Artem Titov 5d7a4c6692 Fixing py lint errors ...

Bug: webrtc:9548
Change-Id: I0daf8dc06fdaac1637c32994ef6ad542ed52202a
Reviewed-on: https://webrtc-review.googlesource.com/90045
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24068}

2018-07-23 15:28:48 +00:00

..

include

Replace rtc::Optional with absl::optional in modules/audio_coding

2018-06-19 12:46:20 +00:00

mock

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

util

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

audio_network_adaptor_config.cc

Create RtcEventLogEncoderLegacy

2017-10-03 13:51:59 +00:00

audio_network_adaptor_impl_unittest.cc

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

audio_network_adaptor_impl.cc

Replace rtc::Optional with absl::optional in modules/audio_coding

2018-06-19 12:46:20 +00:00

audio_network_adaptor_impl.h

Replace rtc::Optional with absl::optional in modules/audio_coding

2018-06-19 12:46:20 +00:00

bitrate_controller_unittest.cc

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

bitrate_controller.cc

Optional: Use nullopt and implicit construction in /modules/audio_coding

2017-11-17 11:58:37 +00:00

bitrate_controller.h

Replace rtc::Optional with absl::optional in modules/audio_coding

2018-06-19 12:46:20 +00:00

channel_controller_unittest.cc

Replace rtc::Optional with absl::optional in modules/audio_coding

2018-06-19 12:46:20 +00:00

channel_controller.cc

Optional: Use nullopt and implicit construction in /modules/audio_coding

2017-11-17 11:58:37 +00:00

channel_controller.h

Replace rtc::Optional with absl::optional in modules/audio_coding

2018-06-19 12:46:20 +00:00

config.proto

Added configurable offsets to the per-packet overhead in the ANA frame length and bitrate controllers.

2017-09-28 08:11:16 +00:00

controller_manager_unittest.cc

Replace rtc::Optional with absl::optional in modules/audio_coding

2018-06-19 12:46:20 +00:00

controller_manager.cc

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

controller_manager.h

Replace rtc::Optional with absl::optional in modules/audio_coding

2018-06-19 12:46:20 +00:00

controller.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

controller.h

Replace rtc::Optional with absl::optional in modules/audio_coding

2018-06-19 12:46:20 +00:00

debug_dump_writer.cc

Stop using ByteSize (deprecated) to get the size of a proto message.

2017-11-30 14:27:50 +00:00

debug_dump_writer.h

Move file_wrapper.h to rtc_base/system/

2018-03-23 11:17:15 +00:00

debug_dump.proto

…

dtx_controller_unittest.cc

Replace rtc::Optional with absl::optional in modules/audio_coding

2018-06-19 12:46:20 +00:00

dtx_controller.cc

Optional: Use nullopt and implicit construction in /modules/audio_coding

2017-11-17 11:58:37 +00:00

dtx_controller.h

Replace rtc::Optional with absl::optional in modules/audio_coding

2018-06-19 12:46:20 +00:00

event_log_writer_unittest.cc

Optional: Use nullopt and implicit construction in /modules/audio_coding

2017-11-17 11:58:37 +00:00

event_log_writer.cc

Use absl::make_unique and absl::WrapUnique directly

2018-07-05 10:59:49 +00:00

event_log_writer.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

fec_controller_plr_based_unittest.cc

Replace rtc::Optional with absl::optional in modules/audio_coding

2018-06-19 12:46:20 +00:00

fec_controller_plr_based.cc

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

fec_controller_plr_based.h

Replace rtc::Optional with absl::optional in modules/audio_coding

2018-06-19 12:46:20 +00:00

fec_controller_rplr_based_unittest.cc

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

fec_controller_rplr_based.cc

Optional: Use nullopt and implicit construction in /modules/audio_coding

2017-11-17 11:58:37 +00:00

fec_controller_rplr_based.h

Replace rtc::Optional with absl::optional in modules/audio_coding

2018-06-19 12:46:20 +00:00

frame_length_controller_unittest.cc

Replace rtc::Optional with absl::optional in modules/audio_coding

2018-06-19 12:46:20 +00:00

frame_length_controller.cc

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

frame_length_controller.h

Replace rtc::Optional with absl::optional in modules/audio_coding

2018-06-19 12:46:20 +00:00

parse_ana_dump.py

Fixing py lint errors

2018-07-23 15:28:48 +00:00