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_coding/neteq/tools/audio_loop.h

@@ -26,10 +26,7 @@ namespace test {
 class AudioLoop {
  public:
   AudioLoop()
-      : next_index_(0),
-        loop_length_samples_(0),
-        block_length_samples_(0) {
-  }
+      : next_index_(0), loop_length_samples_(0), block_length_samples_(0) {}
 
   virtual ~AudioLoop() {}
 
@@ -38,7 +35,8 @@ class AudioLoop {
   // greater. Otherwise, the loop length is the same as the file length.
   // The audio will be delivered in blocks of |block_length_samples|.
   // Returns false if the initialization failed, otherwise true.
-  bool Init(const std::string file_name, size_t max_loop_length_samples,
+  bool Init(const std::string file_name,
+            size_t max_loop_length_samples,
             size_t block_length_samples);
 
   // Returns a (pointer,size) pair for the next block of audio. The size is