generic-library/webrtc
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity
3,729 Commits 1 Branch 0 Tags 108 MiB
fee739c224
Commit Graph

34 Commits

Author SHA1 Message Date
fischman@webrtc.org
fe6b57187d AppRTCDemo: delete hosted android_channel.html now that it's no longer necessary. 
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. 
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. 
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. 
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. 
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. 
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. 
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. 
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. 
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 
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. 
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
pbos@webrtc.org
b4a0623e43 Fix of lint script errors in apprtc.py 
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. 
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 
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. 
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 
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
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. 
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
braveyao@webrtc.org
488d4c9493 Submit symlink in apprtc from Linux since it fails from Win 
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 
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 
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