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_processing/audio_processing_impl_locking_unittest.cc b/modules/audio_processing/audio_processing_impl_locking_unittest.cc
index 39f8b8b..b770fef 100644
--- a/modules/audio_processing/audio_processing_impl_locking_unittest.cc
+++ b/modules/audio_processing/audio_processing_impl_locking_unittest.cc
@@ -146,8 +146,7 @@
// Create test config for the first processing API function set.
test_configs.push_back(test_config);
- test_config.render_api_function =
- RenderApiImpl::AnalyzeReverseStreamImpl;
+ test_config.render_api_function = RenderApiImpl::AnalyzeReverseStreamImpl;
test_config.capture_api_function = CaptureApiImpl::ProcessStreamImpl3;
test_configs.push_back(test_config);
}
@@ -482,8 +481,7 @@
for (size_t k = 0; k < frame->samples_per_channel_; k++) {
// Store random 16 bit number between -(amplitude+1) and
// amplitude.
- frame_data[k * ch] =
- rand_gen->RandInt(2 * amplitude + 1) - amplitude - 1;
+ frame_data[k * ch] = rand_gen->RandInt(2 * amplitude + 1) - amplitude - 1;
}
}
}
