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Autres articles (56)
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Multilang : améliorer l’interface pour les blocs multilingues
18 février 2011, parMultilang est un plugin supplémentaire qui n’est pas activé par défaut lors de l’initialisation de MediaSPIP.
Après son activation, une préconfiguration est mise en place automatiquement par MediaSPIP init permettant à la nouvelle fonctionnalité d’être automatiquement opérationnelle. Il n’est donc pas obligatoire de passer par une étape de configuration pour cela. -
Ajouter des informations spécifiques aux utilisateurs et autres modifications de comportement liées aux auteurs
12 avril 2011, parLa manière la plus simple d’ajouter des informations aux auteurs est d’installer le plugin Inscription3. Il permet également de modifier certains comportements liés aux utilisateurs (référez-vous à sa documentation pour plus d’informations).
Il est également possible d’ajouter des champs aux auteurs en installant les plugins champs extras 2 et Interface pour champs extras. -
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 (11678)
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ffmpeg take input and out put as webm on port
28 septembre 2019, par RussellHarrowerI am wondering how I can take an audio input and output it as a webm
This is what I thought would work to get it to rtp out but running into issue.
ffmpeg -i http://stream.radiomedia.com.au:8003/stream -c copy -f rtp rtp://127.0.0.1/streamID:54321it returns
ffmpeg -i http://stream.radiomedia.com.au:8003/stream -c copy -f rtp rtp://127.0.0.1/streamID:54321 ffmpeg version 4.2.1-0york0~18.04 Copyright (c) 2000-2019 the FFmpeg developers
built with gcc 7 (Ubuntu 7.4.0-1ubuntu1~18.04.1)
configuration: --prefix=/usr --extra-version='0york0~18.04' --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --arch=amd64 --enable-gpl --disable-stripping --enable-avresample --disable-filter=resample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libcodec2 --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libjack --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librsvg --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-lv2 --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-nonfree --enable-libfdk-aac --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libx264 --enable-shared
libavutil 56. 31.100 / 56. 31.100
libavcodec 58. 54.100 / 58. 54.100
libavformat 58. 29.100 / 58. 29.100
libavdevice 58. 8.100 / 58. 8.100
libavfilter 7. 57.100 / 7. 57.100
libavresample 4. 0. 0 / 4. 0. 0
libswscale 5. 5.100 / 5. 5.100
libswresample 3. 5.100 / 3. 5.100
libpostproc 55. 5.100 / 55. 5.100
[mp3 @ 0x564cf51de940] invalid concatenated file detected - using bitrate for duration
Input #0, mp3, from 'http://stream.radiomedia.com.au:8003/stream':
Metadata:
track : 1
title : DNR1 Signal 1
comment : www.dvdvideosoft.com
date : 2019
icy-br : 128
icy-description : Radio Media PTY LTD
icy-genre : Indie
icy-name : DRN1
icy-pub : 0
icy-url : https://www.drn1.com.au
StreamTitle :
Duration: N/A, start: 0.025057, bitrate: 320 kb/s
Stream #0:0: Audio: mp3, 44100 Hz, stereo, fltp, 320 kb/s
Metadata:
encoder : LAME3.99r
Output #0, rtp, to 'rtp://127.0.0.1/streamID:54321':
Metadata:
track : 1
title : DNR1 Signal 1
comment : www.dvdvideosoft.com
date : 2019
icy-br : 128
icy-description : Radio Media PTY LTD
icy-genre : Indie
icy-name : DRN1
icy-pub : 0
icy-url : https://www.drn1.com.au
StreamTitle :
encoder : Lavf58.29.100
Stream #0:0: Audio: mp3, 44100 Hz, stereo, fltp, 320 kb/s
Metadata:
encoder : LAME3.99r
SDP:
v=0
o=- 0 0 IN IP4 127.0.0.1
s=DNR1 Signal 1
c=IN IP4 127.0.0.1
t=0 0
a=tool:libavformat 58.29.100
m=audio 0 RTP/AVP 14
b=AS:320
a=control:streamid=0
Stream mapping:
Stream #0:0 -> #0:0 (copy)
Press [q] to stop, [?] for help
av_interleaved_write_frame(): Invalid argument Error writing trailer of rtp://127.0.0.1/streamID:54321: Invalid argument
size= 0kB time=-00:00:00.02 bitrate=N/A speed=N/A video:0kB audio:1kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown
Conversion failed! -
ffmpeg take input and output as webm on port
12 octobre 2019, par RussellHarrowerI am wondering how I can take an audio input and output it as a webm
This is what I thought would work to get it to rtp out but running into issue.
