minyue-webrtc 5fcd1cc4bb Decouple FEC controller's decision from the decision making for frame length. ...

Bug: webrtc:8098
Change-Id: If313c8069a16bfb3c2752dcca7fb1cfca24c1caf
Reviewed-on: https://chromium-review.googlesource.com/643299
Reviewed-by: Michael T <tschumim@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19627}

2017-08-31 12:08:56 +00:00

..

audio_coding

Decouple FEC controller's decision from the decision making for frame length.

2017-08-31 12:08:56 +00:00

audio_conference_mixer

Recently we moved webrtc/base to webrtc/rtc_base, so these

2017-08-30 11:45:58 +00:00

audio_device

Recently we moved webrtc/base to webrtc/rtc_base, so these

2017-08-30 11:45:58 +00:00

audio_mixer

Recently we moved webrtc/base to webrtc/rtc_base, so these

2017-08-30 11:45:58 +00:00

audio_processing

Replace remaining gflags usages with rtc_base/flags

2017-08-31 10:21:39 +00:00

bitrate_controller

Recently we moved webrtc/base to webrtc/rtc_base, so these

2017-08-30 11:45:58 +00:00

congestion_controller

Recently we moved webrtc/base to webrtc/rtc_base, so these

2017-08-30 11:45:58 +00:00

desktop_capture

Flip IsWindowOnScreen behavior when native APIs fail

2017-08-31 02:43:14 +00:00

include

Recently we moved webrtc/base to webrtc/rtc_base, so these

2017-08-30 11:45:58 +00:00

media_file

Recently we moved webrtc/base to webrtc/rtc_base, so these

2017-08-30 11:45:58 +00:00

pacing

Recently we moved webrtc/base to webrtc/rtc_base, so these

2017-08-30 11:45:58 +00:00

remote_bitrate_estimator

Replace remaining gflags usages with rtc_base/flags

2017-08-31 10:21:39 +00:00

rtp_rtcp

Recently we moved webrtc/base to webrtc/rtc_base, so these

2017-08-30 11:45:58 +00:00

utility

Recently we moved webrtc/base to webrtc/rtc_base, so these

2017-08-30 11:45:58 +00:00

video_capture

Recently we moved webrtc/base to webrtc/rtc_base, so these

2017-08-30 11:45:58 +00:00

video_coding

Replace remaining gflags usages with rtc_base/flags

2017-08-31 10:21:39 +00:00

video_processing

Recently we moved webrtc/base to webrtc/rtc_base, so these

2017-08-30 11:45:58 +00:00

BUILD.gn

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

module_common_types_unittest.cc

Add mute state field to AudioFrame and switch some callers to use it. Also make AudioFrame::data_ private and instead provide:

2017-06-12 19:45:32 +00:00

OWNERS

Remove all references to GYP

2016-11-16 19:11:38 +00:00