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24 Commits

Author SHA1 Message Date
vikasmarwaha@webrtc.org
ee6d0ddbe6 Upload Demo page to allow edit offer & Answer sdp in pc1 demo. 
R=dutton@google.com

Review URL: https://webrtc-codereview.appspot.com/2296004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4895 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-01 18:43:07 +00:00
vikasmarwaha@webrtc.org
19134bae95 Updated device-switch demo page to work with Chrome M30. 
BUG=2218
R=braveyao@webrtc.org, dutton@google.com

Review URL: https://webrtc-codereview.appspot.com/2025004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4892 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-01 17:02:32 +00:00
vikasmarwaha@webrtc.org
7a7b929882 Updated dc1.html to support SCTP transport. 
R=dutton@google.com

Review URL: https://webrtc-codereview.appspot.com/2058004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4814 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-23 18:03:33 +00:00
hta@webrtc.org
cc39484770 IP address display from stats. 
This CL demonstrates a couple of methods to extract more complex properties from the stats that are linked via stats IDs.

RISK=P3
TESTED=manual test
BUG=
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1667005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4584 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-21 17:00:54 +00:00
mcasas@webrtc.org
d4d9480c05 Added gum4.html, a multiple camera opening demo, each opening with a different resolution and/or frame rate. 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4300 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-05 09:12:04 +00:00
vikasmarwaha@webrtc.org
40298d452c Added webaudio-and-webtrc.html to the demos index.html. 
R=dutton@google.com, henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1425005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4039 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-16 00:50:38 +00:00
henrika@webrtc.org
7a5615bc84 New WebAudio-WebRTC demo. 
Capture microphone input and stream it out to a peer with a processing effect applied to the audio.

The audio stream is: 

o Recorded using live-audio input.
o Filtered using an HP filter with fc=1500 Hz.
o Encoded using Opus.
o Transmitted (in loopback) to remote peer using RTCPeerConnection where it is decoded.
o Finally, the received remote stream is used as source to an <audio> tag and played out locally.

Press any key to add an effect to the transmitted audio while talking.

Please note that: 

o Linux is currently not supported.
o Sample rate and channel configuration must be the same for input and output sides on Windows.
o Only the Default microphone device can be used for capturing.

R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1256004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4006 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-13 09:29:13 +00:00
vikasmarwaha@webrtc.org
77ac84814d Added new demo states.html & updated existing demos to work on firefox. 
Review URL: https://webrtc-codereview.appspot.com/1327007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3905 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-25 23:22:03 +00:00
andrew@webrtc.org
ceaedc0014 Remove executable bit from dc1.html. 
Review URL: https://webrtc-codereview.appspot.com/1320010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3867 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-18 01:56:07 +00:00
hta@webrtc.org
f1bf3a00b2 A device switcher code example, with fake. 
This demo shows the usage of the proposed getDeviceInfo call and its
associatied permissions model.

Review URL: https://webrtc-codereview.appspot.com/1320008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3862 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-17 14:24:21 +00:00
vikasmarwaha@webrtc.org
4c44fe0561 Updated pranswer, dtmf demos & deleted pc1-deprecated.html. 
Review URL: https://webrtc-codereview.appspot.com/1287007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3783 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-08 21:23:58 +00:00
hta@webrtc.org
37bf5847dc Show stats from both sides 
This change shows the stats generated both at the sending PeerConnection
and at the receiving PeerConnection.

BUG=

Review URL: https://webrtc-codereview.appspot.com/1290005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3774 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-06 10:05:55 +00:00
hta@webrtc.org
3ed599adb5 Bandwidth stats display in constraints-and-stats. 
Also shows off the report type and ID field, and logs less useless info.

BUG=

Review URL: https://webrtc-codereview.appspot.com/1212007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3706 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-22 08:48:16 +00:00
hta@webrtc.org
ecfd32880e Changed stats reporting to not use local/remote 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1216004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3688 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-19 08:45:47 +00:00
vikasmarwaha@webrtc.org
eddc5a6654 Updated local-audio-rendering.html to remove unmute. 
Review URL: https://webrtc-codereview.appspot.com/1193004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3670 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-14 23:34:19 +00:00
vikasmarwaha@webrtc.org
da0f7086e1 Update demos to have local audio control muted by default. 
Review URL: https://webrtc-codereview.appspot.com/1160007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3649 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-11 16:58:07 +00:00
wu@webrtc.org
3137a21068 Dtmf twinkle-twinkle. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1160005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3635 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-07 21:59:23 +00:00
phoglund@webrtc.org
5d37139374 Fixed a ton of Python lint errors, enabled python lint checking. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1166004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3627 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-07 09:59:43 +00:00
hta@webrtc.org
db3f42782c Using adapter.js and getRemoteStreams 
Needed to make the stats demo work on M26.

BUG=

Review URL: https://webrtc-codereview.appspot.com/1165004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3612 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-05 15:23:40 +00:00
vikasmarwaha@webrtc.org
a856db26a6 Moved trace function to adapter.js and removed from pc1 & multiple.html. 
Review URL: https://webrtc-codereview.appspot.com/1156005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3608 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-05 03:35:26 +00:00
vikasmarwaha@webrtc.org
7881b574dd Updated path of adapter.js for dtmf & pc1-audio demos. 
TBR = wu@webrtc.org,juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1151005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3606 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-05 02:04:07 +00:00
vikasmarwaha@webrtc.org
99f13464df Typo in index.html and updated svn propset for dtmf & pc1-audio demos. 
Review URL: https://webrtc-codereview.appspot.com/1145007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3603 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-04 19:34:46 +00:00
vikasmarwaha@webrtc.org
b203540e25 Redirect webrtc-demos.appspot.com to svn site and added dtmf & pc1-audio demos. Also updated index page to include information about new demos. 
Review URL: https://webrtc-codereview.appspot.com/1148004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3602 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-04 18:57:09 +00:00
vikasmarwaha@webrtc.org
98fce15c6f Adding webrtc-sample demos under trunk/samples. 
Review URL: https://webrtc-codereview.appspot.com/1126005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3578 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-27 23:22:10 +00:00