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Autres articles (38)
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Contribute to documentation
13 avril 2011Documentation is vital to the development of improved technical capabilities.
MediaSPIP welcomes documentation by users as well as developers - including : critique of existing features and functions articles contributed by developers, administrators, content producers and editors screenshots to illustrate the above translations of existing documentation into other languages
To contribute, register to the project users’ mailing (...) -
Selection of projects using MediaSPIP
2 mai 2011, parThe examples below are representative elements of MediaSPIP specific uses for specific projects.
MediaSPIP farm @ Infini
The non profit organizationInfini develops hospitality activities, internet access point, training, realizing innovative projects in the field of information and communication technologies and Communication, and hosting of websites. It plays a unique and prominent role in the Brest (France) area, at the national level, among the half-dozen such association. Its members (...) -
Use, discuss, criticize
13 avril 2011, parTalk to people directly involved in MediaSPIP’s development, or to people around you who could use MediaSPIP to share, enhance or develop their creative projects.
The bigger the community, the more MediaSPIP’s potential will be explored and the faster the software will evolve.
A discussion list is available for all exchanges between users.
Sur d’autres sites (8271)
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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% -
Parse and drop gain control data, so that SSR packets decode.
16 février 2018, par Dale CurtisParse and drop gain control data, so that SSR packets decode.
This will result in poor quality audio for SSR streams, but they
will at least demux and decode without error ; partially fixing
ticket #1693.This pulls in the decode_gain_control() function from the
ffmpeg summer-of-code repo (original author Maxim Gavrilov) at
svn ://svn.mplayerhq.hu/soc/aac/aac.c with some minor modifications
and adds AOT_AAC_SSR to decode_audio_specific_config_gb().Signed-off-by : Dale Curtis <dalecurtis@chromium.org>
Co-authored-by : Maxim Gavrilov <maxim.gavrilov@gmail.com>