braveyao@webrtc.org
641340944b
Show the signaling state and ice connection state in AppRTC by hooking up the peerconnections .onsignalingstatechange and .oniceconnectionstatechange events.
...
Hopefully this will increase the quality of the "it does not work" reports from users by giving them more information about what is going on under the hood.
R=juberti@google.com
Review URL: https://webrtc-codereview.appspot.com/2174004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4718 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-10 17:37:16 +00:00
braveyao@webrtc.org
be588f9a58
Apprtc Demo: calling createOffer/Answer without failureCallback is deprecated in FF
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BUG=2313
Test=Manual test
R=dutton@google.com , juberti@google.com , vikasmarwaha@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2175004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4683 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-05 16:44:55 +00:00
fischman@webrtc.org
4498d013f6
apprtc: rationalize whitespace
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- Remove ^M DOS line endings
- Remove trailing whitespace
- Remove leading 2-space indents from files that have carried this indent since their contents was removed from within enclosing contexts that required it.
- Add a newline to avoid 82-column line.
R=vikasmarwaha@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2112004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4619 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-26 18:01:28 +00:00
fischman@webrtc.org
5a035b4279
apprtc: add ctrl+i Info window showing gathered ICE candidate types
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R=vikasmarwaha@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2109004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4617 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-26 17:44:38 +00:00
hta@webrtc.org
cc39484770
IP address display from stats.
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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
vikasmarwaha@webrtc.org
83ffb0dd5c
Added functionality in apprtc demo to close the capture device on hangup.
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BUG=1589
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2018004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4540 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-13 17:53:37 +00:00
mallinath@webrtc.org
5a27e49f35
This CL will add support of passing all turn urls returned by the CEOD to PeerConnection object.
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R=juberti@webrtc.org , vikasmarwaha@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4506 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-08 19:52:08 +00:00
vikasmarwaha@webrtc.org
6e7c203aee
Modified apprtc demo code to detect browser by checking user_agent in apprtc.py. Now we will use Mozilla stun server if FF is detected as the browser. The CL is an improvement to r4388.
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R=braveyao@webrtc.org , dutton@google.com
Review URL: https://webrtc-codereview.appspot.com/1928004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4491 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-05 22:05:20 +00:00
braveyao@webrtc.org
10bbfeff5b
Apprtc: add 'event' parameter to onkeydown event handler.
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BUG=
TEST=Manual test
R=dutton@google.com
Review URL: https://webrtc-codereview.appspot.com/1898005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4425 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-30 09:27:49 +00:00
vikasmarwaha@webrtc.org
b63c29f48c
Minor bug fix in r4388, had to change pc_config variable to pcConfig for apprtc demo.
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R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1856004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4389 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-23 23:13:35 +00:00
vikasmarwaha@webrtc.org
59fb7a60f2
Use Mozilla STUN server in apprtc demo for FF. Currently FF cannot work with Google STUN server as it expects XOR-MAPPED address while Google STUN server provides MAPPED address.
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R=dutton@google.com
Review URL: https://webrtc-codereview.appspot.com/1849004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4388 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-23 22:06:51 +00:00
mcasas@webrtc.org
d4d9480c05
Added gum4.html, a multiple camera opening demo, each opening with a different resolution and/or frame rate.
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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
bb25256775
Added changes in apprtc demo to ignore turn address through query string for FF. Also made sure that the iceServers array doesnot include transport parameter in turn url for FF. Finally removed turn: from the turn_url when creating iceservers for pre-M28 chrome.
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R=dutton@google.com , juberti@google.com
Review URL: https://webrtc-codereview.appspot.com/1627006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4259 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-25 14:52:51 +00:00
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.
...
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
...
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.
...
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