webrtc_m130

History

ilnik ee42d19b70 Reland of Add Jpeg frame writer for test support.

Also, use it to save worst psnr frame in video quality tests. It is indented that these saved frames from perfbots will be uploaded to the cloud and will be available in chrome perf dashboard. Because of that size of the saved frame is somewhat an issue. Also, y4m is not convenient to view.

BUG=webrtc:8030

Review-Url: https://codereview.webrtc.org/2998143002
Cr-Commit-Position: refs/heads/master@{#19450}

2017-08-22 14:16:20 +00:00

api

Reimplement the builtin audio codec factories using the new stuff in api/

2017-08-22 12:39:19 +00:00

audio

Use SingleThreadedTaskQueue in DirectTransport

2017-08-22 11:02:52 +00:00

call

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

2017-08-22 12:43:23 +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

Remove dep on system_wrappers for apprtc and use only framework.

2017-08-16 14:28:03 +00:00

logging

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

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

media

Add video timing frames to set of default RTP header extensions

2017-08-22 14:05:47 +00:00

modules

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

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

ortc

Wire up RTP keep-alive in ortc api.

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

p2p

Add TLS TURN tests.

2017-08-17 23:03:04 +00:00

Relanding: Adding support for Unified Plan offer/answer negotiation to the mediasession layer.

2017-08-17 21:10:50 +00:00

rtc_base

Fix a delete type mismatch (deleting a sockaddr_in6* as a sockaddr*)

2017-08-18 07:15:19 +00:00

rtc_tools

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

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

sdk

Remove unnecessary release call in EglRenderer.

2017-08-22 07:21:48 +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

Reland of Add Jpeg frame writer for test support.

2017-08-22 14:16:20 +00:00

video

Reland of Add Jpeg frame writer for test support.

2017-08-22 14:16:20 +00:00

voice_engine

Revert of Stop silently accepting unsupported flags in test binaries (patchset #5 id:150001 of https://codereview.webrtc.org/2968003003/ )

2017-08-18 14:22:36 +00:00

.gitignore

…

BUILD.gn

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

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