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2714 Commits

Author SHA1 Message Date
henrik.lundin@webrtc.org
a366e810a9 Adding NetEq performance test to webrtc_perf_tests 
The performance test is based on the neteq4_speed_test application. The
bulk of the test code is extracted into a test class, and included into
the neteq_unittest_tools target. The actual gtest that runs the
performance test is implemented in neteq_performance_unittest.cc, and
built as a part of webrtc_perf_tests.

The old stand-alone test application is now made dependent on the new
test class, to avoid code duplication.

BUG=2397
R=andrew@webrtc.org, kjellander@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5362 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-10 08:24:04 +00:00
bjornv@webrtc.org
fa8d534e09 Delay Estimator: Adds unittests for robust validation. 
In addition to unittests a cast losing constness was corrected.
The tests added are:
1. Adjusting allowed_offset when robust validation is disabled should have no impact.
2. For noise free signals there should be no difference between robust validation or not.
3. Robust validation acts faster during startup.

BUG=None
TESTED=modules_unittests, trybots
R=aluebs@webrtc.org, andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5361 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-10 07:42:07 +00:00
henrik.lundin@webrtc.org
e7ce437333 Fixing lint errors in NetEq4 
Just taking care of a few old lint errors.

R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5359 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-09 14:01:55 +00:00
andresp@webrtc.org
c5aeb2aa15 Make code simpler on VCMEncodedCallback. 
R=marpan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5358 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-09 08:04:32 +00:00
andresp@webrtc.org
1df9dc3957 Isolate register post encode callback in video coding module to simplify code and critical sections. 
R=marpan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5357 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-09 08:01:57 +00:00
andresp@webrtc.org
b08a12d6e8 Isolate debug recording from video sender into a thread safe small class. 
R=marpan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5353 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-08 12:38:22 +00:00
solenberg@webrtc.org
ab2405164a Add another test case for AST/TOF switching. 
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5899005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5352 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-08 08:59:44 +00:00
bjornv@webrtc.org
bccd53de57 Delay Estimator: Converts a constant into a configurable parameter. 
The parameter is used in the robust validation scheme, which will be turned on in a separate CL.

* Setter and getter for allowed delay offset.
* Updated unittests.

BUG=None
TESTED=modules_unittests, trybots
R=aluebs@webrtc.org, andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5351 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-08 08:18:15 +00:00
andrew@webrtc.org
d335094852 Init to 16 kHz in the fixed-point profile. 
Fixes modules_unittests for fixed-point builds (Android).

TBR=bjornv

Review URL: https://webrtc-codereview.appspot.com/6709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5349 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-07 18:57:10 +00:00
andrew@webrtc.org
b6541ca3a1 Ensure capture_levels_ is sized correctly at init time. 
Fixes failing voe_auto_test and audioproc_perf.

TBR=bjornv

Review URL: https://webrtc-codereview.appspot.com/6699004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5348 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-07 18:36:10 +00:00
phoglund@webrtc.org
cf9d364063 Now printing less output from compare_videos.py. 
Alternative solution to the one in
https://codereview.chromium.org/114003006/.

I considered adding a verbose flag, but it needs to be passed through
like 5 functions, so I didn't think it was worth it for a function of
such speculative use.

BUG=chromium:327990
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5347 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-07 17:59:30 +00:00
andrew@webrtc.org
60730cfe3c Remove the requirement to call set_sample_rate_hz and friends. 
Instead have ProcessStream transparently handle changes to the stream
audio parameters (sample rate and channels). This removes two locks
per 10 ms ProcessStream call taken by VoiceEngine (four total with the
audio level indicator.)

Also, prepare future improvements by having the splitting filter take
a length parameter. This will allow it to work at different sample
rates. Remove the useless splitting_filter wrapper.

TESTED=voe_cmd_test with audio processing enabled and switching between
codecs; unit tests.

R=aluebs@webrtc.org, bjornv@webrtc.org, turaj@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5346 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-07 17:45:09 +00:00
pbos@webrtc.org
39669c5c8f Remove outdated DestroyVideoSendStream comment. 
BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5345 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-07 12:27:22 +00:00
sprang@webrtc.org
ccd42840bc Wire up statistics in video send stream of new video engine api 
Note, this CL does not contain any tests. Those are implemeted as call
tests and will be submitted when the receive stream is wired up as well.

