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Autres articles (85)
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L’agrémenter visuellement
10 avril 2011MediaSPIP est basé sur un système de thèmes et de squelettes. Les squelettes définissent le placement des informations dans la page, définissant un usage spécifique de la plateforme, et les thèmes l’habillage graphique général.
Chacun peut proposer un nouveau thème graphique ou un squelette et le mettre à disposition de la communauté. -
Websites made with MediaSPIP
2 mai 2011, parThis page lists some websites based on MediaSPIP.
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Creating farms of unique websites
13 avril 2011, parMediaSPIP platforms can be installed as a farm, with a single "core" hosted on a dedicated server and used by multiple websites.
This allows (among other things) : implementation costs to be shared between several different projects / individuals rapid deployment of multiple unique sites creation of groups of like-minded sites, making it possible to browse media in a more controlled and selective environment than the major "open" (...)
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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.mp3till 2:05 seconds ofb.mp3and append to the end ofa.mp3(its 3:23 long).Command :
ffmpeg -i a.mp3 -i b.mp3Output :
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:':
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