minyue-webrtc 12d30840d8 Correct the calculation of discard rate. ...

Bug: webrtc:7903
Change-Id: Ib5d6fd882a994dd542b616e5fe1c75710346dd31
Reviewed-on: https://chromium-review.googlesource.com/575057
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19101}

2017-07-20 09:15:46 +00:00

..

audio_coding

Correct the calculation of discard rate.

2017-07-20 09:15:46 +00:00

audio_conference_mixer

Remove remains of webrtc/base

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

audio_device

Remove remains of webrtc/base

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

audio_mixer

Remove remains of webrtc/base

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

audio_processing

Remove remains of webrtc/base

2017-07-19 17:40:47 +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

Remove remains of webrtc/base

2017-07-19 17:40:47 +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 remains of webrtc/base

2017-07-19 17:40:47 +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

Move temporal-layer properties to FrameConfig.

2017-07-20 00:04:02 +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