generic-library/webrtc
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity
5,431 Commits 1 Branch 0 Tags 108 MiB
f3a5e6afc4
Commit Graph

21 Commits

Author SHA1 Message Date
kjellander@webrtc.org
52fd65b16a Partial revert of "Removing samples directory following move to Github" 
Reason: Unfortunately we depend on AppRTC being in this location
for the bots in our Chromium WebRTC waterfalls so I'm reverting
this until we've solved that dependency.

This reverts apprtc and adapter.js from being removed in r5871.

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

Review URL: https://webrtc-codereview.appspot.com/11529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5873 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-09 13:52:24 +00:00
dutton@google.com
7ecc142d6b Removing samples directory following move to Github 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5871 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-09 09:55:54 +00:00
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