danilchap a04d9c31a0 Remove RtpRtcp::RemoteRTCPStat(RTCPSenderInfo*) as unused ...

BUG=webrtc:5565

Review-Url: https://codereview.webrtc.org/2986543002
Cr-Commit-Position: refs/heads/master@{#19128}

2017-07-25 11:03:39 +00:00

..

audio_coding

Disable some Opus tests pending an update

2017-07-24 09:17:38 +00:00

audio_conference_mixer

Remove remains of webrtc/base

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

audio_device

Fixing lint issue

2017-07-21 14:28:41 +00:00

audio_mixer

Remove remains of webrtc/base

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

audio_processing

Integer overflow bug in low_cut_filter.

2017-07-24 11:03:46 +00:00

bitrate_controller

Remove remains of webrtc/base

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

congestion_controller

Remove remains of webrtc/base

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

desktop_capture

Add zijiehe@chromium.org as OWNERS in WebRTC DesktopCapturer related logic

2017-07-23 21:37:59 +00:00

include

Add AudioFrame::ResetWithoutMuting() to address performance regression.

2017-07-08 23:36:26 +00:00

media_file

Remove remains of webrtc/base

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

pacing

Remove remains of webrtc/base

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

remote_bitrate_estimator

Remove remains of webrtc/base

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

rtp_rtcp

Remove RtpRtcp::RemoteRTCPStat(RTCPSenderInfo*) as unused

2017-07-25 11:03:39 +00:00

utility

Remove remains of webrtc/base

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

video_capture

Remove remains of webrtc/base

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

video_coding

Throttle log message in FrameBuffer.

2017-07-24 15:26:53 +00:00

video_processing

Remove remains of webrtc/base

2017-07-19 17:40:47 +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