commit 665174fdbb4e0540eccb27cf7412348f1b65534c
author Yves Gerey <yvesg@webrtc.org> Tue Jun 19 13:03:05 2018
committer Yves Gerey <yvesg@webrtc.org> Tue Jun 19 14:00:39 2018
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}

media/engine/fakewebrtccall.h

diff --git a/media/engine/fakewebrtccall.h b/media/engine/fakewebrtccall.h
index 00bd079..4ee6a80 100644
--- a/media/engine/fakewebrtccall.h
+++ b/media/engine/fakewebrtccall.h
@@ -45,8 +45,8 @@
     int duration_ms = 0;
   };
 
-  explicit FakeAudioSendStream(
-      int id, const webrtc::AudioSendStream::Config& config);
+  explicit FakeAudioSendStream(int id,
+                               const webrtc::AudioSendStream::Config& config);
 
   int id() const { return id_; }
   const webrtc::AudioSendStream::Config& GetConfig() const override;
@@ -62,7 +62,9 @@
   void Stop() override { sending_ = false; }
   void SendAudioData(std::unique_ptr<webrtc::AudioFrame> audio_frame) override {
   }
-  bool SendTelephoneEvent(int payload_type, int payload_frequency, int event,
+  bool SendTelephoneEvent(int payload_type,
+                          int payload_frequency,
+                          int event,
                           int duration_ms) override;
   void SetMuted(bool muted) override;
   webrtc::AudioSendStream::Stats GetStats() const override;
@@ -80,7 +82,8 @@
 class FakeAudioReceiveStream final : public webrtc::AudioReceiveStream {
  public:
   explicit FakeAudioReceiveStream(
-      int id, const webrtc::AudioReceiveStream::Config& config);
+      int id,
+      const webrtc::AudioReceiveStream::Config& config);
 
   int id() const { return id_; }
   const webrtc::AudioReceiveStream::Config& GetConfig() const;