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Autres articles (20)
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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 (...) -
Encoding and processing into web-friendly formats
13 avril 2011, parMediaSPIP automatically converts uploaded files to internet-compatible formats.
Video files are encoded in MP4, Ogv and WebM (supported by HTML5) and MP4 (supported by Flash).
Audio files are encoded in MP3 and Ogg (supported by HTML5) and MP3 (supported by Flash).
Where possible, text is analyzed in order to retrieve the data needed for search engine detection, and then exported as a series of image files.
All uploaded files are stored online in their original format, so you can (...) -
Contribute to a better visual interface
13 avril 2011MediaSPIP is based on a system of themes and templates. Templates define the placement of information on the page, and can be adapted to a wide range of uses. Themes define the overall graphic appearance of the site.
Anyone can submit a new graphic theme or template and make it available to the MediaSPIP community.
Sur d’autres sites (5901)
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fftools/qsv : add extra_hw_frames support
25 juillet 2018, par Zhong Lifftools/qsv : add extra_hw_frames support
Currently extra_hw_frames can't be applied to qsv since it
doesn't call function avcodec_get_hw_frames_parameters().Give an option to fix ticket #7261 though it is not a perfect soultion
(allocate the minimum pool size internally and automatically).Signed-off-by : Zhong Li <zhong.li@intel.com>
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Silence in RTP stream in ffmpeg
9 octobre 2020, par OdatiaI am trying to record a webRTC stream. The stream is coming in in the OPUS codec, and is then being piped into ffmpeg for recording, using the
-c:a copy
flag. However, if there are missing packets, for example, if I mute my microphone (or even stay quiet for a short period), when the audio is played back (webm file), it jumps across these "silences".

I have tried to use the DTX option to insert comfort noise, but my input is 4800MHz which is not supported.


My full mpeg command is :


ffmpeg -nostdin -protocol_whitelist file,rtp,udp -use_wallclock_as_timestamps 1 -fflags +genpts -i /tmp/sdpfilesfhmw4rymzy8wv25ybm610xte -map 0:a:0 -c:a copy -f webm -flags +global_header -y ./server/lib/recording/test-icc-networks-xzsyoe1f.webm



Any help would be appreciated !


Thanks,
Joe


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Merge commit 'e4cdef00263dc8b3c8de9d34ceacd00dc68979c0'
12 février 2018, par Mark Thompson