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Author SHA1 Message Date
dutton@google.com
fe165ded46 Added warning for Github move ****THESE_FILES_ARE_MOVING**** 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5837 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-03 19:57:06 +00:00
fischman@webrtc.org
ccb33a67b9 turn-prober: enable running headlessly and only emit output on error. 
With these changes I have the script running in a 10m cronjob on my desktop and
emailing me on failure.  (extremely poor man's monitoring; still, baby steps)

BUG=2187
R=juberti@google.com

Review URL: https://webrtc-codereview.appspot.com/9659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5709 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-17 16:27:41 +00:00
fischman@webrtc.org
bf88eccf33 Added turn-prober.sh: a super-simple prober for TURN servers & candidates. 
BUG=2187
R=juberti@google.com, juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5604 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-24 21:52:59 +00:00
wu@webrtc.org
78ea3d50e0 Check pcConfig (which can be null) before use. 
BUG=

TEST=manully with pc1.html
R=juberti@google.com

Review URL: https://webrtc-codereview.appspot.com/9079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5603 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-24 21:51:58 +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
hta@webrtc.org
1009798b31 Demo of multi-pass encode - used for testing limits. 
This demo creates a sequence of PeerConnections, and passes
a videostream through all of them.
This allows one to test how many PeerConnections and how
many encodes/decodes the implementation will support before
breaking down or slowing down enough to be unusable.

BUG=
R=fischman@webrtc.org, hta@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5557 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-15 06:13:41 +00:00
vikasmarwaha@webrtc.org
c5a839c3a9 Updated demos so they work on FF, the createOffer api cannot have null parameters according to spec. Same applies to createAnswer. 
R=juberti@google.com

Review URL: https://webrtc-codereview.appspot.com/8219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5503 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-07 19:08:38 +00:00
vikasmarwaha@webrtc.org
b307e86076 Updated demos to use the sucess and failure callback in addIceCandidate api. 
R=dutton@google.com

Review URL: https://webrtc-codereview.appspot.com/7969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5497 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-06 22:38:37 +00:00
juberti@webrtc.org
5db9a3f32a Added new create-offer and ice-servers demos to test the exact output of createOffer and .onicecandidate. 
Updated a few demos to work on Firefox.

R=dutton@google.com

Review URL: https://webrtc-codereview.appspot.com/1581006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5464 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-30 23:38:44 +00:00
vikasmarwaha@webrtc.org
ecc96af15b Expose errors in apprtc demo to div. Currently the errors only show in the console, the CL attempts to expose critical errors on to the div element. 
BUG=2786
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7539005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5443 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-27 21:13:54 +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
24999d44c2 Allow ?audio=false&video=false to be used in combination to instantiate a recv-only client. 
R=braveyao@webrtc.org, juberti@google.com

Review URL: https://webrtc-codereview.appspot.com/6819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5425 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-24 12:25:50 +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
vikasmarwaha@webrtc.org
bb0de3ca9f Updated Demos so they work on FF, changed the third argument in CreateOffer to null as it doesnot really require sdpConstraints. 
R=juberti@google.com

Review URL: https://webrtc-codereview.appspot.com/6769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5356 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-09 00:51:19 +00:00
fischman@webrtc.org
0b7d8e6fcb AppRTC: Alert the user to failure to acquire TURN server. 
Hopefully will result in quicker turnaround time for CEOD/turnserver fixes.
Might trigger undesirable levels of bogus/spammy/unhelpful/PEBCAK reports to
discuss-webrtc, in which case I'll remove the second part of the message.

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

Review URL: https://webrtc-codereview.appspot.com/4779005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5343 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-06 23:46:53 +00:00
vikasmarwaha@webrtc.org
7bdaf837d4 Updated PeerConnection samples so they run on FF. 
R=braveyao@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5340 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-03 23:13:01 +00:00
vikasmarwaha@webrtc.org
a63fc87139 Fix JS error in adapter.js for FF for the case when ?transport=xxx is missing in TURN url. 
BUG=2737
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5331 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-20 22:10:17 +00:00
hta@webrtc.org
df02283279 Adds audio volume demo to the index page. 
BUG=
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5589005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5263 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-11 14:44:10 +00:00
hta@webrtc.org
26c40ba166 Removed audio element from volume measuring demo. 
This removes the possibility of feedback loops, which can happen if you
run this demo on an Android device.

BUG=
R=dutton@google.com

Review URL: https://webrtc-codereview.appspot.com/5589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5258 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-11 11:12:39 +00:00
hta@webrtc.org
1133ffda4b Merged OWNERS of JS demo directories 
This allows Sam Dutton to maintain code samples, and demo managers to
modify js/base/adapter.js.

BUG=
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5549006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5257 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-11 08:51:56 +00:00
hta@webrtc.org
c4038d795d Rewriting the SoundMeter class to be RMS and be encapsulated differently 
This CL changes the SoundMeter to be root-mean-square.
It also changes the interface between the meter and the display to be based on the display calling down to the meter rather than the meter calling up to the display.

A graphic display of the results is also added.

BUG=
R=cwilso@google.com, dutton@google.com, henrika@webrtc.org, juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5256 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-11 08:36:16 +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
hta@webrtc.org
758db4baea Demo showing how to measure volume using WebAudio 
This adds a page to the demos page, it does not affect any running code.

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

Review URL: https://webrtc-codereview.appspot.com/5099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5237 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-06 14:47:34 +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
wu@webrtc.org
aa74b5d690 Add success/error callback to set sdp calls. 
Add a workaround for crbug/322756 to append a line break to the end of sdp if needed.

R=juberti@webrtc.org, vikasmarwaha@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5167 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-23 00:37:50 +00:00
vikasmarwaha@webrtc.org
442c5e47cd Update adapter.js to use TURN transport parameters for FF version 27 & above. 
R=juberti@google.com

Review URL: https://webrtc-codereview.appspot.com/2829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5031 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-24 20:31:57 +00:00
vikasmarwaha@webrtc.org
d674a566d3 Update dc1 demo as it was using invalid data Constraint (Reliable:true) for SCTP. The constraint Reliable is not supported by Standard and ignored in our implementation. See issue 2511. 
R=dutton@google.com, jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2739004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5030 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-24 19:38:47 +00:00
vikasmarwaha@webrtc.org
90d8719fd7 Radix should be specified when calling ParseInt function in adapter.js. Refer to issue 2490. 
R=dutton@google.com, juberti@google.com

Review URL: https://webrtc-codereview.appspot.com/2709006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5017 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-22 18:02:41 +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
ee6d0ddbe6 Upload Demo page to allow edit offer & Answer sdp in pc1 demo. 
R=dutton@google.com

Review URL: https://webrtc-codereview.appspot.com/2296004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4895 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-01 18:43:07 +00:00
vikasmarwaha@webrtc.org
19134bae95 Updated device-switch demo page to work with Chrome M30. 
BUG=2218
R=braveyao@webrtc.org, dutton@google.com

Review URL: https://webrtc-codereview.appspot.com/2025004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4892 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-01 17:02:32 +00:00
vikasmarwaha@webrtc.org
7a7b929882 Updated dc1.html to support SCTP transport. 
R=dutton@google.com

Review URL: https://webrtc-codereview.appspot.com/2058004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4814 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-23 18:03:33 +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
braveyao@webrtc.org
a80ee74f69 AppRTC: using a footer element instead of div#footer in CSS. 
R=dutton@google.com

Review URL: https://webrtc-codereview.appspot.com/2200004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4724 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-11 16:24:07 +00:00
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 
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 
- 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 
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. 
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. 
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. 
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. 
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. 
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. 
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. 
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. 
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. 
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