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Head down (wav version)
26 septembre 2011, par
Mis à jour : Avril 2013
Langue : English
Type : Audio
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Echoplex (wav version)
26 septembre 2011, par
Mis à jour : Avril 2013
Langue : English
Type : Audio
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Discipline (wav version)
26 septembre 2011, par
Mis à jour : Avril 2013
Langue : English
Type : Audio
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Letting you (wav version)
26 septembre 2011, par
Mis à jour : Avril 2013
Langue : English
Type : Audio
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1 000 000 (wav version)
26 septembre 2011, par
Mis à jour : Avril 2013
Langue : English
Type : Audio
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26 septembre 2011, par
Mis à jour : Avril 2013
Langue : English
Type : Audio
Autres articles (66)
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Encoding and processing into web-friendly formats
13 avril 2011, parMediaSPIP automatically converts uploaded files to internet-compatible formats.
Video files are encoded in MP4, Ogv and WebM (supported by HTML5) and MP4 (supported by Flash).
Audio files are encoded in MP3 and Ogg (supported by HTML5) and MP3 (supported by Flash).
Where possible, text is analyzed in order to retrieve the data needed for search engine detection, and then exported as a series of image files.
All uploaded files are stored online in their original format, so you can (...) -
L’utiliser, en parler, le critiquer
10 avril 2011La première attitude à adopter est d’en parler, soit directement avec les personnes impliquées dans son développement, soit autour de vous pour convaincre de nouvelles personnes à l’utiliser.
Plus la communauté sera nombreuse et plus les évolutions seront rapides ...
Une liste de discussion est disponible pour tout échange entre utilisateurs. -
Le profil des utilisateurs
12 avril 2011, parChaque utilisateur dispose d’une page de profil lui permettant de modifier ses informations personnelle. Dans le menu de haut de page par défaut, un élément de menu est automatiquement créé à l’initialisation de MediaSPIP, visible uniquement si le visiteur est identifié sur le site.
L’utilisateur a accès à la modification de profil depuis sa page auteur, un lien dans la navigation "Modifier votre profil" est (...)
Sur d’autres sites (5003)
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Use fluent-ffmpeg to tell if a file is a video or audio
8 mai 2021, par afterglowleeI am using node-fluent-ffmpeg module in NodeJS. It is very good that fluent-ffmpeg provides functions to get the metadata of a video and audio file.



https://github.com/schaermu/node-fluent-ffmpeg#reading-video-metadata



I have tried on Mac OS to use the "resolution" attribute in the metadata to tell if a file is audio only or video, i.e. if both resolution.w and resolution.h are 0, then this file is an audio. This work fine on Mac OS. But some strange things happened that this doesn't work on Windows platform (I have tried Windows 7 64bit and Windows 2008) using the latest ffmpeg. Even though I put a .mp3 file through fluent-ffmpeg,the result looks something like this :



video:
{
 container:'mp3',
 ...
 resolution: {w:300,h:300},
 resolutionSquare: {w:300,h:300},
 aspectString: '1:1',
 ...
}
audio:
{
 codec:'mp3',
 bitrate:64,
 sample_rate:44100,
 stream:0,
 channels:1
}




I am not why there is a "resolution" since it is a pure audio file. So is there any solid way to find out if the file is audio only or video from the metadata ? Or should I use ffmpeg commandline to find it out ?


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Create HLS streamable audio file from mp3
15 août 2023, par isADonI am using following command to create a hls aac audio file for web streaming



ffmpeg -y -i song.mp3 -c:a aac -b:a 128k -f hls -hls_time 7 -hls_list_size 0 -hls_segment_filename file%d.m4a playlist.m3u8




This command works only with some audio files. With many mp3 files I receive following output :



