minyue-webrtc 82931001c2 Use small BWE period when there is a true network degradation. ...

Bug: webrtc:8105
Change-Id: I751b89194f3fdb10ea41c6f9e48e38edefcbef1a
Reviewed-on: https://chromium-review.googlesource.com/616724
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Michael T <tschumim@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19469}

2017-08-23 17:22:28 +00:00

..

audio_coding

Reland of Stop silently accepting unsupported flags in test binaries (patchset #1 id:1 of https://codereview.webrtc.org/3002963002/ )

2017-08-22 17:55:47 +00:00

audio_conference_mixer

Remove remains of webrtc/base

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

audio_device

Resolves threading issues when audio is interrupted on iOS.

2017-08-23 14:12:07 +00:00

audio_mixer

Remove remains of webrtc/base

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

audio_processing

Robustified the detection of linear filter quality in AEC3

2017-08-23 16:18:49 +00:00

bitrate_controller

Final version of BBR, with tweaks made for WebRTC, major changes:

2017-08-20 16:19:58 +00:00

congestion_controller

Use small BWE period when there is a true network degradation.

2017-08-23 17:22:28 +00:00

desktop_capture

Check child windows in CroppingWindowCapturerWin

2017-08-22 21:49:24 +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

Move PacedSender ownership to RtpTransportControllerSend.

2017-08-22 14:16:49 +00:00

remote_bitrate_estimator

Use small BWE period when there is a true network degradation.

2017-08-23 17:22:28 +00:00

rtp_rtcp

Reland of Add a flags field to video timing extension. (patchset #1 id:1 of https://codereview.webrtc.org/2995953002/ )

2017-08-18 09:51:12 +00:00

utility

Tracking mock_process_thread with a GN target

2017-08-03 12:57:11 +00:00

video_capture

Replace old logging mechanism in webrtc/modules/video_capture/linux

2017-08-16 13:51:27 +00:00

video_coding

Move video send/receive stream headers to webrtc/call.

2017-08-22 12:43:23 +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