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Artem Titov 3ecec176a8 Extract third party part of g711 codec into separate target 
Bug: webrtc:8366
Change-Id: I34c7ea707213e0c1a50826896da01f70c072eae5
Reviewed-on: https://webrtc-review.googlesource.com/84741
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23724}
2018-06-25 11:26:59 +00:00
..
audio_coding Extract third party part of g711 codec into separate target 2018-06-25 11:26:59 +00:00
audio_device Replace rtc::Optional with absl::optional 2018-06-21 09:32:56 +00:00
audio_mixer Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_processing Remove nonlinear beamformer API from APM 2018-06-21 08:49:52 +00:00
bitrate_controller Removing usage of //build/config/compiler:no_size_t_to_int_warning. 2018-06-20 13:44:26 +00:00
congestion_controller Removing usage of //build/config/compiler:no_size_t_to_int_warning. 2018-06-20 13:44:26 +00:00
desktop_capture Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
include Add RTPVideoHeader const accessor. 2018-06-21 09:49:40 +00:00
pacing Removing usage of //build/config/compiler:no_size_t_to_int_warning. 2018-06-20 13:44:26 +00:00
remote_bitrate_estimator Removing some MSVC warning suppression flags. 2018-06-20 12:41:46 +00:00
rtp_rtcp Generalize SimulcastEncoderAdapter, use for H264 & VP8. 2018-06-21 15:57:43 +00:00
utility Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
video_capture Revert "Remove deprecated mac capture code." 2018-06-20 12:24:16 +00:00
video_coding Fix for VP9 K-SVC video freeze frame when send bandwidth is restricted. 2018-06-21 17:53:35 +00:00
video_processing Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
BUILD.gn Replace rtc::Optional with absl::optional 2018-06-21 09:32:56 +00:00
module_common_types_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