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Reformat the WebRTC code base

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Running clang-format with chromium's style guide.

The goal is n-fold:
 * providing consistency and readability (that's what code guidelines are for)
 * preventing noise with presubmit checks and git cl format
 * building on the previous point: making it easier to automatically fix format issues
 * you name it

Please consider using git-hyper-blame to ignore this commit.

Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
This commit is contained in:
Yves Gerey
2018-06-19 15:03:05 +02:00
parent b602123a5a
commit 665174fdbb
1569 changed files with 30495 additions and 30309 deletions
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16
modules/audio_processing/gain_control_impl.cc
View File

@ -67,9 +67,7 @@ class GainControlImpl::GainController {
set_capture_level(capture_level);
}
void set_capture_level(int capture_level) {
capture_level_ = capture_level;
}
void set_capture_level(int capture_level) { capture_level_ = capture_level; }
int get_capture_level() {
RTC_DCHECK(capture_level_);
@ -259,7 +257,7 @@ int GainControlImpl::stream_analog_level() {
data_dumper_->DumpRaw("gain_control_stream_analog_level", 1,
&analog_capture_level_);
// TODO(ajm): enable this assertion?
//RTC_DCHECK_EQ(kAdaptiveAnalog, mode_);
// RTC_DCHECK_EQ(kAdaptiveAnalog, mode_);
return analog_capture_level_;
}
@ -303,8 +301,7 @@ GainControl::Mode GainControlImpl::mode() const {
return mode_;
}
int GainControlImpl::set_analog_level_limits(int minimum,
int maximum) {
int GainControlImpl::set_analog_level_limits(int minimum, int maximum) {
if (minimum < 0) {
return AudioProcessing::kBadParameterError;
}
@ -419,11 +416,10 @@ int GainControlImpl::Configure() {
WebRtcAgcConfig config;
// TODO(ajm): Flip the sign here (since AGC expects a positive value) if we
// change the interface.
//RTC_DCHECK_LE(target_level_dbfs_, 0);
//config.targetLevelDbfs = static_cast<int16_t>(-target_level_dbfs_);
// 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.compressionGaindB = static_cast<int16_t>(compression_gain_db_);
config.limiterEnable = limiter_enabled_;
int error = AudioProcessing::kNoError;