ffmpeg -i http://stream.radiomedia.com.au:8003/stream -c copy -f rtp rtp://127.0.0.1/streamID:54321it returns
ffmpeg -i http://stream.radiomedia.com.au:8003/stream -c copy -f rtp rtp://127.0.0.1/streamID:54321 ffmpeg version 4.2.1-0york0~18.04 Copyright (c) 2000-2019 the FFmpeg developers
built with gcc 7 (Ubuntu 7.4.0-1ubuntu1~18.04.1)
configuration: --prefix=/usr --extra-version='0york0~18.04' --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --arch=amd64 --enable-gpl --disable-stripping --enable-avresample --disable-filter=resample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libcodec2 --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libjack --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librsvg --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-lv2 --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-nonfree --enable-libfdk-aac --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libx264 --enable-shared
libavutil 56. 31.100 / 56. 31.100
libavcodec 58. 54.100 / 58. 54.100
libavformat 58. 29.100 / 58. 29.100
libavdevice 58. 8.100 / 58. 8.100
libavfilter 7. 57.100 / 7. 57.100
libavresample 4. 0. 0 / 4. 0. 0
libswscale 5. 5.100 / 5. 5.100
libswresample 3. 5.100 / 3. 5.100
libpostproc 55. 5.100 / 55. 5.100
[mp3 @ 0x564cf51de940] invalid concatenated file detected - using bitrate for duration
Input #0, mp3, from 'http://stream.radiomedia.com.au:8003/stream':
Metadata:
track : 1
title : DNR1 Signal 1
comment : www.dvdvideosoft.com
date : 2019
icy-br : 128
icy-description : Radio Media PTY LTD
icy-genre : Indie
icy-name : DRN1
icy-pub : 0
icy-url : https://www.drn1.com.au
StreamTitle :
Duration: N/A, start: 0.025057, bitrate: 320 kb/s
Stream #0:0: Audio: mp3, 44100 Hz, stereo, fltp, 320 kb/s
Metadata:
encoder : LAME3.99r
Output #0, rtp, to 'rtp://127.0.0.1/streamID:54321':
Metadata:
track : 1
title : DNR1 Signal 1
comment : www.dvdvideosoft.com
date : 2019
icy-br : 128
icy-description : Radio Media PTY LTD
icy-genre : Indie
icy-name : DRN1
icy-pub : 0
icy-url : https://www.drn1.com.au
StreamTitle :
encoder : Lavf58.29.100
Stream #0:0: Audio: mp3, 44100 Hz, stereo, fltp, 320 kb/s
Metadata:
encoder : LAME3.99r
SDP:
v=0
o=- 0 0 IN IP4 127.0.0.1
s=DNR1 Signal 1
c=IN IP4 127.0.0.1
t=0 0
a=tool:libavformat 58.29.100
m=audio 0 RTP/AVP 14
b=AS:320
a=control:streamid=0
Stream mapping:
Stream #0:0 -> #0:0 (copy)
Press [q] to stop, [?] for help
av_interleaved_write_frame(): Invalid argument Error writing trailer of rtp://127.0.0.1/streamID:54321: Invalid argument
size= 0kB time=-00:00:00.02 bitrate=N/A speed=N/A video:0kB audio:1kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown
Conversion failed! -
ffmpeg clean all noise background silences in a poscast
23 mars 2019, par fireDevelop.comI have hundreds of podcast without music, just the voice and the room silence.
In the silences, I have many clicks, respirations, etc...
I need to clean all silences with a script, keeping intact the voice.In this picture you can see my dirty silences
And here the result I want in all my audios
When I use some scripts of sox. I don`t get the result I spect because the voice is affected by the script, the room-silence disappear and some clic still in the silences.
Then in order to keep intact the voice, I want to do this :
- Delete all the silences longer than 3 seconds.
-
Split all the audio and silences with in a sequence numbers. ie. :
- 001-Silence-2.0seconds.wav
- 002-voice.wav
- 003-Silence-0.25seconds.wav
- 004-voice.wav
- 005-Silence-0.75seconds.wav
- 006-voice.wav
- ...
- ...
-
Before, run the script I created manually many files with silences of diferents silences I will use :
- myManuallySilence-0.25seconds.wav
- myManuallySilence-0.50seconds.wav
- myManuallySilence-0.75seconds.wav
- myManuallySilence-0.1seconds.wav
- myManuallySilence-1.25seconds.wav
- ...
- ...
- myManuallySilence-2.50seconds.wav
- myManuallySilence-2.75seconds.wav
- myManuallySilence-3.0seconds.wav
- the script will check the dirty silences duration and replace by the files myManuallySilence-x.xseconds.wav
- merge all files in one wav file, with the original voice and all the silences cleanned.
At the moment I have only this script :
# get the path of Adobe Audition and add timestamp in the output
filename
fileName=out
current_time=$(date "+%Y.%m.%d-%H.%M.%S")
newFileName=$fileName.$current_time.wav
#yourPathAPP=/Applications/Adobe\ Audition\ CC\ 2019/Adobe\ Audition\
CC\ 2019.app
yourPathAPP=/Volumes/6TB/Applications/ocenaudio.app
# # First denoise audio
# ## Get noise sample
ffmpeg -i in.wav -vn -ss 00:00:00 -t 00:00:01 noise-sample.wav
# ## Create noise profile
sox noise-sample.wav -n noiseprof noise.prof
# ## Clean audio from noise
sox in.wav $newFileName noisered noise.prof 0.50
# # Split audio by noise
sox -V3 $newFileName output.wav silence 1 00:00:02.000 - 80d 1
00:00:02.000 -80d : newfile : restart
# ####### (these settings worked for my computer mic - maybe we need to
finetune them later) #######Is getting all the voice in separate files like this :
output001.wav
output002.wav
output003.wav
output004.wav
...
output00x.wavPlease, any suggestion will be appreciated.
Thanks so much in advance !
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