Stefan Holmer 1acbd68718 Move RtpExtension to api/ directory and config.h/.cc to call/. ...

BUG=webrtc:5876
R=deadbeef@webrtc.org, solenberg@webrtc.org

Review-Url: https://codereview.webrtc.org/3004723002 .
Cr-Commit-Position: refs/heads/master@{#19639}

2017-09-01 13:29:30 +00:00

..

audio_coding

Move RtpExtension to api/ directory and config.h/.cc to call/.

2017-09-01 13:29:30 +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

Removes unused APIs from the ADM (part II).

2017-08-31 13:47:32 +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

Move RtpExtension to api/ directory and config.h/.cc to call/.

2017-09-01 13:29:30 +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

Move RtpExtension to api/ directory and config.h/.cc to call/.

2017-09-01 13:29:30 +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

Split up VideoProcessorIntegrationTest files.

2017-09-01 07:10:27 +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