lliuu ff2824b917 Fix compiler warnings due to type mismatch. ...

BUG=webrtc:8169

Change-Id: Ie98dcaa09e1e5b7c9f48410cd3049740ccac67b9
Reviewed-on: https://chromium-review.googlesource.com/639314
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Commit-Queue: Lu Liu <lliuu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19567}

2017-08-28 20:31:58 +00:00

..

audio_coding

Fix compiler warnings due to type mismatch.

2017-08-28 20:31:58 +00:00

audio_conference_mixer

Remove remains of webrtc/base

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

audio_device

Removes usage audio_device_test_api.

2017-08-25 14:47:31 +00:00

audio_mixer

Remove remains of webrtc/base

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

audio_processing

Decrease the default reverberation in AEC3

2017-08-25 14:07:30 +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

Move GetOutstandingBytes() call inside field trial check to avoid taking an unneccessary lock.

2017-08-26 13:33:43 +00:00

desktop_capture

Use windows' content rectangles rather than their borders to determine overlapping

2017-08-26 01:10:16 +00:00

include

Implement move constructor for RTPFragmentationHeader.

2017-08-28 10:22:39 +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 Verify sender ssrc when receiving rtcp target bitrate (patchset #1 id:1 of https://codereview.webrtc.org/3005633002/ )

2017-08-28 12:49:12 +00:00

utility

Now that https://codereview.webrtc.org/3003643002 is landed we can

2017-08-28 06:40:10 +00:00

video_capture

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

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

video_coding

Removing dependencies on stub headers within WebRTC.

2017-08-28 13:46:48 +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