admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
webrtc_m130/webrtc
History
oprypin 30cda5ef98 Espresso test for loopback video quality testing 
Update the AppRTCMobileTestStubbedVideoIO test to run on
phones without Internet connection. This is done by bringing up
a local instance of AppRTC on the Linux machine connected to
the Android device.

Running this test will need the webrtc.DEPS solution to be configured
for the checkout, since that will pull down the precompiled AppRTC
package that is needed.

Continued from http://crrev.com/2780493002#ps20001 (by kjellander@)
Continued from http://crrev.com/2741743002#ps180001 (by mandermo@)

BUG=webrtc:7185

Review-Url: https://codereview.webrtc.org/2825313002
Cr-Commit-Position: refs/heads/master@{#17838}
2017-04-24 11:15:13 +00:00
..
api
Enable GN check for webrtc/{p2p,system_wrappers}
2017-04-21 12:17:08 +00:00
audio
Fix lint errors to enable stricter PyLint rules
2017-04-12 19:06:13 +00:00
base
Negotiate the same SRTP crypto suites for every DTLS association formed.
2017-04-21 10:23:33 +00:00
call
Replace first_packet_sent_ms_ in Call.
2017-04-20 06:28:53 +00:00
common_audio
Enable GN check for webrtc/base
2017-04-19 15:37:36 +00:00
common_video
Delete unneeded includes of deprecated system_wrappers include files.
2017-03-30 07:31:15 +00:00
examples
Espresso test for loopback video quality testing
2017-04-24 11:15:13 +00:00
logging
Delay based logging.
2017-04-11 08:50:23 +00:00
media
Enable GN check for webrtc/{p2p,system_wrappers}
2017-04-21 12:17:08 +00:00
modules
Reland of GN: Enable ARC for Mac and iOS in rtc_* templates (patchset #1 id:1 of https://codereview.webrtc.org/2827223003/ )
2017-04-24 07:57:16 +00:00
ortc
Relanding: Remove rtc_p2p_unittests from ortc_unittests and rtc_media_unittests
2017-04-18 22:49:09 +00:00
p2p
Fixing SignalSentPacket for TCP connections.
2017-04-21 21:22:23 +00:00
pc
Negotiate the same SRTP crypto suites for every DTLS association formed.
2017-04-21 10:23:33 +00:00
sdk
Reland of GN: Enable ARC for Mac and iOS in rtc_* templates (patchset #1 id:1 of https://codereview.webrtc.org/2827223003/ )
2017-04-24 07:57:16 +00:00
stats
Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.
2017-03-01 09:02:45 +00:00
system_wrappers
Enable GN check for webrtc/{p2p,system_wrappers}
2017-04-21 12:17:08 +00:00
test
Reland of GN: Enable ARC for Mac and iOS in rtc_* templates (patchset #1 id:1 of https://codereview.webrtc.org/2827223003/ )
2017-04-24 07:57:16 +00:00
tools
Add network tester client [android] to be able to test mobile networks in terms of packet size and sending rates.
2017-04-20 13:56:27 +00:00
video
Only increment scale counter and adaptation stats (and store last_adaptation_request_) if sink_wants_ is updated.
2017-04-21 08:47:31 +00:00
voice_engine
Fix invalid output buffer usage
2017-04-20 22:19:10 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
BUILD.gn
Reland of Loosening the coupling between WebRTC and //third_party/protobuf (patchset #1 id:1 of https://codereview.webrtc.org/2786363002/ )
2017-04-07 07:59:12 +00:00
call.h
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
common_types.cc
Add read support of RtpStreamId/RepairedRtpStreamId header extensions.
2017-04-19 09:59:48 +00:00
common_types.h
Add read support of RtpStreamId/RepairedRtpStreamId header extensions.
2017-04-19 09:59:48 +00:00
config.cc
Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2809653004/ )
2017-04-11 17:34:31 +00:00
config.h
Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2809653004/ )
2017-04-11 17:34:31 +00:00
DEPS
Reland of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #1 id:1 of https://codereview.webrtc.org/2794033002/ )
2017-04-05 10:02:20 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete unused spreadsort implementation.
2016-12-05 11:03:26 +00:00
OWNERS
Cleanup unused rules in webrtc/DEPS + add kjellander to OWNERS for it
2016-11-29 10:52:22 +00:00
PATENTS
…
PRESUBMIT.py
PRESUBMIT.py: accept variants on the copyright message that are present in the codebase.
2014-05-23 17:27:18 +00:00
README.chromium
…
typedefs.h
Make WARN_UNUSED_RESULT a no-op on gcc
2017-04-06 11:32:27 +00:00
video_decoder.h
Reland of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #1 id:1 of https://codereview.webrtc.org/2794033002/ )
2017-04-05 10:02:20 +00:00
video_encoder.h
Reland of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #1 id:1 of https://codereview.webrtc.org/2794033002/ )
2017-04-05 10:02:20 +00:00
video_frame.h
Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2809653004/ )
2017-04-11 17:34:31 +00:00
video_receive_stream.h
Delete VieRemb class, move functionality to PacketRouter.
2017-04-19 06:38:35 +00:00
video_send_stream.h
Add number of quality adapt changes to VideoSendStream stats.
2017-04-04 12:51:49 +00:00
webrtc.gni
Reland of GN: Enable ARC for Mac and iOS in rtc_* templates (patchset #1 id:1 of https://codereview.webrtc.org/2827223003/ )
2017-04-24 07:57:16 +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.