webrtc/modules/audio_coding/acm2
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
..
acm_codec_database.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
acm_codec_database.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
acm_receive_test.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
acm_receive_test.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
acm_receiver.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
acm_receiver.h Replace rtc::Optional with absl::optional in modules/audio_coding 2018-06-19 12:46:20 +00:00
acm_receiver_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
acm_resampler.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
acm_resampler.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
acm_send_test.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
acm_send_test.h Remove dependencies on modules:module_api from AudioProcessing. 2018-04-12 22:05:27 +00:00
audio_coding_module.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_coding_module_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
call_statistics.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
call_statistics.h Remove dependencies on modules:module_api from AudioProcessing. 2018-04-12 22:05:27 +00:00
call_statistics_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
codec_manager.cc Optional: Use nullopt and implicit construction in /modules/audio_coding 2017-11-17 11:58:37 +00:00
codec_manager.h Replace rtc::Optional with absl::optional in modules/audio_coding 2018-06-19 12:46:20 +00:00
codec_manager_unittest.cc Removed Die mock from MockAudioEncoder 2018-02-22 12:53:38 +00:00
rent_a_codec.cc Replace rtc::Optional with absl::optional in modules/audio_coding 2018-06-19 12:46:20 +00:00
rent_a_codec.h Replace rtc::Optional with absl::optional in modules/audio_coding 2018-06-19 12:46:20 +00:00
rent_a_codec_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00