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Autres articles (39)
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Gestion générale des documents
13 mai 2011, parMédiaSPIP ne modifie jamais le document original mis en ligne.
Pour chaque document mis en ligne il effectue deux opérations successives : la création d’une version supplémentaire qui peut être facilement consultée en ligne tout en laissant l’original téléchargeable dans le cas où le document original ne peut être lu dans un navigateur Internet ; la récupération des métadonnées du document original pour illustrer textuellement le fichier ;
Les tableaux ci-dessous expliquent ce que peut faire MédiaSPIP (...) -
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. -
HTML5 audio and video support
13 avril 2011, parMediaSPIP uses HTML5 video and audio tags to play multimedia files, taking advantage of the latest W3C innovations supported by modern browsers.
The MediaSPIP player used has been created specifically for MediaSPIP and can be easily adapted to fit in with a specific theme.
For older browsers the Flowplayer flash fallback is used.
MediaSPIP allows for media playback on major mobile platforms with the above (...)
Sur d’autres sites (5507)
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DXVA2/D3D11 encoders not available in FFmpeg for no apparent reason
3 juillet 2020, par user1134621this is a weird one. I am trying to set up FFmpeg for hardware encoding of a live input into h264. I got it working fine on Linux with VAAPI. On Windows I was probably successful as well with NVENC but I do not have a NVENC capable card to check. What has been giving me a headache is encoding through DXVA2 or D3D11VA. FFmpeg picks up available hardware but it will not give me any suitable encoders. I even built FFmpeg myself to make sure that all the CONFIG_DXVA_HWACCEL switches are enabled, I can grep the binary for
h264_dxva2
etc. and see that they are there. However, callingavcodec_find_encoder_by_name
always returns NULL.h264_vaapi
orh264_nvenc
work fine though. I am obviously missing something stupid. Do you have any suggestions where to look ? Thanks.

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FFMPEG update from 5.0 to 6.0 out_0_0 buffer queued [closed]
16 mai 2023, par KevittoI've been using ffmpeg 5.0 for some time, encoding an audio stream to an rtp server, but since I updated to ffmpeg 6.0 I get this :


[out_0_0 @ 0x55ac187b60] 100 buffers queued in out_0_0, something may be wrong.



Below is the ffmpeg call :


ffmpeg -re -f alsa -i default:CARD:card1 -ac 2 -af aresample=async=1 -acodec libopus -b:a 48000 -f rtp "rtp://127.0.0.1:5002"



And here is the full startup log :


ffmpeg version 549430e Copyright (c) 2000-2023 the FFmpeg developers
 built with gcc 10 (Debian 10.2.1-6)
 configuration: --extra-cflags=-I/usr/local/include --extra-ldflags=-L/usr/local/lib --extra-libs='-lpthread -lm -latomic' --arch=arm64 --enable-gmp --enable-gpl --enable-libopus --enable-nonfree --enable-version3 --target-os=linux --enable-pthreads --enable-openssl --enable-hardcoded-tables
 libavutil 58. 2.100 / 58. 2.100
 libavcodec 60. 3.100 / 60. 3.100
 libavformat 60. 3.100 / 60. 3.100
 libavdevice 60. 1.100 / 60. 1.100
 libavfilter 9. 3.100 / 9. 3.100
 libswscale 7. 1.100 / 7. 1.100
 libswresample 4. 10.100 / 4. 10.100
 libpostproc 57. 1.100 / 57. 1.100
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, alsa, from 'default:CARD=pisound':
 Duration: N/A, start: 1684250059.973334, bitrate: 1536 kb/s
 Stream #0:0: Audio: pcm_s16le, 48000 Hz, 2 channels, s16, 1536 kb/s
Stream mapping:
 Stream #0:0 -> #0:0 (pcm_s16le (native) -> opus (libopus))
Press [q] to stop, [?] for help
Output #0, rtp, to 'rtp://127.0.0.1:5002':
 Metadata:
 encoder : Lavf60.3.100
 Stream #0:0: Audio: opus, 48000 Hz, stereo, s16, 48 kb/s
 Metadata:
 encoder : Lavc60.3.100 libopus
SDP:
v=0
o=- 0 0 IN IP4 127.0.0.1
s=No Name
c=IN IP4 127.0.0.1
t=0 0
a=tool:libavformat 60.3.100
m=audio 5002 RTP/AVP 97
b=AS:48
a=rtpmap:97 opus/48000/2
a=fmtp:97 sprop-stereo=1

