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32 Commits

Author SHA1 Message Date
hta@webrtc.org
1009798b31 Demo of multi-pass encode - used for testing limits. 
This demo creates a sequence of PeerConnections, and passes
a videostream through all of them.
This allows one to test how many PeerConnections and how
many encodes/decodes the implementation will support before
breaking down or slowing down enough to be unusable.

BUG=
R=fischman@webrtc.org, hta@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5557 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-15 06:13:41 +00:00
vikasmarwaha@webrtc.org
c5a839c3a9 Updated demos so they work on FF, the createOffer api cannot have null parameters according to spec. Same applies to createAnswer. 
R=juberti@google.com

Review URL: https://webrtc-codereview.appspot.com/8219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5503 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-07 19:08:38 +00:00
vikasmarwaha@webrtc.org
b307e86076 Updated demos to use the sucess and failure callback in addIceCandidate api. 
R=dutton@google.com

Review URL: https://webrtc-codereview.appspot.com/7969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5497 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-06 22:38:37 +00:00
juberti@webrtc.org
5db9a3f32a Added new create-offer and ice-servers demos to test the exact output of createOffer and .onicecandidate. 
Updated a few demos to work on Firefox.

R=dutton@google.com

Review URL: https://webrtc-codereview.appspot.com/1581006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5464 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-30 23:38:44 +00:00
vikasmarwaha@webrtc.org
bb0de3ca9f Updated Demos so they work on FF, changed the third argument in CreateOffer to null as it doesnot really require sdpConstraints. 
R=juberti@google.com

Review URL: https://webrtc-codereview.appspot.com/6769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5356 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-09 00:51:19 +00:00
vikasmarwaha@webrtc.org
7bdaf837d4 Updated PeerConnection samples so they run on FF. 
R=braveyao@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5340 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-03 23:13:01 +00:00
hta@webrtc.org
26c40ba166 Removed audio element from volume measuring demo. 
This removes the possibility of feedback loops, which can happen if you
run this demo on an Android device.

BUG=
R=dutton@google.com

Review URL: https://webrtc-codereview.appspot.com/5589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5258 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-11 11:12:39 +00:00
hta@webrtc.org
c4038d795d Rewriting the SoundMeter class to be RMS and be encapsulated differently 
This CL changes the SoundMeter to be root-mean-square.
It also changes the interface between the meter and the display to be based on the display calling down to the meter rather than the meter calling up to the display.

A graphic display of the results is also added.

BUG=
R=cwilso@google.com, dutton@google.com, henrika@webrtc.org, juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5256 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-11 08:36:16 +00:00
hta@webrtc.org
758db4baea Demo showing how to measure volume using WebAudio 
This adds a page to the demos page, it does not affect any running code.

BUG=
R=dutton@google.com, phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5237 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-06 14:47:34 +00:00
wu@webrtc.org
aa74b5d690 Add success/error callback to set sdp calls. 
Add a workaround for crbug/322756 to append a line break to the end of sdp if needed.

R=juberti@webrtc.org, vikasmarwaha@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5167 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-23 00:37:50 +00:00
vikasmarwaha@webrtc.org
d674a566d3 Update dc1 demo as it was using invalid data Constraint (Reliable:true) for SCTP. The constraint Reliable is not supported by Standard and ignored in our implementation. See issue 2511. 
R=dutton@google.com, jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2739004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5030 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-24 19:38:47 +00:00
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
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
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
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