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/video_coding/jitter_buffer_unittest.cc

@@ -214,8 +214,7 @@ class TestBasicJitterBuffer : public ::testing::TestWithParam<std::string>,
     clock_.reset(new SimulatedClock(0));
     jitter_buffer_.reset(new VCMJitterBuffer(
         clock_.get(),
-        std::unique_ptr<EventWrapper>(event_factory_.CreateEvent()),
-        this,
+        std::unique_ptr<EventWrapper>(event_factory_.CreateEvent()), this,
         this));
     jitter_buffer_->Start();
     seq_num_ = 1234;
@@ -333,8 +332,8 @@ class TestRunningJitterBuffer : public ::testing::TestWithParam<std::string>,
     oldest_packet_to_nack_ = 250;
     jitter_buffer_ = new VCMJitterBuffer(
         clock_.get(),
-        std::unique_ptr<EventWrapper>(event_factory_.CreateEvent()),
-        this, this);
+        std::unique_ptr<EventWrapper>(event_factory_.CreateEvent()), this,
+        this);
     stream_generator_ = new StreamGenerator(0, clock_->TimeInMilliseconds());
     jitter_buffer_->Start();
     jitter_buffer_->SetNackSettings(max_nack_list_size_, oldest_packet_to_nack_,