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

webrtc/modules/audio_processing/gain_control_impl.cc

Index: webrtc/modules/audio_processing/gain_control_impl.cc
diff --git a/webrtc/modules/audio_processing/gain_control_impl.cc b/webrtc/modules/audio_processing/gain_control_impl.cc
index 6bb1d2029b626d84e7388a648aca4b3ff1107fe5..aa4316de28b67f1847c2d28e1222dddab9dba6de 100644
--- a/webrtc/modules/audio_processing/gain_control_impl.cc
+++ b/webrtc/modules/audio_processing/gain_control_impl.cc
@@ -296,7 +296,7 @@ int GainControlImpl::set_stream_analog_level(int level) {
 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_;
 }
@@ -482,7 +482,7 @@ int GainControlImpl::Configure() {
   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 =