BUG=2235
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5559006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5344 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-07 09:54:34 +00:00
bjornv@webrtc.org
a89d17d5b7 Delay Estimator: robust_validation should be stored over a reset 
BUG=None
TESTED=modules_unittests, trybots
R=aluebs@webrtc.org, andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5337 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-02 07:07:04 +00:00
braveyao@webrtc.org
2fb72cfeec Add include guards to forward_error_correction_internal.h 
R=henrika@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5789005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5335 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-24 05:06:12 +00:00
braveyao@webrtc.org
0062a6d099 Fix the include guard in transmit_mixer.h 
R=henrika@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5334 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-24 03:58:51 +00:00
braveyao@webrtc.org
a7cfa6704a Fix the include guard in transmit_mixer.h 
R=henrika@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5333 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-24 03:39:10 +00:00
fischman@webrtc.org
000dde99c8 Android build: make it quiet on success and not overly noisy on failure. 
- OpenSLDemo and WebRTCDemo get the sauce that AppRTCDemo got in r5271
- libjingle_peerconnection_jar is now silent on success
- Fix a bug introduced by r5271 which caused ant logs to be emitted to a subdir of talk/examples instead of in the gyp output directory.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6199005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5332 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-20 22:49:35 +00:00
andresp@webrtc.org
f6acf98a46 Fix the android clang bot for compiling with thread annotations. 
TBR=niklas.enbom@webrtc.org
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6279005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5330 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-20 21:54:26 +00:00
andresp@webrtc.org
7fb75ecbd4 Add thread_annotations for clang targets. 
TESTED: As expected clang bots catched a few issues which are fixed with this CL, other bots ignore the annotations and compile fine.

R=niklas.enbom@webrtc.org, phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6209004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5328 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-20 20:20:50 +00:00
mflodman@webrtc.org
6031001565 If the configured start bitrate is higher than the configures max 
bitrate, cap the star rate accordingly.

BUG=2720
R=asapersson@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6239004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5327 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-20 15:07:12 +00:00
sprang@webrtc.org
8dbca8d665 Race condition in ViECapturer::RegisterObserver 
Critical section ViECapturer.observer_cs_ should be taken when
registering an observer.

BUG=2734
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5326 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-20 11:36:03 +00:00
tnakamura@webrtc.org
a463d73b99 Update WebRTC to version 3.48 
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6189004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5324 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-19 22:38:38 +00:00
sprang@webrtc.org
54ae4ffb9e Add callbacks for receive channel RTCP statistics. 
This allows a listener to receive new statistics as it is generated - avoiding the need to poll. This also makes handling stats from multiple RTP streams more tractable.
The change is primarily targeted at the new video engine API.

TEST=Unit test in ReceiveStatisticsTest. Integration tests to follow as call tests when fully wired up.

BUG=2235
R=henrika@webrtc.org, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5323 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-19 13:26:02 +00:00
andresp@webrtc.org
e682aa5077 Refactoring MediaOptimization so it can easily be turned into a thread-safe class. 
BUG=2732
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5322 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-19 10:59:48 +00:00
stefan@webrtc.org
faada6e604 Integrate fake_network_pipe into direct_transport. 
TEST=trybots
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5321 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-18 20:28:25 +00:00
kjellander@webrtc.org
5fe2d65c43 Remove metrics_unittests 
This target has been merged into video_engine_tests in r5284.

BUG=webrtc:1843
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5319 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-18 13:27:37 +00:00
pbos@webrtc.org
8a54417968 Remove media_file from VideoEngine dependencies. 
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6139004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5318 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-18 10:00:29 +00:00
mflodman@webrtc.org
b429e516a9 cpplint cleaning new API and its implementation files. 
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6089005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5317 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-18 09:46:22 +00:00
mflodman@webrtc.org
bcd124cdba Reduced execution time for CallTest::ReceivesPliAndRecovers, by dropping only one packet and made it predictable by removing rand(). 
Follow up steps is to support NackConfig.rtp_hostory_ms and/or increase fake encoder bitrate.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6109005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5316 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-18 09:45:45 +00:00
mflodman@webrtc.org
1fa41be66a Speeding up CallTest.ReceivesAndRetransmitsNack and removed the random packet loss. 
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5315 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-18 09:44:53 +00:00
sergeyu@chromium.org
8ae72560dd Make MouseCursor mutable 
MouseCursor objects were previous immutable which makes it harder to
implement deserializers when MouseCursor is sent over IPC in Chromium.

R=dcaiafa@chromium.org

Committed: https://code.google.com/p/webrtc/source/detail?r=5310

Review URL: https://webrtc-codereview.appspot.com/6059005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5314 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-18 02:18:01 +00:00
fischman@webrtc.org
f8be8df33a audio_processing_unittest: unbreak clang compilation. 
BUG=2735
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5313 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-17 23:46:39 +00:00
fischman@webrtc.org
179908c81c JNI Audio: remove dead members. 
BUG=2735
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6049004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5312 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-17 23:46:14 +00:00
sergeyu@chromium.org
e4c927208b Revert "Make MouseCursor mutable" 
This reverts commit a6db8ab8bc4b569a26633b0ca3665297f1a5349b.

