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Médias (3)
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MediaSPIP Simple : futur thème graphique par défaut ?
26 septembre 2013, par
Mis à jour : Octobre 2013
Langue : français
Type : Video
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GetID3 - Bloc informations de fichiers
9 avril 2013, par
Mis à jour : Mai 2013
Langue : français
Type : Image
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GetID3 - Boutons supplémentaires
9 avril 2013, par
Mis à jour : Avril 2013
Langue : français
Type : Image
Autres articles (48)
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MediaSPIP v0.2
21 juin 2013, parMediaSPIP 0.2 est la première version de MediaSPIP stable.
Sa date de sortie officielle est le 21 juin 2013 et est annoncée ici.
Le fichier zip ici présent contient uniquement les sources de MediaSPIP en version standalone.
Comme pour la version précédente, il est nécessaire d’installer manuellement l’ensemble des dépendances logicielles sur le serveur.
Si vous souhaitez utiliser cette archive pour une installation en mode ferme, il vous faudra également procéder à d’autres modifications (...) -
Mise à disposition des fichiers
14 avril 2011, parPar défaut, lors de son initialisation, MediaSPIP ne permet pas aux visiteurs de télécharger les fichiers qu’ils soient originaux ou le résultat de leur transformation ou encodage. Il permet uniquement de les visualiser.
Cependant, il est possible et facile d’autoriser les visiteurs à avoir accès à ces documents et ce sous différentes formes.
Tout cela se passe dans la page de configuration du squelette. Il vous faut aller dans l’espace d’administration du canal, et choisir dans la navigation (...) -
Installation en mode ferme
4 février 2011, parLe mode ferme permet d’héberger plusieurs sites de type MediaSPIP en n’installant qu’une seule fois son noyau fonctionnel.
C’est la méthode que nous utilisons sur cette même plateforme.
L’utilisation en mode ferme nécessite de connaïtre un peu le mécanisme de SPIP contrairement à la version standalone qui ne nécessite pas réellement de connaissances spécifique puisque l’espace privé habituel de SPIP n’est plus utilisé.
Dans un premier temps, vous devez avoir installé les mêmes fichiers que l’installation (...)
Sur d’autres sites (8584)
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Revision 82957 : bugfix sur le scoring fulltext : - la liste des champs=>poids n’était pas ...
11 juin 2014, par cedric@… — Logbugfix sur le scoring fulltext :
la liste des champs=>poids n’était pas prise en compte par la faute d’un mauvais nommage de variable (utilisation sur l’index titre de fulltext)
Amelioration du scoring :le score calculé par champs/poids classique SPIP est plus précis car compte les occurences dans un champs, on peut donc le mixer avec le score fulltext pour améliorer les résultats (mais plus couteux en terme de calcul) : avec
define(’_FULLTEXT_FIELD_SCORE’,true) les 2 scores sont ajoutés -
Correct command to transmit audio to ip camera using ffmpeg ?
4 novembre 2016, par the_naiveSo I found some hints in this discussion on the correct command to transmit audio to Axis IP camera through using ffmpeg in windows, but still I have not managed to successfully transmit audio to the camera.
The command I’m using is the following :
ffmpeg -v debug -y -re -f dshow -i "audio=Microphone (2- High Definition Audio Device)" -c:a pcm_mulaw -ac 1 -ar 16000 -b:a 128k -f flv http://oper
ator:operator@10.10.210.2/axis-cgi/audio/transmit.cgi -multiple_requests 1 -reconnect_at_eof 1 -reconnect_streamed 1 -content_type "audio/basic" -reportThe ouput I get following this command is the following :
ffmpeg started on 2016-11-04 at 17:32:13
Report written to "ffmpeg-20161104-173213.log"
Command line:
ffmpeg -v debug -y -re -f dshow -i "audio=Microphone (2- High Definition Audio Device)" -c:a pcm_mulaw -ac 1 -ar 16000 -b:a 128k -f flv http://operator:operator@10.10.210.2/axis-cgi/audio/transmit.cgi -content_type audio/basic -multiple_requests 1 -reconnect 1 -reconnect_at_eof 1 -reconnect_streamed 1 -report
ffmpeg version N-82225-gb4e9252 Copyright (c) 2000-2016 the FFmpeg developers
built with gcc 5.4.0 (GCC)
configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-dxva2 --enable-libmfx --enable-nvenc --enable-avisynth --enable-bzlib --enable-libebur128 --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenh264 --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-decklink --enable-zlib
libavutil 55. 35.100 / 55. 35.100
libavcodec 57. 66.101 / 57. 66.101
libavformat 57. 57.100 / 57. 57.100
libavdevice 57. 2.100 / 57. 2.100
libavfilter 6. 66.100 / 6. 66.100
libswscale 4. 3.100 / 4. 3.100
libswresample 2. 4.100 / 2. 4.100
libpostproc 54. 2.100 / 54. 2.100
Splitting the commandline.
