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19 Commits

Author SHA1 Message Date
braveyao@webrtc.org
bc0470f559 AppRTC Sample: Switch AppRTC to use RTCIceServer.urls. 
BUG=2832
TEST=Manual Test
R=juberti@google.com

Review URL: https://webrtc-codereview.appspot.com/7739004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5599 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-24 03:43:03 +00:00
braveyao@webrtc.org
37c2976511 Samples, add IPv6 supporting into Apprtc demo. 
BUG=2828
TEST=Manual Test
R=juberti@google.com

Review URL: https://webrtc-codereview.appspot.com/7509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5432 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-27 03:08:16 +00:00
andresp@webrtc.org
8c5b27de9a Allow to skip turn by passing ts=false to apprtc. 
R=braveyao@webrtc.org, fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5384 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-14 17:00:23 +00:00
braveyao@webrtc.org
c329529047 Apply transaction to setting connected to Room entities, to resolve a possible race condition at two clients connecting simultaneously. 
BUG = 1742
Test = Apprtc Integration Test

R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5247 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-09 19:37:45 +00:00
braveyao@webrtc.org
54e8bfafba Apprtc demo: add DSCP support. 
BUG=2669
TEST=Manual Test
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5194 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-29 02:38:20 +00:00
phoglund@webrtc.org
03c7a35ac0 Fixing long lines in apprtc.py. 
These long lines causes the presubmit to get angry.

BUG=webrtc:2678
R=braveyao@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5193 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-28 17:45:08 +00:00
andrew@webrtc.org
20078e2f9b Support video constraints and use key/value pairs. 
- Remove the minre and maxre parameters in favour of setting video
constraints directly.
- In order to support non-boolean values, have constraints passed as
key/value pairs, rather than the leading "-" syntax used earlier to
specify false.

TESTED=Verified that setting various audio and video constraints has
the desired effect, including "true" and "false". Verified that the "hd"
parameter still works.

R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2360005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4931 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-05 02:26:50 +00:00
andrew@webrtc.org
bab2aa5113 Add audio and video parameters for setting media constraints. 
- These replace the media parameter, now removed.
- Organize the parameter getting a bit.

To describe the new parameters, I'll just copy the code comments here:
Use "audio" and "video" to set the media stream constraints. "true" and
"false" are recognized and interpreted as bools, for example:
  "?audio=true&video=false" (start an audio-only call).
  "?audio=false" (start a video-only call)
If unspecified, the constraint defaults to True.

audio-specific parsing:
To set certain constraints, pass in a comma-separated list of audio
constraint strings. If preceded by a "-", the constraint will be set to
False, and otherwise to True. There is no validation of constraint
strings. Examples:
  "?audio=googEchoCancellation" (enables echo cancellation)
  "?audio=-googEchoCancellation,googAutoGainControl" (disables echo
      cancellation and enables gain control)

TESTED=Verified that passing true, false and various audio constraints
has the desired effect in apprtc.

R=vikasmarwaha@google.com

Review URL: https://webrtc-codereview.appspot.com/2345004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4919 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-03 22:37:29 +00:00
vikasmarwaha@webrtc.org
cee0dfb57a Made sure that DTLS/SRTP is set to false in apprtc demo when testing loopback. See crbug/294881 for details. 
R=juberti@google.com, mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2268004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4805 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-20 21:26:07 +00:00
wu@webrtc.org
bc189fb3b9 * Prefer to send ISAC on clank. 
* Add url option asc and arc to allow setting preferred audio send/receive codec.

TESTED=mobile as caller and callee:
pc-n7: pc sends opus, n7 sends isac 
pc-n4: pc sends opus, n4 sends isac
pc-pc opus-opus

R=braveyao@webrtc.org, juberti@google.com

Review URL: https://webrtc-codereview.appspot.com/2196006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4742 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-13 20:11:47 +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. 
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
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
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
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
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
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
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
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