TBR=dcaiafa@chromium.org

Review URL: https://webrtc-codereview.appspot.com/6079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5311 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-17 22:48:50 +00:00
sergeyu@chromium.org
8fd1d26536 Make MouseCursor mutable 
MouseCursor objects were previous immutable which makes it harder to
implement deserializers when MouseCursor is sent over IPC in Chromium.

R=dcaiafa@chromium.org

Review URL: https://webrtc-codereview.appspot.com/6059005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5310 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-17 22:19:12 +00:00
pbos@webrtc.org
052fa6243a Stop transport in test SuspendBelowMinBitrate. 
Avoids race when packets are still left in the network while the Call is
being destroyed.

R=mflodman@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/6009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5307 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-17 11:19:58 +00:00
mflodman@webrtc.org
e6b871bb29 Added method for getting default module state and protect agains a 
read/write race for child_modules_.

BUG=2731
TEST=tsan
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5919005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5306 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-17 08:30:40 +00:00
pbos@webrtc.org
eb7b7bce3d Modify video_render/ to allow a single old frame. 
This stabilizes tests as a single frame reaches end-to-end, as well as
allowing slow or heavily-loaded systems to see any video updates even if
the frame takes more than 500ms in the pipeline.

R=mflodman@webrtc.org, stefan@webrtc.org
BUG=2724

Review URL: https://webrtc-codereview.appspot.com/5949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5303 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-16 18:24:37 +00:00
pbos@webrtc.org
919f87fb36 Delete capturers after destroying streams in test. 
Since the renderers in CallTest.SendsAndReceiveStreams also stopped the
capturers they must be deleted after the VideoReceiveStream is stopped
or an use-after-free may occur.

R=mflodman@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/5969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5300 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-16 14:55:54 +00:00
asapersson@webrtc.org
e7b1e11283 Revert 5285 "Revert 5228 "Use the RTT from RtcpRttStats class if..." 
> Revert 5228 "Use the RTT from RtcpRttStats class if provided whe..."
> 
> > Use the RTT from RtcpRttStats class if provided when sending/receiving NACK.
> > 
> > R=holmer@google.com
> > 
> > Review URL: https://webrtc-codereview.appspot.com/5049004
> 
> TBR=asapersson@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/5799004

TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5299 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-16 14:40:36 +00:00
bjornv@webrtc.org
1e7d61270c Simplification of histogram normalization in delay estimator. 
- Replaces a for loop with a single element update to save complexity. No regression in performance seen on set of recordings.
- Removes UpdatesMadeUponChange() and put code straight into ProcessBinarySpectrum().

BUG=None
TESTED=module_unittest, trybots, verified manually on set of recordings.
R=aluebs@webrtc.org, andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5298 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-16 13:37:28 +00:00
pbos@webrtc.org
5ab756703e Revert r5294 to re-roll r5293. 
To fix races in test each stream now owns its own encoder/decoder.

R=mflodman@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/5919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5297 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-16 12:24:44 +00:00
bjornv@webrtc.org
5c64508b03 Adds robust validation functionality to the delay estimator 
Evaluated over a 51 recordings:
False positives went from 4.4% to 0.7%
Missed detections unchanged at 0.8%
No increase in complexity, but need to re-evaluate that.

TESTED=trybots, unittests, verified against Matlab implementation
BUG=None
R=aluebs@webrtc.org, andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5296 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-16 10:57:53 +00:00
sprang@webrtc.org
87ad57bc75 Incorrect iterator++ in ModuleRtpRtcpImpl::RegisterVideoBitrateObserver 
The iterator is incremented both in loop header and loop body. Should
only be incremented in header.

BUG=2727
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5295 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-16 07:43:51 +00:00
turaj@webrtc.org
41e2615e02 Revert 5293 "Auto instantiate RBE depending on whether AST or TO..." 
> Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream.
> 
> BUG=
> R=mflodman@webrtc.org, stefan@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/5409004

TBR=solenberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5294 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-15 18:42:32 +00:00
solenberg@webrtc.org
341e91441a Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream. 
BUG=
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5293 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-13 23:57:54 +00:00
turaj@webrtc.org
e1bc6c8d8b Disabled tests on Android. The issue 2723 is filed to investigate the reason for tests failing. 
BUG=
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5809005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5292 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-13 22:04:18 +00:00
stefan@webrtc.org
dd393e7b9d Measure pacer queue size based on when packets are inserted rather than captured. 
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5291 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-13 22:03:27 +00:00