mollyim/webrtc
1
0
Fork 0
mirror of https://github.com/mollyim/webrtc.git synced 2025-05-14 06:10:40 +01:00
Code Issues Projects Releases Packages Wiki Activity Actions
webrtc/modules/audio_coding/codecs
History
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
..
cng Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
g711 Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
g722 Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
ilbc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
isac Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
opus Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
pcm16b Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
red Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
tools Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_decoder.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
audio_encoder.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
audio_format_conversion.cc Replace rtc::Optional with absl::optional in modules/audio_coding 2018-06-19 12:46:20 +00:00
audio_format_conversion.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
builtin_audio_decoder_factory_unittest.cc Replace rtc::Optional with absl::optional in modules/audio_coding 2018-06-19 12:46:20 +00:00
builtin_audio_encoder_factory_unittest.cc Replace rtc::Optional with absl::optional in modules/audio_coding 2018-06-19 12:46:20 +00:00
legacy_encoded_audio_frame.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
legacy_encoded_audio_frame.h Replace rtc::Optional with absl::optional in modules/audio_coding 2018-06-19 12:46:20 +00:00
legacy_encoded_audio_frame_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
OWNERS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00