commit 665174fdbb4e0540eccb27cf7412348f1b65534c
author Yves Gerey <yvesg@webrtc.org> Tue Jun 19 15:03:05 2018 +0200
committer Yves Gerey <yvesg@webrtc.org> Tue Jun 19 14:00:39 2018 +0000
tree 34a9eb770e6de54c282f62ad0b65d2cd69c917f6
parent b602123a5aae1a9f0b6b581e611acf44eb525c08

Reformat the WebRTC code base

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}

modules/audio_processing/gain_control_impl.cc

diff --git a/modules/audio_processing/gain_control_impl.cc b/modules/audio_processing/gain_control_impl.cc
index 8be261c..685a27f 100644
--- a/modules/audio_processing/gain_control_impl.cc
+++ b/modules/audio_processing/gain_control_impl.cc
@@ -67,9 +67,7 @@
     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 @@
   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 @@
   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 @@
   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;