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Autres articles (32)
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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 (...) -
De l’upload à la vidéo finale [version standalone]
31 janvier 2010, parLe chemin d’un document audio ou vidéo dans SPIPMotion est divisé en trois étapes distinctes.
Upload et récupération d’informations de la vidéo source
Dans un premier temps, il est nécessaire de créer un article SPIP et de lui joindre le document vidéo "source".
Au moment où ce document est joint à l’article, deux actions supplémentaires au comportement normal sont exécutées : La récupération des informations techniques des flux audio et video du fichier ; La génération d’une vignette : extraction d’une (...) -
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 (4367)
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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:': -
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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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