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

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Review-Url: https://codereview.webrtc.org/2320053003
Cr-Commit-Position: refs/heads/master@{#14211}
This commit is contained in:
kwiberg
2016-09-14 05:23:22 -07:00
committed by Commit bot
parent 3a7f35b1c4
commit 9e2be5f292
27 changed files with 159 additions and 149 deletions
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4
webrtc/modules/audio_processing/gain_control_impl.cc
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@ -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 =