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Autres articles (52)
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Encodage et transformation en formats lisibles sur Internet
10 avril 2011MediaSPIP transforme et ré-encode les documents mis en ligne afin de les rendre lisibles sur Internet et automatiquement utilisables sans intervention du créateur de contenu.
Les vidéos sont automatiquement encodées dans les formats supportés par HTML5 : MP4, Ogv et WebM. La version "MP4" est également utilisée pour le lecteur flash de secours nécessaire aux anciens navigateurs.
Les documents audios sont également ré-encodés dans les deux formats utilisables par HTML5 :MP3 et Ogg. La version "MP3" (...) -
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 -
Ajouter notes et légendes aux images
7 février 2011, parPour pouvoir ajouter notes et légendes aux images, la première étape est d’installer le plugin "Légendes".
Une fois le plugin activé, vous pouvez le configurer dans l’espace de configuration afin de modifier les droits de création / modification et de suppression des notes. Par défaut seuls les administrateurs du site peuvent ajouter des notes aux images.
Modification lors de l’ajout d’un média
Lors de l’ajout d’un média de type "image" un nouveau bouton apparait au dessus de la prévisualisation (...)
Sur d’autres sites (11170)
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Command line ffmpeg, to insert part of a mp3 audio to another file
14 janvier 2020, par tgkprogWant to copy from 3rd second of
b.mp3
till 2:05 seconds ofb.mp3
and append to the end ofa.mp3
(its 3:23 long).Command :
ffmpeg -i a.mp3 -i b.mp3
Output :
ffmpeg version 4.0.4-0ubuntu1 Copyright (c) 2000-2019 the FFmpeg developers built with gcc 8 (Ubuntu 8.2.0-7ubuntu1)
configuration: --prefix=/usr --extra-version=0ubuntu1 --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-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-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared
libavutil 56. 14.100 / 56. 14.100
libavcodec 58. 18.100 / 58. 18.100
libavformat 58. 12.100 / 58. 12.100
libavdevice 58. 3.100 / 58. 3.100
libavfilter 7. 16.100 / 7. 16.100
libavresample 4. 0. 0 / 4. 0. 0
libswscale 5. 1.100 / 5. 1.100
libswresample 3. 1.100 / 3. 1.100
libpostproc 55. 1.100 / 55. 1.100
Input #0, mp3, from 'a.mp3':
Metadata:
encoder : Lavf58.12.100
Duration: 00:02:48.00, start: 0.011021, bitrate: 144 kb/s
Stream #0:0: Audio: mp3, 48000 Hz, stereo, fltp, 144 kb/s
Metadata:
encoder : Lavc58.18
Input #1, mp3, from 'b.mp3':
Metadata:
major_brand : isom
minor_version : 512
compatible_brands: isomiso2mp41
encoder : Lavf58.12.100
Duration: 00:01:33.02, start: 0.011995, bitrate: 117 kb/s
Stream #1:0: Audio: mp3, 44100 Hz, stereo, fltp, 117 kb/s
Metadata:
encoder : Lavc58.18
At least one output file must be specified -
PyQt-thread. Get dynamically output
1er janvier 2020, par ZProI use PyQt-thread for parallel conversion of mp3 files to aac via ffmpeg.
Here is my code :class SubprocessThread(QThread):
signal = pyqtSignal('PyQt_PyObject')
def __init__(self, command, args):
QThread.__init__(self)
self.command = command
self.args = args
def __del__(self):
self.wait()
def run(self):
output = subprocess.check_output('{0} {1}'.format(self.command, self.args), shell=True).split()
self.signal.emit(output)And here is example of usage :
threads = []
for part in parts.keys():
args = "-i \'{0}.mp3\' -c:a aac -b:a {1}k \'{2}.m4a\'".format(
os.path.join(tmp_dir, str(part)),
int(self.bitrate_cbx.currentText()),
os.path.join(tmp_dir, str(part)))
print(args) # debug
ffmpeg_thread = SubprocessThread('ffmpeg', args)
ffmpeg_thread.signal.connect(self.on_data_ready)
threads.append(ffmpeg_thread)
ffmpeg_thread.start()
self.threads_count += 1I want to make progress bar, based on conversion, but ffmpeg always updates last string in his output (when conversion in progress).
