122caa51b1
After an audio interruption the audio unit no longer invokes its render callback, which results in a loss of audio. Restarting the audio unit post interruption fixes the issue.
...
CL also replaces deprecated AudioSession calls with equivalent AVAudioSession ones.
BUG=3487
R=glaznev@webrtc.org , noahric@chromium.org
Review URL: https://webrtc-codereview.appspot.com/21769004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6697 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-15 20:20:47 +00:00
42fe4350fe
Remove Thread::RunningForChannelManager().
...
I haven't heard of this failing, so it should be safe to remove. Let me know if this isn't the case.
BUG=3388
R=andrew@webrtc.org , wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18659004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6695 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-15 17:52:43 +00:00
89fd1e8e99
Improvements to the pacer where it lost some budget due to truncation errors.
...
With this CL the resolution is increased to microseconds and proper rounding
is done in the Process() function. This means that we will be allowed to send
more than prior to r6664 as we previously truncated away parts of our budget.
We will also not lose budget due to inaccurate calculations in
TimeUntilNextProcess(), which was a regression in r6664.
BUG=cr/393950
TEST=out/Debug/webrtc_perf_tests --gtest_filter=RampUpTest.Simulcast
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6694 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-15 16:40:38 +00:00
376b4ea93f
Fix breakage introduced by r6691.
...
ModuleRtpRtcpImpl returned incorrectly on RemoteNTP as the
RTCPReceiver::NTP changed return type.
BUG=
TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18799004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6693 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-15 15:51:33 +00:00
2f4b14e3f3
Make RTCP sender report send media bytes.
...
r6654 changed RtpSender::Bytes() to return the number of bytes sent
instead of number of media bytes. This is used by VideoEngine for stats.
This change broke RTCP which sends this same count as the number of
payload bytes sent (excluding headers and padding).
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14959004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6691 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-15 15:25:39 +00:00
ffa8dcab1e
Eliminate unnecessary #include
...
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14979004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6690 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-15 12:50:13 +00:00
324f63ca38
rtc::Fatal output: Print space between # and message
...
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6689 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-15 11:41:05 +00:00
bc73871251
Remove the VPM denoiser.
...
The VPM denoiser give bad results, is slow and has not been used in
practice. Instead we use the VP8 denoiser. Testing this denoiser takes
up a lot of runtime on linux_memcheck (about 4 minutes) which we can do
without.
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16069004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6688 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-15 09:50:40 +00:00
92a9bacf9a
Rebase webrtc/base with r6682 version of talk/base:
...
cls ported: r6671, r6672, r6679 (reverts and unreverts in r6680, r6682).
svn diff -r 6656:6682 http://webrtc.googlecode.com/svn/trunk/talk/base >
6682.diff
sed -i.bak "s/talk_base/rtc/g" 6682.diff
sed -i.bak "s/#ifdef WIN32/#if defined(WEBRTC_WIN)/g" 6682.diff
sed -i.bak "s/#if defined(WIN32)/#if defined(WEBRTC_WIN)/g" 6682.diff
patch -p0 -i 6682.diff
BUG=3379
TBR=tommi@webrtc.org ,jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6683 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-14 22:03:57 +00:00
a4da771914
Fix deadlock in Android stopCapture() call.
...
BUG=3467
R=braveyao@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18789004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6673 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-14 17:01:53 +00:00
9bef551ba1
GN: Fix include paths for WebRTC in Chromium build.
...
Most WebRTC source files are using full paths for includes which
requires the root to be in the include path.
This is currently handled in the common_inherited_config config in
webrtc/BUILD.gn: the .. include_dir.
However, when built from Chromium, the include
paths are not inherited in the same way when building the all target.
Building the 'webrtc' target of Chrome works without the changes
in this CL, but the default target fails.
BUG=3441
TEST=Built the default target from a Chromium checkout with
https://codereview.chromium.org/321313006/ applied and
src/third_party/webrtc linked to the webrtc folder of the WebRTC
workspace.
R=brettw@chromium.org
Review URL: https://webrtc-codereview.appspot.com/15989004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6670 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-13 09:02:54 +00:00
9e1acc8728
Fix bugs introduced by https://code.google.com/p/webrtc/source/detail?r=6667 .
...
A few places were relying on temporalIdx being signed. Fix to explicitly check
for kNoTemporalIdx.
TBR=pbos,stefan
Review URL: https://webrtc-codereview.appspot.com/13939005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6669 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-11 20:33:39 +00:00
dd6780d85d
Remove always-true expression.
...
TBR=pbos
Review URL: https://webrtc-codereview.appspot.com/16059004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6668 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-11 19:34:54 +00:00
eec6ecdb1e
Landing pkasting's webrtc fixes for MSVC level 4 warnings in WebRTC.
...
---
Fixes for re-enabling more MSVC level 4 warnings: webrtc/ edition
This contains fixes for the following sorts of issues:
* Possibly-uninitialized local variable
* Signedness mismatch
* Assignment inside conditional
This also contains a small number of other cleanups to nearby code. In
particular several warning-disables for MSVC are removed because they don't seem
to be necessary (either that warning is not enabled or the code does not trigger
it).
