braveyao@webrtc.org
a19333954d
Apprtc CSS: Add flip to local view of FireFox and remove warning of Canary
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BUG=1380
TEST=Manual Test
R=juberti@google.com
Review URL: https://webrtc-codereview.appspot.com/1620004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4225 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-13 03:49:03 +00:00
fischman@webrtc.org
fe6b57187d
AppRTCDemo: delete hosted android_channel.html now that it's no longer necessary.
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This file became redundant with 47220050 which rolled to libjingle opensource in
r327 of talk/examples/android/src/org/appspot/apprtc/GAEChannelClient.java
R=vikasmarwaha@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1606004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4206 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-10 17:22:50 +00:00
braveyao@webrtc.org
5ed7051799
Apprtc: not to start the call until we get Turn response.
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BUG=1795
Test=Manual Test
R=fischman@webrtc.org , juberti@google.com
Review URL: https://webrtc-codereview.appspot.com/1528004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4144 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-31 06:29:41 +00:00
vikasmarwaha@webrtc.org
fddf6be339
Updated apprtc to use new TURN format for chrome versions M28 & above.
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R=dutton@google.com , juberti@google.com
Review URL: https://webrtc-codereview.appspot.com/1563004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4139 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-29 22:13:19 +00:00
braveyao@webrtc.org
5f8f112a7b
Not to request to TURN server for local tests. Follow-up work to issue1197.
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BUG=1197
TEST=Manual test
R=dutton@google.com
Review URL: https://webrtc-codereview.appspot.com/1340004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4083 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-22 07:27:05 +00:00
fischman@webrtc.org
5e2a1bbbc6
AppRTC: make requestTurn() failure non-fatal to call establishment.
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BUG=1795
R=vikasmarwaha@google.com , vikasmarwaha@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1504005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4063 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-17 18:32:23 +00:00
vikasmarwaha@webrtc.org
59a06670b5
Updated apprtc demo to interop with firefox.
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R=juberti@google.com
Review URL: https://webrtc-codereview.appspot.com/1482004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4040 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-16 01:05:19 +00:00
vikasmarwaha@webrtc.org
40298d452c
Added webaudio-and-webtrc.html to the demos index.html.
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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
vikasmarwaha@webrtc.org
1993a559e8
Added Stereo url paramter to apprtc demo.
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R=dutton@google.com
Review URL: https://webrtc-codereview.appspot.com/1418004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4013 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-13 18:48:09 +00:00
henrika@webrtc.org
7a5615bc84
New WebAudio-WebRTC demo.
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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.
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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
braveyao@webrtc.org
a39a8fec16
Add owner to Apprtc
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Review URL: https://webrtc-codereview.appspot.com/1328007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3874 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-19 02:34:45 +00:00
andrew@webrtc.org
ceaedc0014
Remove executable bit from dc1.html.
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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.
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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.
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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
pbos@webrtc.org
b4a0623e43
Fix of lint script errors in apprtc.py
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BUG=
Review URL: https://webrtc-codereview.appspot.com/1285007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3780 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-08 15:59:24 +00:00
hta@webrtc.org
37bf5847dc
Show stats from both sides
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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
vikasmarwaha@webrtc.org
222e9948f5
Migrating Apprtc to use new TURN service which supports time-limited TURN credentials.
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Review URL: https://webrtc-codereview.appspot.com/1291004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3773 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-06 05:58:15 +00:00
hta@webrtc.org
3ed599adb5
Bandwidth stats display in constraints-and-stats.
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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
braveyao@webrtc.org
f354e1f587
Add audio/video only option in apprtc
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ISSUE = issue 1507
TEST =
Review URL: https://webrtc-codereview.appspot.com/1216007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3692 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-20 00:23:55 +00:00
vikasmarwaha@webrtc.org
ebf49da9b2
Url option to change the resolution.
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Review URL: https://webrtc-codereview.appspot.com/1218005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3691 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-19 22:15:55 +00:00
hta@webrtc.org
ecfd32880e
Changed stats reporting to not use local/remote
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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.
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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.
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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
fischman@webrtc.org
a33037ea6c
Added an android_channel.html reflector page to allow Android apps to use a
...
WebView to speak the Channel API from Google AppEngine.
BUG=webrtc:1169
Review URL: https://webrtc-codereview.appspot.com/1145006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3644 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-10 18:28:08 +00:00
wu@webrtc.org
3137a21068
Dtmf twinkle-twinkle.
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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.
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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
braveyao@webrtc.org
488d4c9493
Submit symlink in apprtc from Linux since it fails from Win
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Review URL: https://webrtc-codereview.appspot.com/1169004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3622 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-06 06:45:14 +00:00
braveyao@webrtc.org
07db4a6918
Add symlink of adapter.js from apprtc to base
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Review URL: https://webrtc-codereview.appspot.com/1160004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3621 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-06 03:35:03 +00:00
hta@webrtc.org
db3f42782c
Using adapter.js and getRemoteStreams
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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.
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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.
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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.
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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.
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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.
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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