webrtc/modules/audio_coding/codecs/opus
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
..
audio_decoder_opus.cc Replace rtc::Optional with absl::optional in modules/audio_coding 2018-06-19 12:46:20 +00:00
audio_decoder_opus.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
audio_encoder_opus.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_encoder_opus.h Replace rtc::Optional with absl::optional in modules/audio_coding 2018-06-19 12:46:20 +00:00
audio_encoder_opus_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
opus_bandwidth_unittest.cc Replace rtc::Optional with absl::optional in modules/audio_coding 2018-06-19 12:46:20 +00:00
opus_complexity_unittest.cc Optional: Use nullopt and implicit construction in /modules/audio_coding 2017-11-17 11:58:37 +00:00
opus_fec_test.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
opus_inst.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
opus_interface.c Implement Opus bandwidth adjustment behind a FieldTrial 2017-11-20 20:04:19 +00:00
opus_interface.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
opus_speed_test.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
opus_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00