Test RTC_DCHECK_IS_ON instead of checking DCHECK_ALWAYS_ON everywhere

webrtc/common_audio/resampler/push_resampler.cc

 /*
  *  Copyright (c) 2013 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 11 matching lines @@
 // These checks were factored out into a non-templatized function
 // due to problems with clang on Windows in debug builds.
 // For some reason having the DCHECKs inline in the template code
 // caused the compiler to generate code that threw off the linker.
 // TODO(tommi): Re-enable when we've figured out what the problem is.
 // http://crbug.com/615050
 void CheckValidInitParams(int src_sample_rate_hz, int dst_sample_rate_hz,
                           size_t num_channels) {
 // The below checks are temporarily disabled on WEBRTC_WIN due to problems
 // with clang debug builds.
-#if !defined(WEBRTC_WIN) && defined(__clang__) && !defined(NDEBUG)
+#if !defined(WEBRTC_WIN) && defined(__clang__)
   RTC_DCHECK_GT(src_sample_rate_hz, 0);
   RTC_DCHECK_GT(dst_sample_rate_hz, 0);
   RTC_DCHECK_GT(num_channels, 0u);
   RTC_DCHECK_LE(num_channels, 2u);
 #endif
 }
 
 void CheckExpectedBufferSizes(size_t src_length,
                               size_t dst_capacity,
                               size_t num_channels,
                               int src_sample_rate,
                               int dst_sample_rate) {
 // The below checks are temporarily disabled on WEBRTC_WIN due to problems
 // with clang debug builds.
 // TODO(tommi): Re-enable when we've figured out what the problem is.
 // http://crbug.com/615050
-#if !defined(WEBRTC_WIN) && defined(__clang__) && !defined(NDEBUG)
+#if !defined(WEBRTC_WIN) && defined(__clang__)
   const size_t src_size_10ms = src_sample_rate * num_channels / 100;
   const size_t dst_size_10ms = dst_sample_rate * num_channels / 100;
-  RTC_CHECK_EQ(src_length, src_size_10ms);
-  RTC_CHECK_GE(dst_capacity, dst_size_10ms);
+  RTC_DCHECK_EQ(src_length, src_size_10ms);
+  RTC_DCHECK_GE(dst_capacity, dst_size_10ms);
 #endif
 }
 }
 
 template <typename T>
 PushResampler<T>::PushResampler()
     : src_sample_rate_hz_(0),
       dst_sample_rate_hz_(0),
       num_channels_(0) {
 }
@@ skipping 74 matching lines @@
     return static_cast<int>(
         sinc_resampler_->Resample(src, src_length, dst, dst_capacity));
   }
 }
 
 // Explictly generate required instantiations.
 template class PushResampler<int16_t>;
 template class PushResampler<float>;
 
 }  // namespace webrtc