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.h b/modules/audio_coding/neteq/tools/audio_loop.h
index abb1a36..876c2d7 100644
--- a/modules/audio_coding/neteq/tools/audio_loop.h
+++ b/modules/audio_coding/neteq/tools/audio_loop.h
@@ -26,10 +26,7 @@
class AudioLoop {
public:
AudioLoop()
- : next_index_(0),
- loop_length_samples_(0),
- block_length_samples_(0) {
- }
+ : next_index_(0), loop_length_samples_(0), block_length_samples_(0) {}
virtual ~AudioLoop() {}
@@ -38,7 +35,8 @@
// greater. Otherwise, the loop length is the same as the file length.
// The audio will be delivered in blocks of |block_length_samples|.
// Returns false if the initialization failed, otherwise true.
- bool Init(const std::string file_name, size_t max_loop_length_samples,
+ bool Init(const std::string file_name,
+ size_t max_loop_length_samples,
size_t block_length_samples);
// Returns a (pointer,size) pair for the next block of audio. The size is