BUG=crbug.com/81439
TEST=none
R=henrika@webrtc.org , pkasting@chromium.org
Review URL: https://webrtc-codereview.appspot.com/18769004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6667 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-11 19:09:59 +00:00
180e516bef
Thread annotate RTCPSender.
...
Also fixes data races in RTCPSender::SetCSRCStatus() and
RTCPSender::SetStartTimestamp().
BUG=
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6666 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-11 15:36:26 +00:00
168f23faa5
Move pacer to fully use webrtc::Clock instead of webrtc::TickTime.
...
This required rewriting the send-side delay stats api to be callback based, as otherwise the SuspendBelowMinBitrate test started flaking much more frequently since it had lock order inversion problems.
R=pbos@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21869005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6664 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-11 13:44:02 +00:00
a1bfcad3a3
Cast payload types to int for logging.
...
uint8_t gets interpreted as char and printed as such, instead of being
printed in decimal, casting them to int allows us to read what payload
types are actually used without converting them from ASCII first.
BUG=chromium:390874
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6662 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-11 12:33:45 +00:00
fb2e7c22a0
Document that channels are stored contiguously in AudioBuffer
...
R=andrew@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20909004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6661 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-11 11:40:48 +00:00
d212ffcfc6
Remove unnecessary build message.
...
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13909004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6660 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-11 11:15:35 +00:00
4ef438e2de
Remove the send-side cname getter APIs from voice and video engine.
...
These APIs aren't being used, and introduces deadlocks when using GetStats() in the new Call api. Having getters for cname at the send-side is pointless, as it's always the user who sets the cname.
R=henrika@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6659 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-11 09:55:30 +00:00
b614d0626f
Rebase webrtc/base with r6655 version of talk/base:
...
cls to port: r6633,r6639 (there is no cl in between that affects base and all other talk/base cls took care of webrtc/base as well (see r6569, r6624)):
svn diff -r 6632:6639 http://webrtc.googlecode.com/svn/trunk/talk/base > 6655.diff
sed -i.bak "s/talk_base/rtc/g" 6655.diff
patch -p0 -i 6555.diff
BUG=3379
TBR=tommi@webrtc.org ,jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21879004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6656 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-10 22:47:02 +00:00
72491b9a90
Count total bytes sent in RTPSender::Bytes().
...
Previously only media bytes were included, this adds header bytes and
padding bytes to the calculation.
BUG=
R=stefan@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19939004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6654 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-10 16:24:54 +00:00
0422100818
Fix data race in VCMTiming::ResetDecodeTime.
...
Also thread annotating class.
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17939004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6653 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-10 15:25:37 +00:00
bd9c0920ec
Skip encoding in fake VP8 encoder.
...
Broke memcheck, FakeEncoder::Encode doesn't produce valid VP8 frames.
BUG=3424
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6652 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-10 13:21:40 +00:00
7ae9108b60
Remove more unused tsan suppressions and fix call test passing the same decoder to multiple received streams.
...
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6651 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-10 10:35:12 +00:00
91f1752f2d
Support VP8 encoder settings in VideoSendStream.
...
Stop-gap solution to support VP8 codec settings in the new API until
encoder settings can be passed on to the VideoEncoder without requiring
explicit support for the codec.
BUG=3424
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17929004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6650 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-10 10:13:37 +00:00
8f1512140e
Refactor registerable callbacks for FrameCountObserver from rtp_rtcp module into vie_channel.
...
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16839004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6649 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-10 09:39:23 +00:00
5bde66e913
audio_processing: Updates aec_core_sse2.c with changes made to aec_common.h
...
The change of definitions moved to aec_common.h was done in CL17839005.
BUG=3131
TBR=kwiberg@webrtc.org
TESTED=builds locally
Review URL: https://webrtc-codereview.appspot.com/16859004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6648 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-10 08:09:50 +00:00
555fc78f27
Neon version of SubbandCoherence()
...
The performance gain on a Nexus 7 reported by audioproc is ~1.4%
The output is NOT bit exact. Any difference seen is +-1.
BUG=3131
R=bjornv@webrtc.org , cd@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17839005
Patch from Scott LaVarnway <slavarnw@gmail.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6647 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-10 08:03:11 +00:00
ac800c8004
Neon version of rftbsub_128()
...
The performance gain on a Nexus 7 reported by audioproc is ~4.5%
The output is bit exact.
BUG=3131
TESTED=trybots and manually
R=bjornv@webrtc.org , cd@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19919005
Patch from Scott LaVarnway <slavarnw@gmail.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6646 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-10 07:53:13 +00:00
5ac876bae0
Revert "Remove remains of WEBRTC_NO_STL." (rev 6641).
...
Reason breaks linux_memcheck.
TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16869004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6645 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-10 07:41:59 +00:00
47d1c98a4e
Remove remains of WEBRTC_NO_STL.
