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Rennes Emotion Map 2010-11
19 octobre 2011, par
Mis à jour : Juillet 2013
Langue : français
Type : Texte
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Submit bugs and patches
13 avril 2011Unfortunately a software is never perfect.
If you think you have found a bug, report it using our ticket system. Please to help us to fix it by providing the following information : the browser you are using, including the exact version as precise an explanation as possible of the problem if possible, the steps taken resulting in the problem a link to the site / page in question
If you think you have solved the bug, fill in a ticket and attach to it a corrective patch.
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Des sites réalisés avec MediaSPIP
2 mai 2011, parCette page présente quelques-uns des sites fonctionnant sous MediaSPIP.
Vous pouvez bien entendu ajouter le votre grâce au formulaire en bas de page. -
La sauvegarde automatique de canaux SPIP
1er avril 2010, parDans le cadre de la mise en place d’une plateforme ouverte, il est important pour les hébergeurs de pouvoir disposer de sauvegardes assez régulières pour parer à tout problème éventuel.
Pour réaliser cette tâche on se base sur deux plugins SPIP : Saveauto qui permet une sauvegarde régulière de la base de donnée sous la forme d’un dump mysql (utilisable dans phpmyadmin) mes_fichiers_2 qui permet de réaliser une archive au format zip des données importantes du site (les documents, les éléments (...)
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FFmpeg : no image is displayed when broadcasting flv made of speex / mellymoser and libx264
15 septembre 2013, par wvxvwffmpeg -f alsa -i hw:PCH,0 -f x11grab -s 1920x1200 -r 15 -i :0.0 \
-acodec nellymoser -ar 22050 -ac 1 -audio_preload 1000000 -vcodec libx264 \
-preset fast -pix_fmt yuv420p -s 1280x800 -maxrate 1000k -threads 0 -f flv "$URL"This is the command I'm using, but I don't get any visible output, only sound. FFmpeg gives no warnings. What am I doing wrong ?
If I remove -audio_preload, the streaming works, but the audio comes with increasing delay (grows about a minute every minute).
The output, as per request :
ffmpeg version 1.0.7 Copyright (c) 2000-2013 the FFmpeg developers
built on May 14 2013 21:59:35 with gcc 4.7.2 (GCC) 20121109 (Red Hat 4.7.2-8)
configuration: --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg --incdir=/usr/include/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man --arch=x86_64 --optflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic' --enable-bzlib --disable-crystalhd --enable-frei0r --enable-gnutls --enable-libass --enable-libcdio --enable-libcelt --enable-libdc1394 --disable-indev=jack --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-openal --enable-libopencv --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libv4l2 --enable-libvpx --enable-libx264 --enable-libxvid --enable-x11grab --enable-avfilter --enable-postproc --enable-pthreads --disable-static --enable-shared --enable-gpl --disable-debug --disable-stripping --shlibdir=/usr/lib64 --enable-runtime-cpudetect
libavutil 51. 73.101 / 51. 73.101
libavcodec 54. 59.100 / 54. 59.100
libavformat 54. 29.104 / 54. 29.104
libavdevice 54. 2.101 / 54. 2.101
libavfilter 3. 17.100 / 3. 17.100
libswscale 2. 1.101 / 2. 1.101
libswresample 0. 15.100 / 0. 15.100
libpostproc 52. 0.100 / 52. 0.100
[alsa @ 0xc5e8e0] Estimating duration from bitrate, this may be inaccurate
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, alsa, from 'hw:PCH,0':
Duration: N/A, start: 1379234523.677560, bitrate: 1536 kb/s
Stream #0:0: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s
[x11grab @ 0xc6cda0] device: :0.0 -> display: :0.0 x: 0 y: 0 width: 1920 height: 1200
[x11grab @ 0xc6cda0] shared memory extension found
[x11grab @ 0xc6cda0] Estimating duration from bitrate, this may be inaccurate
Input #1, x11grab, from ':0.0':
Duration: N/A, start: 1379234523.843273, bitrate: 1105920 kb/s
Stream #1:0: Video: rawvideo (BGR[0] / 0x524742), bgr0, 1920x1200, 1105920 kb/s, 15 tbr, 1000k tbn, 15 tbc
[libx264 @ 0xc59e00] VBV maxrate specified, but no bufsize, ignored
[libx264 @ 0xc59e00] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.2 AVX
[libx264 @ 0xc59e00] profile High, level 3.2
[libx264 @ 0xc59e00] 264 - core 128 r2223 f6a8615 - H.264/MPEG-4 AVC codec - Copyleft 2003-2012 - http://www.videolan.org/x264.html - options: cabac=1 ref=2 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=6 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=12 lookahead_threads=2 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=1 keyint=250 keyint_min=15 scenecut=40 intra_refresh=0 rc_lookahead=30 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
Output #0, flv, to 'rtmp://live.justin.tv/app/live_key':
Metadata:
encoder : Lavf54.29.104
Stream #0:0: Video: h264 ([7][0][0][0] / 0x0007), yuv420p, 1280x800, q=-1--1, 1k tbn, 15 tbc
Stream #0:1: Audio: nellymoser ([6][0][0][0] / 0x0006), 22050 Hz, mono, flt, 128 kb/s
Stream mapping:
Stream #1:0 -> #0:0 (rawvideo -> libx264)
Stream #0:0 -> #0:1 (pcm_s16le -> nellymoser)
Press [q] to stop, [?] for help
[alsa @ 0xc5e8e0] ALSA buffer xrun. -
How to transmux ismv file to mp4
21 septembre 2013, par PriyalI wanted to convert ismv file to mp4 without any packet loss.