Reading option '-v' ... matched as option 'v' (set logging level) with argument 'debug'.
Reading option '-y' ... matched as option 'y' (overwrite output files) with argument '1'.
Reading option '-re' ... matched as option 're' (read input at native frame rate) with argument '1'.
Reading option '-f' ... matched as option 'f' (force format) with argument 'dshow'.
Reading option '-i' ... matched as input file with argument 'audio=Microphone (2- High Definition Audio Device)'.
Reading option '-c:a' ... matched as option 'c' (codec name) with argument 'pcm_mulaw'.
Reading option '-ac' ... matched as option 'ac' (set number of audio channels) with argument '1'.
Reading option '-ar' ... matched as option 'ar' (set audio sampling rate (in Hz)) with argument '16000'.
Reading option '-b:a' ... matched as option 'b' (video bitrate (please use -b:v)) with argument '128k'.
Reading option '-f' ... matched as option 'f' (force format) with argument 'flv'.
Reading option 'http://operator:operator@10.10.210.2/axis-cgi/audio/transmit.cgi' ... matched as output file.
Reading option '-content_type' ... matched as AVOption 'content_type' with argument 'audio/basic'.
Reading option '-multiple_requests' ... matched as AVOption 'multiple_requests' with argument '1'.
Reading option '-reconnect' ... matched as AVOption 'reconnect' with argument '1'.
Reading option '-reconnect_at_eof' ... matched as AVOption 'reconnect_at_eof' with argument '1'.
Reading option '-reconnect_streamed' ... matched as AVOption 'reconnect_streamed' with argument '1'.
Reading option '-report' ... matched as option 'report' (generate a report) with argument '1'.
Trailing options were found on the commandline.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option v (set logging level) with argument debug.
Applying option y (overwrite output files) with argument 1.
Applying option report (generate a report) with argument 1.
Successfully parsed a group of options.
Parsing a group of options: input file audio=Microphone (2- High Definition Audio Device).
Applying option re (read input at native frame rate) with argument 1.
Applying option f (force format) with argument dshow.
Successfully parsed a group of options.
Opening an input file: audio=Microphone (2- High Definition Audio Device).
[dshow @ 00000000000279e0] Selecting pin Capture on audio only
dshow passing through packet of type audio size 88200 timestamp 310221040000 orig timestamp 310221040000 graph timestamp 310226130000 diff 5090000 Microphone (2- High Definition Audio Device)
[dshow @ 00000000000279e0] All info found
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, dshow, from 'audio=Microphone (2- High Definition Audio Device)':
Duration: N/A, start: 31022.104000, bitrate: 1411 kb/s
Stream #0:0, 1, 1/10000000: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s
Successfully opened the file.
Parsing a group of options: output file http://operator:operator@10.10.210.2/axis-cgi/audio/transmit.cgi.
Applying option c:a (codec name) with argument pcm_mulaw.
Applying option ac (set number of audio channels) with argument 1.
Applying option ar (set audio sampling rate (in Hz)) with argument 16000.
Applying option b:a (video bitrate (please use -b:v)) with argument 128k.
Applying option f (force format) with argument flv.
Successfully parsed a group of options.
Opening an output file: http://operator:operator@10.10.210.2/axis-cgi/audio/transmit.cgi.
[http @ 0000000001c94040] Setting default whitelist 'http,https,tls,rtp,tcp,udp,crypto,httpproxy'
[http @ 0000000001c94040] request: POST /axis-cgi/audio/transmit.cgi HTTP/1.1
Transfer-Encoding: chunked
User-Agent: Lavf/57.57.100
Accept: */*
Expect: 100-continue
Connection: close
Host: 10.10.210.2
Icy-MetaData: 1
[http @ 0000000001c94040] request: POST /axis-cgi/audio/transmit.cgi HTTP/1.1
Transfer-Encoding: chunked
User-Agent: Lavf/57.57.100
Accept: */*
Connection: close
Host: 10.10.210.2
Icy-MetaData: 1
Authorization: Digest username="operator", realm="AXIS_ACCC8E027F47", nonce="0EcsO3xABQA=ab5efc4740a6c625ecf6a6729d0d67d2b62b615a", uri="/axis-cgi/audio/transmit.cgi", response="4bd3a627b20d6bcaba9e2f595ef6cd2a", algorithm="MD5", qop="auth", cnonce="6a579dd6664b57eb", nc=00000001
Successfully opened the file.
