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/remix_resample.cc

@@ -68,9 +68,9 @@ void RemixAndResample(const int16_t* src_data,
   // how much to zero here; or 2) make resampler accept a hint that the input is
   // zeroed.
   const size_t src_length = samples_per_channel * audio_ptr_num_channels;
-  int out_length = resampler->Resample(audio_ptr, src_length,
-                                       dst_frame->mutable_data(),
-                                       AudioFrame::kMaxDataSizeSamples);
+  int out_length =
+      resampler->Resample(audio_ptr, src_length, dst_frame->mutable_data(),
+                          AudioFrame::kMaxDataSizeSamples);
   if (out_length == -1) {
     FATAL() << "Resample failed: audio_ptr = " << audio_ptr
             << ", src_length = " << src_length