admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
webrtc_m130/webrtc
History
brandtr ed0e436688 VideoProcessorIntegrationTest: mini-fixes in preparation for task queue CL. 
* Rename some members.
* Shorten visualization file names.
* Make some member functions static, in preparation for moving them
  to be helper functions in an anonymous namespace.

BUG=webrtc:6634

Review-Url: https://codereview.webrtc.org/3001193002
Cr-Commit-Position: refs/heads/master@{#19424}
2017-08-21 11:15:24 +00:00
..
api
iSAC fixed-point implementation of the Audio{En,De}coderFactoryTemplate APIs
2017-08-18 11:09:40 +00:00
audio
Replace absolute path with relative path for GN files.
2017-08-15 15:57:36 +00:00
call
Reduce code repetition in RtcpDemuxerTest.
2017-08-18 18:41:39 +00:00
common_audio
Replace absolute path with relative path for GN files.
2017-08-15 15:57:36 +00:00
common_video
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
examples
Remove dep on system_wrappers for apprtc and use only framework.
2017-08-16 14:28:03 +00:00
logging
Add rtc_event_log_unittest_helper.h to relevant BUILD.gn
2017-08-17 15:32:39 +00:00
media
Move kMinPixelsPerFrame constant in VideoStreamEncoder to VideoEncoder::ScalingSettings.
2017-08-17 15:58:54 +00:00
modules
VideoProcessorIntegrationTest: mini-fixes in preparation for task queue CL.
2017-08-21 11:15:24 +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
pc
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
Delete old CongestionController class
2017-08-21 10:43:33 +00:00
sdk
Fix a bug where listener custom drawer was ignored in EglRenderer.
2017-08-18 18:38:32 +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
Delete old CongestionController class
2017-08-21 10:43:33 +00:00
video
Revert of Add Jpeg frame writer for test support. (patchset #12 id:220001 of https://codereview.webrtc.org/2990563002/ )
2017-08-20 19:50:56 +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
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
BUILD.gn
Reland of Make it possible for tests to set up trace event handlers. (patchset #1 id:1 of https://codereview.webrtc.org/2997963002/ )
2017-08-17 14:59:54 +00:00
codereview.settings
…
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 webrtc/{tools => rtc_tools}
2017-06-30 10:04:59 +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
Protected streams report RTP messages directly to the FlexFec streams
2017-08-02 14:39:07 +00:00
video_send_stream.h
Move RTP keep-alive config from VideoSendStream::Config to Call::Config
2017-07-11 10:44:17 +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.