detected 8 logical cores
[graph 0 input from stream 0:0 @ 0000000001c9f6e0] Setting 'time_base' to value '1/44100'
[graph 0 input from stream 0:0 @ 0000000001c9f6e0] Setting 'sample_rate' to value '44100'
[graph 0 input from stream 0:0 @ 0000000001c9f6e0] Setting 'sample_fmt' to value 's16'
[graph 0 input from stream 0:0 @ 0000000001c9f6e0] Setting 'channel_layout' to value '0x3'
[graph 0 input from stream 0:0 @ 0000000001c9f6e0] tb:1/44100 samplefmt:s16 samplerate:44100 chlayout:0x3
[audio format for output stream 0:0 @ 0000000001c9fa20] Setting 'sample_fmts' to value 's16'
[audio format for output stream 0:0 @ 0000000001c9fa20] Setting 'sample_rates' to value '16000'
[audio format for output stream 0:0 @ 0000000001c9fa20] Setting 'channel_layouts' to value '0x4'
[audio format for output stream 0:0 @ 0000000001c9fa20] auto-inserting filter 'auto-inserted resampler 0' between the filter 'Parsed_anull_0' and the filter 'audio format for output stream 0:0'
[AVFilterGraph @ 000000000002ab20] query_formats: 4 queried, 6 merged, 3 already done, 0 delayed
[auto-inserted resampler 0 @ 0000000001ca4060] [SWR @ 0000000001ca4a80] Using s16p internally between filters
[auto-inserted resampler 0 @ 0000000001ca4060] [SWR @ 0000000001ca4a80] Matrix coefficients:
[auto-inserted resampler 0 @ 0000000001ca4060] [SWR @ 0000000001ca4a80] FC: FL:0.500000 FR:0.500000
[auto-inserted resampler 0 @ 0000000001ca4060] ch:2 chl:stereo fmt:s16 r:44100Hz -> ch:1 chl:mono fmt:s16 r:16000Hz
Output #0, flv, to 'http://operator:operator@10.10.210.2/axis-cgi/audio/transmit.cgi':
Metadata:
encoder : Lavf57.57.100
Stream #0:0, 0, 1/1000: Audio: pcm_mulaw ([8][0][0][0] / 0x0008), 16000 Hz, mono, s16, 128 kb/s
Metadata:
encoder : Lavc57.66.101 pcm_mulaw
Stream mapping:
Stream #0:0 -> #0:0 (pcm_s16le (native) -> pcm_mulaw (native))
Press [q] to stop, [?] for help
cur_dts is invalid (this is harmless if it occurs once at the start per stream)
av_interleaved_write_frame(): Unknown error
No more output streams to write to, finishing.
Error writing trailer of http://operator:operator@10.10.210.2/axis-cgi/audio/transmit.cgi: Error number -10053 occurredsize= 8kB time=00:00:00.49 bitrate= 131.2kbits/s speed=79.6x
video:0kB audio:8kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 2.492485%
Input file #0 (audio=Microphone (2- High Definition Audio Device)):
Input stream #0:0 (audio): 1 packets read (88200 bytes); 1 frames decoded (22050 samples);
Total: 1 packets (88200 bytes) demuxed
Output file #0 (http://operator:operator@10.10.210.2/axis-cgi/audio/transmit.cgi):
Output stream #0:0 (audio): 1 frames encoded (7984 samples); 1 packets muxed (7984 bytes);
Total: 1 packets (7984 bytes) muxed
1 frames successfully decoded, 0 decoding errors
[AVIOContext @ 0000000001c9e4c0] Statistics: 0 seeks, 2 writeouts
dshow passing through packet of type audio size 12152 timestamp 310226130000 orig timestamp 310226130000 graph timestamp 310226820000 diff 690000 Microphone (2- High Definition Audio Device)
Conversion failed!For some reason, despite setting
multiple_requests
,reconnect_eof
,reconnect_streamed
all to true, connection becomes closed.Could you please tell me what I’m doing wrong ?
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GStreamer x264enc and SIGSEGV on Android
18 juillet 2012, par CipiI am trying to play around with GStreamer-java and it's h.264 encoder (x264enc) on Android.
Now, I tried to encode raw video from test video source (set its is-live property to TRUE), and when data starts flowing into the x264enc Element, I get SIGSEGV in libx264.so and my app crashes. And it looks like code is dereferencing a NULL... Exact error location is in file "gstreamer_ndk_bundle/x264/encoder/analyse.c" line 963. And it is this line :
h->mc.memcpy_aligned( h->mb.pic.i8x8_dct_buf, h->dct.luma8x8, sizeof(h->mb.pic.i8x8_dct_buf) );
My device's CPU is Cortex 8 and it has NEON feature...
I tried to printout to a file all
h
structure fields, and no field is uninitialized.Android DEBUG info from LogCat : HERE
My questions :
- How can I know what is throwing SIGSEGV, because this line of code seems to be ok ?
- Is this a known bug or an expected behavior of x264 encoder on ARM CPU or I am missing some encoder configuration ?
- I noticed that if I set property analyse of the encoder to 0x3:0x113 (some random value I found online) I don't get SIGSEGV, encoder seems to work, but I get error in pipeline : Data flow error. Can I just set analyse to a meaningful value ?
Update
I fprintf-ed all important "h" fields just before execution runs in x264_mb_analyse_intra() function and just before the line that it breaks, guess it could help : http://pastebin.com/sAfX239j