webrtc/modules/audio_processing: Use RTC_DCHECK() instead of assert()

webrtc/modules/audio_processing/gain_control_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 278 matching lines @@
     return AudioProcessing::kBadParameterError;
   }
   analog_capture_level_ = level;
 
   return AudioProcessing::kNoError;
 }
 
 int GainControlImpl::stream_analog_level() {
   rtc::CritScope cs(crit_capture_);
   // TODO(ajm): enable this assertion?
-  //assert(mode_ == kAdaptiveAnalog);
+  //RTC_DCHECK_EQ(kAdaptiveAnalog, mode_);
 
   return analog_capture_level_;
 }
 
 int GainControlImpl::Enable(bool enable) {
   rtc::CritScope cs_render(crit_render_);
   rtc::CritScope cs_capture(crit_capture_);
   if (enable && !enabled_) {
     enabled_ = enable;  // Must be set before Initialize() is called.
 
@@ skipping 165 matching lines @@
     render_signal_queue_->Clear();
   }
 }
 
 int GainControlImpl::Configure() {
   rtc::CritScope cs_render(crit_render_);
   rtc::CritScope cs_capture(crit_capture_);
   WebRtcAgcConfig config;
   // TODO(ajm): Flip the sign here (since AGC expects a positive value) if we
   //            change the interface.
-  //assert(target_level_dbfs_ <= 0);
+  //RTC_DCHECK_LE(target_level_dbfs_, 0);
   //config.targetLevelDbfs = static_cast<int16_t>(-target_level_dbfs_);
   config.targetLevelDbfs = static_cast<int16_t>(target_level_dbfs_);
   config.compressionGaindB =
       static_cast<int16_t>(compression_gain_db_);
   config.limiterEnable = limiter_enabled_;
 
   int error = AudioProcessing::kNoError;
   for (auto& gain_controller : gain_controllers_) {
     const int handle_error =
         WebRtcAgc_set_config(gain_controller->state(), config);
     if (handle_error != AudioProcessing::kNoError) {
       error = handle_error;
     }
   }
   return error;
 }
 }  // namespace webrtc