size= 0kB time=-577014:32:22.77 bitrate= -0.0kbits/s speed=N/A 
size= 0kB time=-577014:32:22.77 bitrate= -0.0kbits/s speed=N/A 
[out_0_0 @ 0x559f530c60] 100 buffers queued in out_0_0, something may be wrong.
size= 0kB time=-577014:32:22.77 bitrate= -0.0kbits/s speed=N/A 
size= 0kB time=-577014:32:22.77 bitrate= -0.0kbits/s speed=N/A 
size= 0kB time=-00:00:00.00 bitrate= -0.0kbits/s speed=N/A 
size= 0kB time=-00:00:00.00 bitrate= -0.0kbits/s speed=N/A 
[alsa @ 0x559f4db460] ALSA buffer xrun.
size= 6kB time=00:00:02.81 bitrate= 16.7kbits/s speed=0.93x 
size= 6kB time=00:00:02.81 bitrate= 16.7kbits/s speed=0.797x 
size= 6kB time=00:00:02.83 bitrate= 16.6kbits/s speed=0.703x 
size= 6kB time=00:00:02.83 bitrate= 16.6kbits/s speed=0.624x 
size= 6kB time=00:00:02.83 bitrate= 16.6kbits/s speed=0.562x 
size= 6kB time=00:00:02.83 bitrate= 16.6kbits/s speed=0.511x 
[alsa @ 0x559f4db460] ALSA buffer xrun.
size= 10kB time=00:00:05.67 bitrate= 14.8kbits/s speed=0.939x 
size= 10kB time=00:00:05.67 bitrate= 14.8kbits/s speed=0.866x 
size= 10kB time=00:00:05.67 bitrate= 14.8kbits/s speed=0.805x 
size= 10kB time=00:00:05.67 bitrate= 14.8kbits/s speed=0.751x 
size= 10kB time=00:00:05.69 bitrate= 14.8kbits/s speed=0.707x 
[alsa @ 0x559f4db460] ALSA buffer xrun.
size= 13kB time=00:00:05.95 bitrate= 17.8kbits/s speed=0.696x 
size= 16kB time=00:00:08.51 bitrate= 15.2kbits/s speed=0.939x 
size= 16kB time=00:00:08.51 bitrate= 15.2kbits/s speed=0.89x 
size= 16kB time=00:00:08.53 bitrate= 15.2kbits/s speed=0.847x 
size= 16kB time=00:00:08.53 bitrate= 15.2kbits/s speed=0.806x 
size= 16kB time=00:00:08.53 bitrate= 15.2kbits/s speed=0.77x 
[alsa @ 0x559f4db460] ALSA buffer xrun.
size= 21kB time=00:00:11.37 bitrate= 14.8kbits/s speed=0.981x 



I tried changing the output to
-f null /dev/null
to see if the rtp was the issue, but I get the same thing. I made sure the user running it was a member to the "audio" group andarecord -l
andaplay -l
both show the card with the right name and information. I even tried to use its hw code instead of the default name, and same issue.

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x264vfw - is the H.264/MPEG-4 AVC codec advisable to use ?
26 avril 2012, par Dhaval KariyaI have generated a graph using the GraphEdit utility of Directshow.
My Graph is as follow :
osprey card -> x264vfw - H.264/MPEG-4 AVC codec -> GDCL Multiplexer -> File writer (.mp4
file)I used
x264vfw - H.264/MPEG-4 AVC codec
, downloaded from
http://sourceforge.net/projects/x264vfw/.I want to know if it is advisable to use this codec to encode the live video streams ?
And what about the support and licensing for it ?