webrtc_m130

History

Niels Möller eaea27799d Delete deprecated static method FileStream::GetSize.

Unused since cl https://codereview.webrtc.org/2694753002

Bug: webrtc:6424
Change-Id: Id22ecd5c004a0b4af3e673da519cbd537f99cd8b
Reviewed-on: https://chromium-review.googlesource.com/626116
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19482}

2017-08-24 08:36:14 +00:00

api

Report max interframe delay over window insdead of interframe delay sum

2017-08-23 12:24:10 +00:00

audio

Move PacedSender ownership to RtpTransportControllerSend.

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

call

Report max interframe delay over window insdead of interframe delay sum

2017-08-23 12:24:10 +00:00

common_audio

Replace absolute path with relative path for GN files.

2017-08-15 15:57:36 +00:00

common_video

Change logging method from WEBRTC_TRACE macro to LOG macro.

2017-08-22 07:29:13 +00:00

examples

Add myself as owner of example apps.

2017-08-24 08:35:05 +00:00

logging

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

2017-08-22 12:43:23 +00:00

media

Reland of Modify profiles for H264 encode SW fallback (patchset #1 id:1 of https://codereview.webrtc.org/2995373002/ )

2017-08-23 21:19:50 +00:00

modules

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

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

ortc

Wire up RTP keep-alive in ortc api.

2017-08-09 13:42:32 +00:00

p2p

Renamed dtlstransportchannel.h/.cc/_unittest.cc.

2017-08-24 01:05:50 +00:00

Delete unneeded include of videocapturer.h

2017-08-24 07:29:04 +00:00

rtc_base

Delete deprecated static method FileStream::GetSize.

2017-08-24 08:36:14 +00:00

rtc_tools

Move PacedSender ownership to RtpTransportControllerSend.

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

sdk

Adding injectable audio decoder and encoder factory support to the RTCPeerConnection obj-c layer.

2017-08-23 22:46:44 +00:00

stats

Audit of kConstants missing the const qualifier

2017-08-08 17:48:15 +00:00

system_wrappers

Replace absolute path with relative path for GN files.

2017-08-15 15:57:36 +00:00

test

The goal of this CL is to separate Obj-C/Obj-C++ code from targets which have

2017-08-24 06:55:54 +00:00

video

Add experiment to disable ulpfec.

2017-08-23 17:40:00 +00:00

voice_engine

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

.gitignore

Flesh out webrtc/.gitignore

2015-11-16 19:02:02 +00:00

BUILD.gn

Reland of Enable -Wunused-lambda-capture warning (patchset #1 id:1 of https://codereview.webrtc.org/3007483002/ )

2017-08-23 19:31:31 +00:00

codereview.settings

Give a more specific URL for creating WebRTC checkout

2016-04-01 07:36:53 +00:00

common_types.cc

Rename webrtc/base -> webrtc/rtc_base

2017-07-06 19:11:40 +00:00

common_types.h

Wire up RTP keep-alive in ortc api.

2017-08-09 13:42:32 +00:00

config_unittest.cc

Rename webrtc/base -> webrtc/rtc_base

2017-07-06 19:11:40 +00:00

config.cc

Audit of kConstants missing the const qualifier

2017-08-08 17:48:15 +00:00

config.h

Audit of kConstants missing the const qualifier

2017-08-08 17:48:15 +00:00

DEPS

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

2017-08-22 12:43:23 +00:00

LICENSE

…

LICENSE_THIRD_PARTY

Delete unused spreadsort implementation.

2016-12-05 11:03:26 +00:00

OWNERS

Don't add or rename files in webrtc/ and webrtc/api/ without a proper review

2017-05-08 13:01:54 +00:00

PATENTS

…

PRESUBMIT.py

…

README.chromium

…

typedefs.h

Make WARN_UNUSED_RESULT a no-op on gcc

2017-04-06 11:32:27 +00:00

video_receive_stream.h

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

2017-08-22 12:43:23 +00:00

video_send_stream.h

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

2017-08-22 12:43:23 +00:00

webrtc.gni

Replace absolute path with relative path for GN files.

2017-08-15 15:57:36 +00:00

whitespace.txt

Moving whitespace file up by one folder

2017-01-24 13:19:42 +00:00

README.chromium

Name: WebRTC
URL: http://www.webrtc.org
Version: 90
License: BSD
License File: LICENSE

Description:
WebRTC provides real time voice and video processing
functionality to enable the implementation of 
PeerConnection/MediaStream.

Third party code used in this project is described 
in the file LICENSE_THIRD_PARTY.