Here is an example of ffmpeg output while files are converting :user@host$ ffmpeg -i '/home/user/001.mp3' -c:a aac -b:a 128k -vn '/home/user/test.m4a'
ffmpeg version n4.2.1 Copyright (c) 2000-2019 the FFmpeg developers
built with gcc 9.2.0 (GCC)
configuration: --prefix=/usr --disable-debug --disable-static --disable-stripping --enable-fontconfig --enable-gmp --enable-gnutls --enable-gpl --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libdav1d --enable-libdrm --enable-libfreetype --enable-libfribidi --enable-libgsm --enable-libiec61883 --enable-libjack --enable-libmodplug --enable-libmp3lame --enable-libopencore_amrnb --enable-libopencore_amrwb --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libv4l2 --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxcb --enable-libxml2 --enable-libxvid --enable-nvdec --enable-nvenc --enable-omx --enable-shared --enable-version3
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
libswscale 5. 5.100 / 5. 5.100
libswresample 3. 5.100 / 3. 5.100
libpostproc 55. 5.100 / 55. 5.100
Input #0, mp3, from '/home/user/001.mp3':
Metadata:
encoder : Lavf57.41.100
title : test
artist : test
album_artist : test
album : test
composer : test
genre : test
date : 2018
Duration: 00:12:38.02, start: 0.025056, bitrate: 192 kb/s
Stream #0:0: Audio: mp3, 44100 Hz, stereo, fltp, 192 kb/s
Metadata:
encoder : Lavc57.48
Stream #0:1: Video: mjpeg (Baseline), yuvj420p(pc, bt470bg/unknown/unknown), 500x500 [SAR 1:1 DAR 1:1], 90k tbr, 90k tbn, 90k tbc (attached pic)
Metadata:
comment : Cover (front)
Stream mapping:
Stream #0:0 -> #0:0 (mp3 (mp3float) -> aac (native))
Press [q] to stop, [?] for help
Output #0, ipod, to '/home/user/test.m4a':
Metadata:
date : test
title : test
artist : test
album_artist : test
album : test
composer : test
genre : test
encoder : Lavf58.29.100
Stream #0:0: Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 128 kb/s
Metadata:
encoder : Lavc58.54.100 aac
size= 12107kB time=00:12:38.01 bitrate= 130.8kbits/s speed=79.2xHow can I receive this data (string, that begins from "size=...") from my parallel QThreads to calculate overall progress ?
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FFMPEG detect silence command runs correctly but does not give the silence duration
8 janvier 2020, par AizayousafI have a .wav audio file and I need to extract silence/pause duration in this file. I’m using ffmpeg with silence detect filter but I’m unable to understand why its not giving silence duration with this file while it gives result with other files. Can anyone help me to understand the out given below that why its not showing detected silences.
Input Command :
ffmpeg -i "input.wav" -af silencedetect=noise=-30dB:d=0.5 -f null -
OutPut
ffmpeg version 4.2.1 Copyright (c) 2000-2019 the FFmpeg developers
built with gcc 9.1.1 (GCC) 20190807
configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls -- enable-iconv --enable-libass --enable-libdav1d --enable-libbluray --enable-libfreetype --enable-
libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-
libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libtheora --enable-
libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --
enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --
enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --
enable-libaom --enable-libmfx --enable-amf --enable-ffnvcodec --enable-cuvid --enable-d3d11va -- enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt
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
libswscale 5. 5.100 / 5. 5.100
libswresample 3. 5.100 / 3. 5.100
libpostproc 55. 5.100 / 55. 5.100
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, wav, from 'D:\Research\PhD\Carolina\AD\wav\media.io_Wakeman_Rhyne_001_01.wav':
Duration: 00:17:38.04, bitrate: 1411 kb/s
Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s
Stream mapping:
Stream #0:0 -> #0:0 (pcm_s16le (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
Output #0, null, to 'pipe:':