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Autres articles (42)
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Les autorisations surchargées par les plugins
27 avril 2010, parMediaspip core
autoriser_auteur_modifier() afin que les visiteurs soient capables de modifier leurs informations sur la page d’auteurs -
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 (...) -
Support audio et vidéo HTML5
10 avril 2011MediaSPIP utilise les balises HTML5 video et audio pour la lecture de documents multimedia en profitant des dernières innovations du W3C supportées par les navigateurs modernes.
Pour les navigateurs plus anciens, le lecteur flash Flowplayer est utilisé.
Le lecteur HTML5 utilisé a été spécifiquement créé pour MediaSPIP : il est complètement modifiable graphiquement pour correspondre à un thème choisi.
Ces technologies permettent de distribuer vidéo et son à la fois sur des ordinateurs conventionnels (...)
Sur d’autres sites (10546)
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avcodec/dcahuff : Combine tables, use ff_init_vlc_from_lengths()
6 septembre 2022, par Andreas Rheinhardtavcodec/dcahuff : Combine tables, use ff_init_vlc_from_lengths()
Up until now, initializing the dca VLC tables uses ff_init_vlc_sparse()
with length tables of type uint8_t and code tables of type uint16_t
(except for the LBR tables, which uses length and symbols of type
uint8_t ; these tables are interleaved). In case of the quant index
codebooks these arrays were accessed via tables of pointers to the
individual tables.This commit changes this : First, we switch to ff_init_vlc_from_lengths()
to replace the uint16_t code tables by uint8_t symbol tables
(this necessitates ordering the tables from left-to-right in the tree
first). These symbol tables are interleaved with the length tables.Furthermore, these tables are combined in order to remove the table of
pointers to individual tables, thereby avoiding relocations (for x64
elf systems this amounts to 96*24B = 2304B saved in .rela.dyn) and
saving 1280B from .data.rel.ro (for 64bit systems). Meanwhile the
savings in .rodata amount to 2709 + 2 * 334 = 3377B. Due to padding
the actual savings are higher : The ELF x64 ABI requires objects >= 16B
to be padded to 16B and lots of the tables have 2^n + 1 elements
of these were from replacing uint16_t codes with uint8_t symbols ;
the rest was due to the fact that combining the tables eliminated
padding (the ELF x64 ABI requires objects >= 16B to be padded to 16B
and lots of the tables have 2^n + 1 elements)). Taking this into
account gives savings of 4548B. (GCC by default uses an even higher
alignment (controlled by -malign-data) ; for it the savings are 5748B.)These changes also necessitated to modify the init code for
the encoder tables.Signed-off-by : Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
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Dump WebRTC stream to a file
20 novembre 2014, par MondainI’d like to capture the audio and video from a WebRTC stream to a file or pair of files, if audio and video require their own individual files. The audio and video are not muxed together and are known to be available on a set of server udp ports :
Port Encoding 5000 - VP8 video 5001 - RTCP (for video) 5002 - Opus audio @48kHz 2 channels 5003 - RTCP (for audio)
The SDP file / data is not available and DTLS may be used.
I would prefer to use avconv or ffmpeg to capture the stream, unless a better tool is suggested.
Edit : I’ve found that this as inquired will most likely not work. Until I hear otherwise, none of these tools support the initial DTLS handshake followed by the data transmission via SRTP.
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FFMPEG loop to trim silence at beginning and end of file
23 août 2021, par Juan PabloI'm trying to take a folder that contains mp3 and wave files and trim the silence at the beginning and the end in all the individual files.


for f in *.{mp3,wav}; do ffmpeg -i "$f" -af "silenceremove=start_periods=1:start_duration=1:start_threshold=-60dB:detection=peak,aformat=dblp,areverse,silenceremove=start_periods=1:start_duration=1:start_threshold=-60dB:detection=peak,aformat=dblp,areverse" "outputs/${f%.*}.mp3"; done



I keep receiving an error that f was unexpected. Can someone tell me what I'm doing wrong ?