffer* audio) {
   RTC_DCHECK(audio);
 #if defined(WEBRTC_NS_FLOAT)
   rtc::CritScope cs(crit_);
   if (!enabled_) {
     return;
   }
 
   RTC_DCHECK_GE(160u, audio->num_frames_per_band());
-  RTC_DCHECK_EQ(suppressors_.size(),
-                static_cast<size_t>(audio->num_channels()));
+  RTC_DCHECK_EQ(suppressors_.size(), audio->num_channels());
   for (size_t i = 0; i < suppressors_.size(); i++) {
     WebRtcNs_Analyze(suppressors_[i]->state(),
                      audio->split_bands_const_f(i)[kBand0To8kHz]);
   }
 #endif
 }
 
 void NoiseSuppressionImpl::ProcessCaptureAudio(AudioBuffer* audio) {
   RTC_DCHECK(audio);
   rtc::CritScope cs(crit_);
   if (!enabled_) {
     return;
   }
 
   RTC_DCHECK_GE(160u, audio->num_frames_per_band());
-  RTC_DCHECK_EQ(suppressors_.size(),
-                static_cast<size_t>(audio->num_channels()));
+  RTC_DCHECK_EQ(suppressors_.size(), audio->num_channels());
   for (size_t i = 0; i < suppressors_.size(); i++) {
 #if defined(WEBRTC_NS_FLOAT)
     WebRtcNs_Process(suppressors_[i]->state(),
                      audio->split_bands_const_f(i),
                      audio->num_bands(),
                      audio->split_bands_f(i));
 #elif defined(WEBRTC_NS_FIXED)
     WebRtcNsx_Process(suppressors_[i]->state(),
                       audio->split_bands_const(i),
                       audio->num_bands(),
@@ skipping 60 matching lines @@
     probability_average /= suppressors_.size();
   }
   return probability_average;
 #elif defined(WEBRTC_NS_FIXED)
   // TODO(peah): Returning error code as a float! Remove this.
   // Currently not available for the fixed point implementation.
   return AudioProcessing::kUnsupportedFunctionError;
 #endif
 }
 }  // namespace webrtc