C:\ffmpeg>ffmpeg -y -i song.mp3 -c:a aac -b:a 128k -f hls -hls_time 7 -hls_list_size 0 -hls_segment_filename file%d.m4a playlist.m3u8
ffmpeg version git-2020-01-31-62d92a8 Copyright (c) 2000-2020 the FFmpeg developers
 built with gcc 9.2.1 (GCC) 20200122
 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-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt --enable-amf
 libavutil 56. 38.100 / 56. 38.100
 libavcodec 58. 67.100 / 58. 67.100
 libavformat 58. 37.100 / 58. 37.100
 libavdevice 58. 9.103 / 58. 9.103
 libavfilter 7. 72.100 / 7. 72.100
 libswscale 5. 6.100 / 5. 6.100
 libswresample 3. 6.100 / 3. 6.100
 libpostproc 55. 6.100 / 55. 6.100
[mp3 @ 0000027d800babc0] Estimating duration from bitrate, this may be inaccurate
Input #0, mp3, from 'song.mp3':
 Metadata:
 TSS : Logic Pro 8.0.2
 iTunNORM : 000000EE 000000ED 00000C34 00001135 000088F0 0000B505 000080FA 00007577 00009B82 00018F49
 iTunSMPB : 00000000 00000210 00000A07 00000000008783E9 00000000 007AD4E6 00000000 00000000 00000000 00000000 00000000 00000000
 genre : Rock
 TCM : Kevin MacLeod
 album : Funk and Blues
 TKE : C
 TBP : 101
 title : Funkorama
 artist : Kevin MacLeod
 date : 2008-06-16 18:35
 Duration: 00:03:21.46, start: 0.000000, bitrate: 325 kb/s
 Stream #0:0: Audio: mp3, 44100 Hz, stereo, fltp, 320 kb/s
 Stream #0:1: Video: mjpeg (Baseline), yuvj444p(pc, bt470bg/unknown/unknown), 400x400 [SAR 72:72 DAR 1:1], 90k tbr, 90k tbn, 90k tbc (attached pic)
 Metadata:
 comment : Other
Stream mapping:
 Stream #0:1 -> #0:0 (mjpeg (native) -> h264 (libx264))
 Stream #0:0 -> #0:1 (mp3 (mp3float) -> aac (native))
Press [q] to stop, [?] for help
[hls @ 0000027d80100c40] Frame rate very high for a muxer not efficiently supporting it.
Please consider specifying a lower framerate, a different muxer or -vsync 2
[libx264 @ 0000027d800c1280] using SAR=1/1
[libx264 @ 0000027d800c1280] MB rate (56250000) > level limit (16711680)
[libx264 @ 0000027d800c1280] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX FMA3 BMI2 AVX2
[libx264 @ 0000027d800c1280] profile High 4:4:4 Predictive, level 6.2, 4:4:4, 8-bit
[libx264 @ 0000027d800c1280] 264 - core 159 - H.264/MPEG-4 AVC codec - Copyleft 2003-2019 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=4 threads=12 lookahead_threads=2 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
Output #0, hls, to 'playlist.m3u8':
 Metadata:
 TSS : Logic Pro 8.0.2
 iTunNORM : 000000EE 000000ED 00000C34 00001135 000088F0 0000B505 000080FA 00007577 00009B82 00018F49
 iTunSMPB : 00000000 00000210 00000A07 00000000008783E9 00000000 007AD4E6 00000000 00000000 00000000 00000000 00000000 00000000
 genre : Rock
 TCM : Kevin MacLeod
 album : Funk and Blues
 TKE : C
 TBP : 101
 title : Funkorama
 artist : Kevin MacLeod
 date : 2008-06-16 18:35
 encoder : Lavf58.37.100
 Stream #0:0: Video: h264 (libx264), yuvj444p(pc, progressive), 400x400 [SAR 72:72 DAR 1:1], q=-1--1, 90k fps, 90k tbn, 90k tbc (attached pic)
 Metadata:
 comment : Other
 encoder : Lavc58.67.100 libx264
 Side data:
 cpb: bitrate max/min/avg: 0/0/0 buffer size: 0 vbv_delay: N/A
 Stream #0:1: Audio: aac (LC), 44100 Hz, stereo, fltp, 128 kb/s
 Metadata:
 encoder : Lavc58.67.100 aac
[mp3float @ 0000027d80146580] overread, skip -7 enddists: -6 -6 speed=68.6x
[mp3float @ 0000027d80146580] overread, skip -6 enddists: -5 -5
[mp3float @ 0000027d80146580] overread, skip -6 enddists: -4 -4
 Last message repeated 2 times
[mp3float @ 0000027d80146580] overread, skip -7 enddists: -6 -6
 Last message repeated 2 times
[mp3float @ 0000027d80146580] overread, skip -5 enddists: -2 -2
[mp3float @ 0000027d80146580] overread, skip -7 enddists: -6 -6
[mp3float @ 0000027d80146580] overread, skip -6 enddists: -4 -4
 Last message repeated 1 times
[mp3float @ 0000027d80146580] overread, skip -7 enddists: -6 -6
 Last message repeated 1 times
[mp3float @ 0000027d80146580] overread, skip -6 enddists: -4 -4
[mp3float @ 0000027d80146580] overread, skip -5 enddists: -3 -3
[mp3float @ 0000027d80146580] overread, skip -6 enddists: -4 -4
[mp3float @ 0000027d80146580] overread, skip -7 enddists: -6 -6
 Last message repeated 2 times
[mp3float @ 0000027d80146580] overread, skip -5 enddists: -4 -4
[hls @ 0000027d80100c40] Opening 'file0.m4a' for writingate=N/A speed=64.1x
[hls @ 0000027d80100c40] Opening 'playlist.m3u8.tmp' for writing
frame= 1 fps=0.3 q=33.0 Lsize=N/A time=00:03:21.45 bitrate=N/A speed=63.7x
video:7kB audio:3209kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown
[libx264 @ 0000027d800c1280] frame I:1 Avg QP:34.64 size: 6567
[libx264 @ 0000027d800c1280] mb I I16..4: 19.5% 53.0% 27.5%
[libx264 @ 0000027d800c1280] 8x8 transform intra:53.0%
[libx264 @ 0000027d800c1280] coded y,u,v intra: 46.8% 26.1% 15.3%
[libx264 @ 0000027d800c1280] i16 v,h,dc,p: 38% 39% 9% 14%
[libx264 @ 0000027d800c1280] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 21% 14% 26% 8% 5% 6% 5% 7% 7%
[libx264 @ 0000027d800c1280] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 42% 16% 14% 7% 4% 5% 3% 4% 4%
[libx264 @ 0000027d800c1280] kb/s:4728240.00
[aac @ 0000027d800bcc40] Qavg: 2138.508




