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}
diff --git a/modules/audio_coding/test/Channel.h b/modules/audio_coding/test/Channel.h
index e01e33e..e5f5b54 100644
--- a/modules/audio_coding/test/Channel.h
+++ b/modules/audio_coding/test/Channel.h
@@ -20,8 +20,8 @@
 
 namespace webrtc {
 
-#define MAX_NUM_PAYLOADS   50
-#define MAX_NUM_FRAMESIZES  6
+#define MAX_NUM_PAYLOADS 50
+#define MAX_NUM_FRAMESIZES 6
 
 // TODO(turajs): Write constructor for this structure.
 struct ACMTestFrameSizeStats {
@@ -45,7 +45,6 @@
 
 class Channel : public AudioPacketizationCallback {
  public:
-
   Channel(int16_t chID = -1);
   ~Channel() override;
 
@@ -56,7 +55,7 @@
                    size_t payloadSize,
                    const RTPFragmentationHeader* fragmentation) override;
 
-  void RegisterReceiverACM(AudioCodingModule *acm);
+  void RegisterReceiverACM(AudioCodingModule* acm);
 
   void ResetStats();
 
@@ -68,9 +67,7 @@
 
   void PrintStats(CodecInst& codecInst);
 
-  void SetIsStereo(bool isStereo) {
-    _isStereo = isStereo;
-  }
+  void SetIsStereo(bool isStereo) { _isStereo = isStereo; }
 
   uint32_t LastInTimestamp();