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999,999
26 septembre 2011, par
Mis à jour : Septembre 2011
Langue : English
Type : Audio
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The Slip - Artworks
26 septembre 2011, par
Mis à jour : Septembre 2011
Langue : English
Type : Texte
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Demon seed (wav version)
26 septembre 2011, par
Mis à jour : Avril 2013
Langue : English
Type : Audio
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The four of us are dying (wav version)
26 septembre 2011, par
Mis à jour : Avril 2013
Langue : English
Type : Audio
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Corona radiata (wav version)
26 septembre 2011, par
Mis à jour : Avril 2013
Langue : English
Type : Audio
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Lights in the sky (wav version)
26 septembre 2011, par
Mis à jour : Avril 2013
Langue : English
Type : Audio
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Sur d’autres sites (9244)
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Convert Process Output to Progressbar C# [on hold]
28 mars 2018, par adrifcastrSo I am using FFMPEG to convert a audio file, and I need to get a progressbar to show the current conversion process. So as reading through Stackoverflow I found this, which uses a Streamreader to get the complete duration of the file and the current progress into a string. This didn’t work for me, I assumed it’s because I am using WPF. So can someone help me to get those values and then use them to calculate the percentage and show it with a Progressbar ?
Below are samples of what’s displayed in FFMPEG’s output stream.
//Complete Duration of the Audio file being converted
Duration: 06:41:03.68, start: 0.000000, bitrate: 64 kb/s
//Current conversion progress
size= 1197kB time=00:02:02.14 bitrate= 80.3kbits/s speed=42.9x -
Audacity vocal removal failed when ffmpeg-conversion was involved
10 mars 2018, par fyangI downloaded some songs coded with FLAC, and Audacity could remove the vocals quite well.
When I downloaded songs coded with ALAC, I must use ffmpeg to convert them to some other forms because Audacity didn’t recognise .m4a files.
I used the command
ffmpeg -i "song 01.m4a" -f flac "song 01.flac"
. Now Audacity could load the song, but its vocal removal failed to remove the vocals.I tried again with this command in order to be precise,
ffmpeg -i "song 01.m4a" -af "pan=stereo|c0=c0|c1=c1" -f flac "song 01.flac"
, and vocal removal did not work either.I tried to do it manually by splitting, inverting and changing both channels to mono, but the vocals were still there.
I think the problem lies with the ffmpeg conversion step. Is there any fix ? Thanks !
Below is the result of the conversion :
ffmpeg -i "song 01.m4a" -af "pan=stereo|c0=c0|c1=c1" -f flac "song 01.flac"
ffmpeg version N-90143-gb6652f5100 Copyright (c) 2000-2018 the FFmpeg developers
built with gcc 7.3.0 (GCC)
configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-bzlib --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --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-libmfx --enable-amf --enable-cuda --enable-cuvid --enable-d3d11va --enable-nvenc --enable-dxva2 --enable-avisynth
libavutil 56. 7.101 / 56. 7.101
libavcodec 58. 12.102 / 58. 12.102
libavformat 58. 9.100 / 58. 9.100
libavdevice 58. 2.100 / 58. 2.100
libavfilter 7. 12.100 / 7. 12.100
libswscale 5. 0.101 / 5. 0.101
libswresample 3. 0.101 / 3. 0.101
libpostproc 55. 0.100 / 55. 0.100
[mov,mp4,m4a,3gp,3g2,mj2 @ 0000019f2b258000] stream 0, timescale not set
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'song 01.m4a':
Metadata:
major_brand : M4A
minor_version : 0
compatible_brands: M4A mp42isom
creation_time : 2009-12-27T00:15:23.000000Z
track : 1/10
genre :
album :
artist :
comment : ExactAudioCopy v0.95b4
DISCID :
iTunNORM : 00000F32 00000E1D 0000547D 00005B93 0006C3CA 0006C43E 00007FF8 00007FFF 00058227 0003593B
title : song 01
encoder : iTunes 9.0.2.25
date : 2005
album_artist :
lyrics :
Duration: 00:08:10.84, start: 0.000000, bitrate: 921 kb/s
Stream #0:0(und): Audio: alac (alac / 0x63616C61), 44100 Hz, stereo, s16p, 920 kb/s (default)
Metadata:
creation_time : 2009-12-27T00:15:23.000000Z
Stream #0:1: Video: mjpeg, yuvj444p(pc, bt470bg/unknown/unknown), 300x300 [SAR 100:100 DAR 1:1], 90k tbr, 90k tbn, 90k tbc
Stream mapping:
Stream #0:0 -> #0:0 (alac (native) -> flac (native))
Press [q] to stop, [?] for help
[Parsed_pan_0 @ 0000019f2b2a6fc0] Pure channel mapping detected: 0 1
Output #0, flac, to 'song 01.flac':
Metadata:
major_brand : M4A
minor_version : 0
compatible_brands: M4A mp42isom
lyrics :
TRACKNUMBER : 1/10
genre :
album :
artist :
DESCRIPTION : ExactAudioCopy v0.95b4
DISCID :
iTunNORM : 00000F32 00000E1D 0000547D 00005B93 0006C3CA 0006C43E 00007FF8 00007FFF 00058227 0003593B
title : song 01
ALBUMARTIST :
date : 2005
encoder : Lavf58.9.100
Stream #0:0(und): Audio: flac, 44100 Hz, stereo, s16, 128 kb/s (default)
Metadata:
creation_time : 2009-12-27T00:15:23.000000Z
encoder : Lavc58.12.102 flac
size= 54518kB time=00:08:10.84 bitrate= 909.9kbits/s speed= 35x
video:0kB audio:54508kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.018294% -
Convert audio files to mp3 using ffmpeg [closed]
5 août, par Hrishikesh -Rishi- ChoudhariI need to convert audio files to mp3 using ffmpeg.



