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webrtc/modules/audio_coding/audio_network_adaptor
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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.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
audio_network_adaptor_impl_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
bitrate_controller_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +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
channel_controller_unittest.cc 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.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
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_manager_unittest.cc Replace rtc::Optional with absl::optional in modules/audio_coding 2018-06-19 12:46:20 +00:00
debug_dump.proto Moving src/webrtc into src/. 2017-09-15 04:25:06 +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
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
dtx_controller_unittest.cc Replace rtc::Optional with absl::optional in modules/audio_coding 2018-06-19 12:46:20 +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
event_log_writer_unittest.cc Optional: Use nullopt and implicit construction in /modules/audio_coding 2017-11-17 11:58:37 +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_plr_based_unittest.cc Replace rtc::Optional with absl::optional in modules/audio_coding 2018-06-19 12:46:20 +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
fec_controller_rplr_based_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +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
frame_length_controller_unittest.cc 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