...
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6641 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-09 20:18:28 +00:00
10ef8fe611
Create FullScreenChromeWindowDetector in DesktopConfigurationOptions::CreateDefault.
...
BUG=crbug/385294
R=sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/18759004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6640 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-09 19:41:32 +00:00
7af12be781
Thread annotations for vie_encoder.cc/.h
...
Review URL: https://webrtc-codereview.appspot.com/8739005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6638 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-09 14:46:31 +00:00
045a9b17da
Remove unnecessary race suppressions copied from chromium.
...
And added suppressions to allow to run tests with gtest_parallel in which case some new races were showing up.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19909004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6635 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-09 11:44:34 +00:00
b8e9e44eac
Add full stack test cases with a fake network pipe.
...
R=pbos@webrtc.org
BUG=1872
Review URL: https://webrtc-codereview.appspot.com/20889004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6634 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-09 11:29:06 +00:00
b753762ce6
delay_estimator: Increases test coverage and makes input spectrum const
...
Noticed lack in tests verifying initial state is not left if we have zero input spectra. This CL adds such a test and change input spectra to const at affected places.
BUG=N/A
TESTED=trybots and manually
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6631 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-09 06:40:09 +00:00
12b4efefdd
Implement a work around for Chrome full-screen tab switch on Mac.
...
Chrome creates a new window in full-screen and minimizes the old window when a tab is switched to full-screen.
We try to find the new window to continue capturing for window sharing.
BUG=crbug/385294
R=sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/19839004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6629 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-08 22:05:24 +00:00
e55641d4f7
Neon version of rftfsub_128()
...
The performance gain on a Nexus 7 reported by audioproc is ~3.3%
The output is bit exact.
BUG=3131
TESTED=trybots and manually on N7
R=bjornv@webrtc.org , cd@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14819004
Patch from Scott LaVarnway <slavarnw@gmail.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6628 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-08 21:12:23 +00:00
d11bec40b2
Refactor registerable callbacks for VideoBitrateObserver from rtp_rtcp module into vie_channel.
...
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20879004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6626 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-08 14:32:58 +00:00
3d7da88e06
Refactor ramp-up tests to have separate help files for the test classes, to make things more reusable.
...
R=pbos@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18739004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6625 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-08 13:59:46 +00:00
ecb8723402
Change Timing::WallTimeNow to be static.
...
There's no need to construct a Timing object to call this method.
On Windows we were unnecessarily calling CreateWaitableTimer + CloseHandle but never actually using that waitable timer.
There's otherwise no change in functionality.
R=xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20869004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6624 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-08 12:48:29 +00:00
62bafae661
Some refactoring inside rtp_rtcp/.
...
Renaming ModuleRTPUtility -> RtpUtility.
Renaming RTPHeaderParser -> RtpHeaderParser.
Making RtpHeaderParser accept size_t instead of int for packet length.
Making RtpUtility::RtpHeaderParser accept size_t for packet length.
BUG=
R=stefan@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6623 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-08 12:10:51 +00:00
241a9b0b65
Fixing compile error.
...
Made a mistake in https://webrtc-codereview.appspot.com/13849004/ ,
fixing that here.
TBR=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13859004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6622 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-08 11:48:37 +00:00
22292df53b
Adding explicit check for using dummy file devices.
...
Calling into the file device factory without being compiled with file
devices makes no sense and would cause hard-to-debug errors. Therefore
I'm adding an explicit check so this isn't allowed.
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13849004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6621 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-08 11:39:19 +00:00
33d110d8ea
Tight data race suppressions around thread_posix.
...
BUG=3372,3549
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19879004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6620 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-08 10:36:39 +00:00
af38f4e511
Extract RTP-header SSRC inline in Call.
...
Prevents unknown-RTP-header-extension warnings to be flooding from the
RTP-header parsing as there's no way to register RTP extensions for the
parser in Call as they're allowed to differ between RTP streams.
RTP-header parsing should instead be done separately in every
VideoReceiveStream.
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6619 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-08 07:38:12 +00:00
3c637cdaa5
Clean data races from system_wrappers_unittests.
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- Remove unittest_utilities that are not used.
- Remove SetLevelFilter that does not seems necessary and anyhow was racy.
BUG=3549
R=henrike@webrtc.org , henrike
Review URL: https://webrtc-codereview.appspot.com/16819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6617 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-07 20:37:39 +00:00
285e9bc84d
Fix potential deadlock in webrtc/system_wrappers/source/logging_unittest.cc.
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crit_ should not be held while calling Trace.
BUG=3003
R=henrike@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17859005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6616 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-07 20:27:33 +00:00
5f2c81c17f
webrtc/base: Fixes miss in base.gyp for windows. See https://code.google.com/p/webrtc/source/browse/trunk/talk/libjingle.gyp?r=6503#764 for the corresponding condition.
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BUG=3379
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19849004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6615 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-07 17:42:45 +00:00