When I write the command as
ffmpeg -i audio.ogg -acodec mp3 newfile.mp3
, I get the error :


FFmpeg version 0.5.2, Copyright (c) 2000-2009 Fabrice Bellard, et al.
 configuration: 
 libavutil 49.15. 0 / 49.15. 0
 libavcodec 52.20. 1 / 52.20. 1
 libavformat 52.31. 0 / 52.31. 0
 libavdevice 52. 1. 0 / 52. 1. 0
 built on Jun 24 2010 14:56:20, gcc: 4.4.1
Input #0, mp3, from 'ZHRE.mp3':
 Duration: 00:04:12.52, start: 0.000000, bitrate: 208 kb/s
 Stream #0.0: Audio: mp3, 44100 Hz, stereo, s16, 256 kb/s
Output #0, mp3, to 'audio.mp3':
 Stream #0.0: Audio: 0x0000, 44100 Hz, stereo, s16, 64 kb/s
Stream mapping:
 Stream #0.0 -> #0.0
Unsupported codec for output stream #0.0




I also ran this command :



ffmpeg -formats | grep mp3




and got this in response :



FFmpeg version 0.5.2, Copyright (c) 2000-2009 Fabrice Bellard, et al.
 configuration: 
 libavutil 49.15. 0 / 49.15. 0
 libavcodec 52.20. 1 / 52.20. 1
 libavformat 52.31. 0 / 52.31. 0
 libavdevice 52. 1. 0 / 52. 1. 0
 built on Jun 24 2010 14:56:20, gcc: 4.4.1
 DE mp3 MPEG audio layer 3
 D A mp3 MP3 (MPEG audio layer 3)
 D A mp3adu ADU (Application Data Unit) MP3 (MPEG audio layer 3)
 D A mp3on4 MP3onMP4
 text2movsub remove_extra noise mov2textsub mp3decomp mp3comp mjpegadump imxdump h264_mp4toannexb dump_extra




I guess that the mp3 codec isn't installed. Am I on the right track here ?