Zijie He dab31ce1fa Detect whether a window is out of screen when using cropping window capturer ...

We are still using cropping window capturer even the window is out of the screen.
See the bug for details.

Bug: webrtc:8134
Change-Id: I5161b1a17a3a1f8244697eea5eb78975be6908f9
Reviewed-on: https://chromium-review.googlesource.com/627338
Commit-Queue: Zijie He <zijiehe@chromium.org>
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Cr-Commit-Position: refs/heads/master@{#19474}

2017-08-23 19:28:39 +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

Detect whether a window is out of screen when using cropping window capturer

2017-08-23 19:28:39 +00:00

include

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

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

media_file

Replace WEBRTC_TRACE logging in modules/media_file/

2017-08-23 17:30:38 +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