webrtc/modules/audio_processing: Use RTC_DCHECK() instead of assert()
Review-Url: https://codereview.webrtc.org/2320053003
Cr-Commit-Position: refs/heads/master@{#14211}
diff --git a/webrtc/modules/audio_processing/gain_control_impl.cc b/webrtc/modules/audio_processing/gain_control_impl.cc
index 6bb1d20..aa4316d 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::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 @@
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 =