houssainy@google.com
fce8f5d319
NOTE: This code review based on the running issue:
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https://webrtc-codereview.appspot.com/24939004/
R=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7499 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-22 17:24:20 +00:00
houssainy@google.com
0371a37f85
Moving creating TURN configration to the host machine instead of the bots - rtcBot
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R=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24939004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7468 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-17 16:43:50 +00:00
houssainy@google.com
3e2f8ff36c
Selecting bot_type changed to be specified in the test file
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Selecting bot_type changed to be specified in the test file instead of
specify it in the running command.
Now we can write test for rtcBot that run one bot on chrome for android
and the other bot on chrome for desktop.
R=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23069004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7458 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-15 15:01:11 +00:00
houssainy@google.com
24f62e1a28
Adding getStats function to the exposed PeerConnection in RtcBot
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Exposed Peerconnection object has new function "getStats". This function
returns the stats as array of reports, and each report is RTCStatReport
with additional attributes names and stats.
names: array of all the stat names in current report.
Stats: dictionary and the key is the stat name, and value is the value
of this stat.
R=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23779004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7319 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-29 09:36:28 +00:00
houssainy@google.com
07ca949346
Adding webrtc_video_streaming test
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This test is streaming video and audio between two bots using webrtc js api.
R=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28469004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7261 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-22 13:52:39 +00:00
houssainy@google.com
04b853b56a
Bot Browser files moved to /bot/browser/
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because android files will be a different and will need to add more files for Android.
There was a CL to move the browser files form bot/browser/ to /bot/ as i thought it will be the same files used for Android.
R=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7119 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-09 14:50:09 +00:00
houssainy@google.com
38ef664418
Moving the api.js and bot.js to /rtcbot/bot/ to be shared between
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/borwser and /android
R=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25369004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7062 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 13:44:47 +00:00
andresp@webrtc.org
468516c959
RTCBot is a framework that allows to write tests where logic runs on a single
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host that controls multiple endpoints ("bots"). Thus allowing to create more
complex scenarios that would otherwise require non-trival signalling between
multiple parties.
R=houssainy@google.com , phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22239004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7021 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-02 10:52:54 +00:00