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}

audio/audio_state.h

diff --git a/audio/audio_state.h b/audio/audio_state.h
index d4e4e3f..689534b 100644
--- a/audio/audio_state.h
+++ b/audio/audio_state.h
@@ -39,9 +39,7 @@
     RTC_DCHECK(config_.audio_processing);
     return config_.audio_processing.get();
   }
-  AudioTransport* audio_transport() override {
-    return &audio_transport_;
-  }
+  AudioTransport* audio_transport() override { return &audio_transport_; }
 
   void SetPlayout(bool enabled) override;
   void SetRecording(bool enabled) override;
@@ -60,7 +58,8 @@
   void RemoveReceivingStream(webrtc::AudioReceiveStream* stream);
 
   void AddSendingStream(webrtc::AudioSendStream* stream,
-                        int sample_rate_hz, size_t num_channels);
+                        int sample_rate_hz,
+                        size_t num_channels);
   void RemoveSendingStream(webrtc::AudioSendStream* stream);
 
  private: