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1732a591e7
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65 Commits

Author SHA1 Message Date
tkchin@webrtc.org
1732a591e7 Add a UIView for rendering a video track. 
RTCEAGLVideoView provides functionality to render a supplied RTCVideoTrack using OpenGLES2.

R=fischman@webrtc.org
BUG=3188

Review URL: https://webrtc-codereview.appspot.com/12489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6192 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-19 23:26:01 +00:00
fischman@webrtc.org
a150bc9bbf PeerConnection(android): allow initializing either (or neither) of {Voice,Video}Engine. 
Enables applications that don't want to pay the init/startup cost or request
extra permissions (e.g. audio-only app, or DataChannel-only app).

BUG=3234

Review URL: https://webrtc-codereview.appspot.com/15489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6164 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 22:00:50 +00:00
fischman@webrtc.org
14ea7e8922 AppRTCDemo(android): added a Heads-Up Display for bandwidth estimation. 
- tap display to toggle visibility
- increased getStats frequency to 1hz.

R=glaznev@google.com

Review URL: https://webrtc-codereview.appspot.com/19419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6039 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-01 20:57:55 +00:00
fischman@webrtc.org
dd92feb6dd AppRTCDemo(android): send the created SDP, not the local description after setting it 
This is required to allow explicit filtering of ICE candidates.

BUG=3288
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6038 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-01 19:06:18 +00:00
tkchin@webrtc.org
ff2733204d Implement ObjC DataChannel wrapper 
R=fischman@webrtc.org
BUG=3112

Review URL: https://webrtc-codereview.appspot.com/16369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6031 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-30 18:32:33 +00:00
fischman@webrtc.org
7c82adae61 AppRTCDemo was blocking the main thread for network requests. This fixes it by making the background queue serial instead of using @synchronize to make the background operations serial. 
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16379004

Patch from Bridger Maxwell <bridgeyman@gmail.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6028 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-30 00:17:47 +00:00
fischman@webrtc.org
f27fdeb9c9 AppRTCDemo(android): don't initialize process-globals more than once. 
BUG=3257
R=braveyao@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6001 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-28 16:32:38 +00:00
mallinath@webrtc.org
a0d3067575 Use CreatePeerConnection method which accepts port_allocator. 
Other method will be removed, in a different CL.

R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20369006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5987 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-26 00:00:15 +00:00
tkchin@webrtc.org
19b1be159e Provide GetStats method in RTCPeerConnection 
BUG=3144
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12069006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5960 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-22 21:05:38 +00:00
tkchin@webrtc.org
ec3d8ecdcc Fix typo by renaming RTCSessionDescriptonDelegate -> RTCSessionsDescriptionDelegate 
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12059004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5946 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-21 18:47:24 +00:00
fischman@webrtc.org
d1fe6b728e AppRTCDemo(android): fix a couple of SDP-related regressions. 
- r5834 made it so that empty fields are a fatal SDP parsing error, exposing
  opportunities for improvement in the preferISAC; changed split/join to use
  \r\n instead of \n and now omitting the trailing space on the m=audio line
  that triggered the new failure.
- DTLS requires a different role for each endpoint so conflicts with loopback
  calling.  apprtc.py suppresses DTLS for that reason in loopback calls, so the
  android demo app now only enables DTLS by default if it is not suppressed by a
  constraint (matching Chrome).

BUG=3164,3165,2507
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5847 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-04 21:40:46 +00:00
fischman@webrtc.org
49c5ba32bb AppRTCDemo(iOS): now works in the iOS Simulator! 
...which has no camera device emulation or pass-through, so no local video
view.

R=noahric@google.com

Review URL: https://webrtc-codereview.appspot.com/10919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5815 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-31 20:22:19 +00:00
fischman@webrtc.org
61e78fca6c AppRTCDemo(iOS): remote-video reliability fixes 
Previously GAE Channel callbacks would be handled by JS string-encoding the
payload into a URL.  Unfortunately this is limited to the (undocumented,
silently problematic) maximum URL length UIWebView supports.  Replaced this
scheme by a notification from JS to ObjC and a getter from ObjC to JS (which
happens out-of-line to avoid worrying about UIWebView's re-entrancy, or lack
thereof).  Part of this change also moved from a combination of: JSON,
URL-escaping, and ad-hoc :-separated values to simply JSON.

Also incidentally:
- Removed outdated TODO about onRenegotiationNeeded, which is unneeded
- Move handling of PeerConnection callbacks to the main queue to avoid having
  to think about concurrency too hard.
- Replaced a bunch of NSOrderedSame with isEqualToString for clearer code and
  not having to worry about the fact that [nil compare:@"foo"]==NSOrderedSame
  is always true (yay ObjC!).
- Auto-scroll messages view.

BUG=3117
R=noahric@google.com

Review URL: https://webrtc-codereview.appspot.com/10899006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5814 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-31 20:16:49 +00:00
fischman@webrtc.org
fe16488184 AppRTCDemo(android): specify DtlsSrtpKeyAgreement:true in CreatePeerConnection's constraints. 
This is required to interop with Chrome now that SDES is disabled in Chrome (as of r5640).

BUG=2774
R=jiayl@chromium.org

Review URL: https://webrtc-codereview.appspot.com/10749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5809 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-28 19:58:03 +00:00
henrike@webrtc.org
1ca08f65e3 Fix after auto update in r5787. APPRTCVideoView.h/m was removed incorrectly. 
BUG=3121
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10739004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5793 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-26 16:42:14 +00:00
henrike@webrtc.org
dce3feb0b0 (Auto)update libjingle 63738002-> 63773382 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5787 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-26 01:17:30 +00:00
henrike@webrtc.org
ae3347a546 Fix after auto update: removed files were brought back. 
BUG=N/A
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5782 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-25 18:17:02 +00:00
fischman@webrtc.org
76d4f389bb AppRTCDemo(iOS): allow rooms with no incoming audio. 
Also fix a compile-time warning for a leftover unimplemented method
(RTCVideoRenderer:setTransform).

R=noahric@google.com

Review URL: https://webrtc-codereview.appspot.com/10629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5780 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-25 17:40:38 +00:00
henrike@webrtc.org
6e3dbc2a77 (Auto)update libjingle 63648983-> 63738002 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5779 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-25 17:09:47 +00:00
fischman@webrtc.org
385a722646 PeerConnection(iOS): make ARC-clean talk/.../objc* and talk/examples/ios 
- Removes a strong-reference cycle between RTCPeerConnection and
  RTCPeerConnectionObserver
- Gives RTCPeerConnectionObserver a virtual dtor
- Ensures RTCPeerConnectionTest tears down correctly
- Ensures AppRTCDemo tears down correctly

This is the talk/ half; the webrtc/ half is in https://webrtc-codereview.appspot.com/10539005

BUG=3054,3055,3100
R=noahric@google.com

Review URL: https://webrtc-codereview.appspot.com/10499005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5771 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-25 05:16:29 +00:00
fischman@webrtc.org
7fa1fcb72c AppRTCDemo(ios): style/cleanup fixes following cr/62871616-p10 
BUG=2168
R=noahric@google.com

Review URL: https://webrtc-codereview.appspot.com/9709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5768 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-25 00:11:56 +00:00
henrike@webrtc.org
d3d6bce9ed (Auto)update libjingle 62865357-> 62871616 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5674 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-10 20:41:22 +00:00
pbos@webrtc.org
371243dfa3 Remove std:: prefixes from C functions in talk/. 
std::memcpy -> memcpy for instance. This change was motivated by a
compile report complaining that std::rand() was used instead of rand(),
probably with a stdlib.h include instead of cstdlib. Use of C functions
without the std:: prefix is a lot more common, so removing std:: to
address this.

BUG=
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5657 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-07 15:22:04 +00:00
henrike@webrtc.org
40b3b68cdf Update libjingle 62364298->62472237 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5632 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-03 18:30:11 +00:00
henrike@webrtc.org
1bbfb57d71 Rollback of r5629 "(Auto)update libjingle 62364298-> 62368661". 
BUG=N/A
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9329004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5631 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-03 17:37:52 +00:00
henrike@webrtc.org
31413dc635 (Auto)update libjingle 62364298-> 62368661 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5629 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-03 16:47:01 +00:00
fischman@webrtc.org
bcfc1670d6 AppRTCDemo(android): don't send local SDP until it's set. 
This fixes a race condition where the remote participant could receive the
offer, create & set its answer locally, send it back, and then try to set the
answer before the local set completed.  Observed intermittently in loopback
calls when setLocalDescription is intentionally delayed (debugging something
else).

R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5625 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-01 00:02:27 +00:00
henrike@webrtc.org
b8395ebe14 (Auto)update libjingle 62293974-> 62364298 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5623 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-28 21:57:22 +00:00
braveyao@webrtc.org
eaadecaf98 iOS, AppRTCDemo: Fixes exception due to JSON for ice using "urls" instead of "url", which is introduced by r5599. 
BUG=2962
TEST=
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9109005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5610 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-26 04:16:02 +00:00
henrike@webrtc.org
79a1cff65a Android, AppRTCDemo: Fixes java exception due to JSON for ice using "urls" instead of "url". 
BUG=2952
TEST=Manual
TBR=braveyao

Review URL: https://webrtc-codereview.appspot.com/9099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5605 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-24 23:22:18 +00:00
fischman@webrtc.org
c5d506a106 AppRTCDemo(android): clarified README on how to launch app using adb. 
TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8689005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5553 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-14 17:55:13 +00:00
fischman@webrtc.org
3eda643a91 PeerConnection(java): added MediaConstraints support to AudioSource, now fed to AudioTrack. 
BUG=2912
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5540 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-13 04:01:04 +00:00
fischman@webrtc.org
540acde5b3 PeerConnection(java): use MediaCodec for HW-accelerated video encode where available. 
Still disabled by default until https://code.google.com/p/webrtc/issues/detail?id=2899 is resolved.

Also (because I needed them during development):
- make AppRTCDemo "debuggable" for extra JNI checks
- honor audio constraints served by apprtc.appspot.com
- don't "restart" video when it hasn't been stopped (affects running with the
  screen off)

BUG=2575
R=noahric@google.com

Review URL: https://webrtc-codereview.appspot.com/8269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5539 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-13 03:56:14 +00:00
jiayl@webrtc.org
14d80793a8 PeerConnectionClient needs to initialize SSL. 
BUG=2911
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5531 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-12 00:41:59 +00:00
fischman@webrtc.org
82387e4608 Add ability to receive calls for iOS 
BUG=2701
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7989005

Patch from Sajid Hussain <shussain@temasys.com.sg>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5518 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-10 18:47:11 +00:00
henrike@webrtc.org
2ce9a64b75 Talk: Removes deprecated example apps and moves the server apps to trunk/talk/examples. 
BUG=12545067
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7159004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5397 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-16 16:49:53 +00:00
sergeyu@chromium.org
4b26e2eee3 Update libjingle to 59676287 
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5390 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-15 23:15:54 +00:00
fischman@webrtc.org
d7568a08c3 PeerConnection(java): Add OnRenegotiationNeeded support 
Also:
- Make PeerConnectionObserver::OnRenegotiationNeeded() pure virtual to avoid
  this sort of mistake in the future.
- Sprinkle @Override annotations on some callback definitions that were missing
  them.
- Fix a JNI method-signature-lookup typo (s/(V)V/()V/) in PCOJava::OnError()
- Add an explicit ScopedLocalFrameRef to PCOJava::OnError() to match all other
  C++-fired callbacks, for consistency.

BUG=2771
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5376 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-13 22:04:12 +00:00
fischman@webrtc.org
1794693ec8 AppRTCDemo(android): close() the throw-away DataChannel. 
Otherwise, the PeerConnection remembers the channel enough to include an
m=application line in its offer SDP, causing connection to chrome to fail, since
apprtc.appspot.com doesn't include an RtpDataChannels:true constraint in its
RTCPeerConnection constructor call.

R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6729005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5354 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-08 18:29:34 +00:00
wu@webrtc.org
2018269dc3 Revert 5274 "Update talk to 58113193 together with https://webrt..." 
> Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/.
> 
> R=mallinath@webrtc.org, niklas.enbom@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/5719004

TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5275 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-12 22:54:25 +00:00
wu@webrtc.org
a129b6cd13 Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/. 
R=mallinath@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5274 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-12 22:40:39 +00:00
fischman@webrtc.org
f41f06b916 PeerConnection(java): rationalize pointer-to-jlong conversion. 
In r4665 I went a bit crazy with the manual reinterpretation of a pointer to a
jlong (to avoid undefined behavior) but that's what reinterpret_cast<> is for.
So use it directly now.
Added a do-nothing DataChannel to AppRTCDemo to regression test this, since the
only repro I've found of the original bug requires ARM ABI (PeerConnectionTest
on ia32 fails to repro).

BUG=2302
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5269 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-11 21:07:18 +00:00
kjellander@webrtc.org
f9bdbe3619 Roll chromium_revision 232627:238260 
This brings us the updated swarming_client
that has moved out from Chromium into a standalone
project.
Because of this, all .isolate files needed to be
updated as well, similar to the changes in
https://codereview.chromium.org/29993003

TEST=trybots passing
BUG=none
R=andrew@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5260 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-11 13:37:12 +00:00
sergeyu@chromium.org
5bc25c41fc Update libjingle to 57692857 
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5217 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-05 00:24:06 +00:00
sergeyu@chromium.org
a23f0ca4ba Update talk to 56619788 
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3839005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5120 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-13 22:48:52 +00:00
fischman@webrtc.org
9ca93a8b8e Explicitly @synthesize ObjC @properties 
This is required after https://code.google.com/p/gyp/source/detail?r=1768
turned on -Wobjc-missing-property-synthesis for ninja builds (until then it
was only enabled for xcode builds) to allow chromium_deps to roll in
webrtc/DEPS.

BUG=2560
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5047 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-29 00:14:15 +00:00
wu@webrtc.org
97077a3ab2 Update libjingle to 55618622. 
Update libyuv to r826.

TEST=try bots
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5038 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-25 21:18:33 +00:00
fischman@webrtc.org
1c82037494 AppRTCDemo(android): remove vestigial mentions of PowerManager 
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2402004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4995 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-17 20:53:12 +00:00
fischman@webrtc.org
4446134757 AppRTCDemo(android): support boolean value for MediaStreamConstraints.{audio,video}. 
Previously it was assumed that these values were always MediaTrackConstraints but
http://dev.w3.org/2011/webrtc/editor/getusermedia.html#idl-def-MediaStreamConstraints
allows them to be boolean, too.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2352004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4918 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-03 22:34:10 +00:00
fischman@webrtc.org
6c82e04cee Android standalone: remove some usages of deprecated APIs and prevent further regressions. 
Also:
- Fixed WebRTCDemo UI to say "SwitchToBack" at startup since default camera is front
- Rebuild WebRTCDemo APK when resources/layout/strings change

R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2337004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4916 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-03 18:57:48 +00:00