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/api/audio_codecs/audio_encoder.h b/api/audio_codecs/audio_encoder.h
index 7ad9ba4..d277a19 100644
--- a/api/audio_codecs/audio_encoder.h
+++ b/api/audio_codecs/audio_encoder.h
@@ -220,9 +220,8 @@
// Provides target audio bitrate and corresponding probing interval of
// the bandwidth estimator to this encoder to allow it to adapt.
- virtual void OnReceivedUplinkBandwidth(
- int target_audio_bitrate_bps,
- rtc::Optional<int64_t> bwe_period_ms);
+ virtual void OnReceivedUplinkBandwidth(int target_audio_bitrate_bps,
+ rtc::Optional<int64_t> bwe_period_ms);
// Provides RTT to this encoder to allow it to adapt.
virtual void OnReceivedRtt(int rtt_ms);