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/neteq/tools/audio_loop.cc b/modules/audio_coding/neteq/tools/audio_loop.cc
index b5ad881..972921b 100644
--- a/modules/audio_coding/neteq/tools/audio_loop.cc
+++ b/modules/audio_coding/neteq/tools/audio_loop.cc
@@ -21,16 +21,18 @@
size_t max_loop_length_samples,
size_t block_length_samples) {
FILE* fp = fopen(file_name.c_str(), "rb");
- if (!fp) return false;
+ if (!fp)
+ return false;
- audio_array_.reset(new int16_t[max_loop_length_samples +
- block_length_samples]);
- size_t samples_read = fread(audio_array_.get(), sizeof(int16_t),
- max_loop_length_samples, fp);
+ audio_array_.reset(
+ new int16_t[max_loop_length_samples + block_length_samples]);
+ size_t samples_read =
+ fread(audio_array_.get(), sizeof(int16_t), max_loop_length_samples, fp);
fclose(fp);
// Block length must be shorter than the loop length.
- if (block_length_samples > samples_read) return false;
+ if (block_length_samples > samples_read)
+ return false;
// Add an extra block length of samples to the end of the array, starting
// over again from the beginning of the array. This is done to simplify
@@ -54,6 +56,5 @@
return rtc::ArrayView<const int16_t>(output_ptr, block_length_samples_);
}
-
} // namespace test
} // namespace webrtc