webrtc/modules/audio_coding/codecs/isac
Yves Gerey 665174fdbb 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}
2018-06-19 14:00:39 +00:00
..
fix Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
main Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_decoder_isac_t.h Replace rtc::Optional with absl::optional in modules/audio_coding 2018-06-19 12:46:20 +00:00
audio_decoder_isac_t_impl.h Add missing iSAC headers. 2018-01-02 13:01:11 +00:00
audio_encoder_isac_t.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
audio_encoder_isac_t_impl.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
bandwidth_info.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
empty.cc Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
locked_bandwidth_info.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
locked_bandwidth_info.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
unittest.cc Move some numeric utility code from rtc_base/ to rtc_base/numerics/ 2017-11-22 11:21:47 +00:00