I tried it with ffmpeg as :
ffmpeg -i input.ismv output.mp4
But the resultant mp4 file has some packet/frame loss due which video is distracted.
ffmpeg
command line return the following :Metadata: major_brand : isml minor_version : 1 compatible_brands: piffiso2 Duration: 00:03:57.18, start: 0.000000, bitrate: 1612 kb/s
Stream #0:0(und): Audio: wmapro (b[1][0][0] / 0x0162), 44100 Hz, stereo, fltp, 64 kb/s (default)Metadata: creation_time : 2013-09-20 06:18:40 handler_name : Audio
Stream #0:1(und): Video: vc1 (Advanced) (vc-1 / 0x312D6376), yuv420p, 640x480 [SAR 1:1 DAR 4:3], 1343 kb/s, 30 tbr, 10000k tbn, 60 tbc (default) Metadata: creation_time : 2013-09-20 06:18:40
Video Stream #0:2(und): Video: vc1 (Advanced) (vc-1 / 0x312D6376), yuv420p, 480x360 [SAR 1:1 DAR 4:3], 887 kb/s, 30 fps, 30 tbr, 10000k tbn, 60 tbc (default) Metadata: creation_time : 2013-09-20 06:18:40 handler_name : Video
Stream #0:3(und): Video: vc1 (Advanced) (vc-1 / 0x312D6376), yuv420p, 364x27 2 [SAR 136:136 DAR 91:68], 599 kb/s, 30 fps, 30 tbr, 10000k tbn, 60 tbc (default ) Metadata: creation_time : 2013-09-20 06:18:40 handler_name : Video
Stream #0:4(und): Video: vc1 (Advanced) (vc-1 / 0x312D6376), yuv420p, 276x20 8 [SAR 207:208 DAR 14283:10816], 400 kb/s, 30 fps, 30 tbr, 10000k tbn, 60 tbc (d efault) Metadata: creation_time : 2013-09-20 06:18:40 handler_name : Video
[wmapro @ 003fa000] Channel transform bit is not implemented. Update your FFmpeg version to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been implemented.
Error while decoding stream #0:0: Invalid data found when processing input [wmapro @ 003fa000] frame[0] would have to skip -9 bits[wmapro @ 003fa000] Packet loss detected! seq e vs d [wmapro @ 003fa000] Packet loss detected! seq d vs f [wmapro @ 003fa000] Reserved bit is not implemented. Update your FFmpeg version to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been implemented -
webm local udp streaming using FFMPEG
1er octobre 2013, par sinivI was just started to use ffmpeg recently and stumbled on this streaming problem.
Scenario : i want to live stream a webcam in local network. Both server and client will be using windows platform.Current feasible solution : using ffmpeg simple command line
to test it quickly i tried to locally stream it (the input doesn't really matter btw in this question).
On server -> ffmpeg -f dshow -i video="cam1":audio="mic1" -r 30 -g 0 -vcodec h264 -acodec libmp3lame -tune zerolatency -preset ultrafast -f mpegts udp://localhost:6789
On client(the same computer) -> ffplay udp://localhost:6789The above works just fine, except for the latency, which i'm getting at about 1-2 second delay.