Notice the "mp3float overread" message.



It results in a single
file0.m4a
file without splitting it up after every 7 seconds as specified.
This is an example audio file I am trying to convert to a aac hls stream that results the mentioned problem : https://incompetech.com/music/royalty-free/index.html?isrc=USUAN1100474


How can I convert an audio file to a web friendly hls stream with ffmpeg ?


-
Animated line chart with pandas, matplotlib and ffmpeg
10 avril 2020, par Mark KIn producing an animated line chart, I have below data and codes.



But when the chart produced, it shows no line. What did I do wrong ?



Thank you.



import numpy as np
import pandas as pd
import seaborn as sns
import matplotlib
import matplotlib.pyplot as plt
import matplotlib.animation as animation

title = 'Heroin Overdoses'

data = {'Year' : ["1999","2000","2001","2002","2003","2004","2005","2006","2007","2008","2009","2010","2011","2012","2013","2014","2015","2016"], 
'Heroin Overdoses' : [280,443,413,486,475,148,197,170,448,103,137,160,483,356,352,300,466,278]}
overdose = pd.DataFrame(data)

Writer = animation.writers['ffmpeg']
writer = Writer(fps=20, metadata=dict(artist='Me'), bitrate=1800)

fig = plt.figure(figsize=(10,6))
plt.xlim(1999, 2016)
plt.ylim(np.min(overdose)[0], np.max(overdose)[0])
plt.xlabel('Year',fontsize=20)
plt.ylabel(title,fontsize=20)
plt.title('Heroin Overdoses per Year',fontsize=20)

def animate(i):
 data = overdose.iloc[:int(i+1)] #select data range
 p = sns.lineplot(x=data.index, y=data[title], data=data, color="r")
 p.tick_params(labelsize=17)
 plt.setp(p.lines,linewidth=7)

ani = matplotlib.animation.FuncAnimation(fig, animate, frames=17, repeat=True)

ani.save('C:\\folder\\line chart.mp4', writer=writer)