Now i want to try to change the encoder to use libvpx (vp8) for video and vorbis for audio (i changed the input to a pre-recorded h264 video, but it really doesn't matter)
On server
>ffmpeg -i "suits.mp4" -r 30 -g 0 -vcodec libvpx -acodec vorbis -strict -2 -f webm -f mpegts udp://localhost:6789
On client(the same computer) -> ffplay udp://localhost:6789
However this doesn't work... And below are console outputs:
> onserver ->
> ffmpeg version N-56165-gae12d65 Copyright (c) 2000-2013 the FFmpeg
> developers built on Sep 10 2013 19:42:46 with gcc 4.7.3 (GCC)
> configuration: --enable-gpl --enable-version3 --disable-w32threads
> --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetype --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 52. 43.100 / 52. 43.100 libavcodec 55. 31.101 / 55. 31.101 libavformat 55. 16.102 / 55. 16.102 libavdevice 55. 3.100 / 55. 3.100 libavfilter 3. 84.100 / 3. 84.100 libswscale 2. 5.100 /
> 2. 5.100 libswresample 0. 17.103 / 0. 17.103 libpostproc 52. 3.100 / 52. 3.100 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'Suits.mp4': Metadata:
> major_brand : isom
> minor_version : 1
> compatible_brands: isom
> creation_time : 2011-09-08 11:43:25 Duration: 00:42:14.87, start: 0.000000, bitrate: 882 kb/s
> Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv, bt709), 720x402 [SAR 1:1 DAR 120:67], 750 kb/s, 23.98 fps,
> 23.98 tbr, 24k tbn, 47.95 tbc (default)
> Metadata:
> creation_time : 2011-09-08 11:43:25
> Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 126 kb/s (default)
> Metadata:
> creation_time : 2011-09-08 11:43:25 [libvpx @ 05392a80] v1.2.0 Output #0, mpegts, to 'udp://localhost:6789': Metadata:
> major_brand : isom
> minor_version : 1
> compatible_brands: isom
> encoder : Lavf55.16.102
> Stream #0:0(und): Video: vp8 (libvpx), yuv420p, 720x402 [SAR 1:1 DAR 120:67], q=-1--1, 200 kb/s, 90k tbn, 30 tbc (default)
> Metadata:
> creation_time : 2011-09-08 11:43:25
> Stream #0:1(und): Audio: vorbis, 48000 Hz, stereo, fltp (default)
> Metadata:
> creation_time : 2011-09-08 11:43:25 Stream mapping: Stream #0:0 -> #0:0 (h264 -> libvpx) Stream #0:1 -> #0:1 (aac -> vorbis) Press [q] to stop, [?] for help frame=42535 fps= 51 q=0.0 Lsize=
> 143539kB time=00:23:38.28 bitrate= 829.1kbits/s dup=8541 drop=0
> video:99155kB audio:28125kB subtitle:0 global headers:3kB muxing
> overhead 12.772155% Received signal 2: terminating.
> on client
> ffplay version N-56165-gae12d65 Copyright (c) 2003-2013 the FFmpeg
> developers built on Sep 10 2013 19:42:46 with gcc 4.7.3 (GCC)
> configuration: --enable-gpl --enable-version3 --disable-w32threads
> --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetype --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 52. 43.100 / 52. 43.100 libavcodec 55. 31.101 / 55. 31.101 libavformat 55. 16.102 / 55. 16.102 libavdevice 55. 3.100 / 55. 3.100 libavfilter 3. 84.100 / 3. 84.100 libswscale 2. 5.100 /
> 2. 5.100 libswresample 0. 17.103 / 0. 17.103 libpostproc 52. 3.100 / 52. 3.100
> nan : 0.000 fd= 0 aq= 0KB vq= 0KB sq= 0B f=0/0 [mpegts @ 02eb8620] probed stream 0 failed
> nan : 0.000 fd= 0 aq= 0KB vq= 0KB sq= 0B f=0/0 [mp3 @ 02ed75a0] Header missing
> Last message repeated 1 times [mp3 @ 02ed75a0] Header missing
> La Last message repeated 13 times
> nan : 0.000 fd= 0 aq= 0KB vq= 0KB sq= 0B f=0/0 [mp3 @ 02ed75a0] Header missing Last message repeated 13 times
> nan : 0.000 fd= 0 aq= 0KB vq= 0KB sq= 0B f=0/0 [mp3 @ 02ed75a0] Header missing Last message repeated 9 times
> nan : 0.000 fd= 0 aq= 0KB vq= 0KB sq= 0B f=0/0 [mp3 @ 02ed75a0] Header missing [mpegts @ 02eb8620] decoding for
> stream 1 failed [mpegts @ 02eb8620] Could not find codec parameters
> for stream 0 (Unknown: none ([6][0][0][0] / 0x0006)): unknown codec
> Consider increasing the value for the 'analyzeduration' and
> 'probesize' options [mpegts @ 02eb8620] Could not find codec
> parameters for stream 1 (Audio: mp3 ([6][0][0][0] / 0x0006), 0
> channels, s16p): unspecified frame size Consider increasing the value
> for the 'analyzeduration' and 'probesize' options
> udp://localhost:6789: could not find codec parametersSo does the point to point streaming for ffmpeg just doesn't work for vp8 or am i missing something ? Btw, the end goal is to create a similar video chat based framework and i'll appreciate any suggestion. I'm reading up on webRTC now.