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Python Audio Tools 2.2
Python Audio Tools are a collection of audio handling programs which work from the command line. more>>
Python Audio Tools are a collection of audio handling programs which work from the command line. These include programs for CD extraction, track conversion from one audio format to another, track renaming and retagging, track identification, CD burning from tracks, and more.
Supports internationalized track filenames and metadata using Unicode. Works with high-definition, multi-channel audio as well as CD-quality. Track conversion uses multiple CPUs or CPU cores if available to greatly speed the transcoding process. Track metadata can be retrieved from FreeDB or FreeDB-compatible servers. Audio formats supported are:
- WAV
- AIFF
- Sun AU
- FLAC
- WavPack
- Monkeys Audio
- MP3
- MP2
- Ogg Vorbis
- Ogg Speex
- M4A
Easy to install and use:
I dont like programs with lots of obscure dependencies, so Ive written Python Audio Tools to require as few external libraries and programs as possible. Most formats require only one or two common executables and some require none at all. Installation utilizes Pythons built-in distutils, but a provided makefile reduces the process to a simple "make install" command.
The programs that make up Python Audio Tools provide a consistent interface for painless switching between them. man pages are included, as is a lengthy manual for both users and programmers. See the short tutorial in the documentation for an illustration of their usage.
These Python Audio Tools are designed with Unix-like systems in mind. Although most of it is written in pure Python, the libcdio library is used to read from a CD-ROM drive. If one is willing to forgo the CD-based tools, Python Audio Tools is fully portable so long as the required helper applications are present. I have used it on the i386, x86_64 and ppc architectures without difficulty.
Standards compliant
Because Python Audio Tools sticks very close to established, documented standards, it has no trouble interoperating with other audio-handling programs. If you want to rip CDs with Exact Audio Copy, batch convert them to M4As with these tools and play them back in iTunes, thats no problem.
Enhancements:
- Revamped the track metadata system.
- Improved the file detection routines.
- Minor bugfixes.
<<lessSupports internationalized track filenames and metadata using Unicode. Works with high-definition, multi-channel audio as well as CD-quality. Track conversion uses multiple CPUs or CPU cores if available to greatly speed the transcoding process. Track metadata can be retrieved from FreeDB or FreeDB-compatible servers. Audio formats supported are:
- WAV
- AIFF
- Sun AU
- FLAC
- WavPack
- Monkeys Audio
- MP3
- MP2
- Ogg Vorbis
- Ogg Speex
- M4A
Easy to install and use:
I dont like programs with lots of obscure dependencies, so Ive written Python Audio Tools to require as few external libraries and programs as possible. Most formats require only one or two common executables and some require none at all. Installation utilizes Pythons built-in distutils, but a provided makefile reduces the process to a simple "make install" command.
The programs that make up Python Audio Tools provide a consistent interface for painless switching between them. man pages are included, as is a lengthy manual for both users and programmers. See the short tutorial in the documentation for an illustration of their usage.
These Python Audio Tools are designed with Unix-like systems in mind. Although most of it is written in pure Python, the libcdio library is used to read from a CD-ROM drive. If one is willing to forgo the CD-based tools, Python Audio Tools is fully portable so long as the required helper applications are present. I have used it on the i386, x86_64 and ppc architectures without difficulty.
Standards compliant
Because Python Audio Tools sticks very close to established, documented standards, it has no trouble interoperating with other audio-handling programs. If you want to rip CDs with Exact Audio Copy, batch convert them to M4As with these tools and play them back in iTunes, thats no problem.
Enhancements:
- Revamped the track metadata system.
- Improved the file detection routines.
- Minor bugfixes.
Download (0.52MB)
Added: 2007-08-03 License: GPL (GNU General Public License) Price:
817 downloads
Fast Audio Playlist Generator 0.31
FAPG is a tool to generate list of audio files (WAV, MP3, Ogg, etc.) in various formats (M3U, PLS, HTML, etc). more>>
FAPG is a tool to generate list of audio files (WAV, MP3, Ogg, etc.) in various formats (M3U, PLS, HTML, etc). It is very useful for quickly creating a playlist of a large number of audio files.
It is coded in C to be as fast as possible, and does not use any specific audio library (like ID3Lib).
This allow you to deploy it faster and easier, and to have better performances since the less informations are loaded.
In the other hand, this tool is not (yet) compatible with all the known formats.
Here are some examples for usage :
Generate a PLS playlist for an album :
fapg --format=pls --output=~/path/to/album/list.pls ~/path/to/album
You have a file server using Samba which shares a large amount of audio files for Windows powered computers using Winamp.
The directory on the server containing the files is "/samba/mp3" and is visible on the Windows network as "serveurmp3" :
fapg --backslash --output=/samba/mp3/list.m3u --prefix=servermp3 --recursive --windows /samba/mp3
An HTML playlist for an album :
fapg --output=fapg.html ~/path/to/album
Here is the result : fapg.html
<<lessIt is coded in C to be as fast as possible, and does not use any specific audio library (like ID3Lib).
This allow you to deploy it faster and easier, and to have better performances since the less informations are loaded.
In the other hand, this tool is not (yet) compatible with all the known formats.
Here are some examples for usage :
Generate a PLS playlist for an album :
fapg --format=pls --output=~/path/to/album/list.pls ~/path/to/album
You have a file server using Samba which shares a large amount of audio files for Windows powered computers using Winamp.
The directory on the server containing the files is "/samba/mp3" and is visible on the Windows network as "serveurmp3" :
fapg --backslash --output=/samba/mp3/list.m3u --prefix=servermp3 --recursive --windows /samba/mp3
An HTML playlist for an album :
fapg --output=fapg.html ~/path/to/album
Here is the result : fapg.html
Download (0.013MB)
Added: 2006-07-20 License: GPL (GNU General Public License) Price:
1194 downloads
Perl Audio Converter 3.3.2
Perl Audio Converter (PAC) is a simple script for converting multiple audio types from one format to another. more>>
Perl Audio Converter (PAC) is a tool for converting multiple audio types from one format to another. It supports MP2, MP3, Ogg Vorbis, FLAC, Shorten, Monkey Audio, FAAC (AAC/M4A/MP4), Musepack (MPC), Wavpack (WV), OptimFrog (OFR/OFS), TTA, LPAC, Kexis (KXS), AIFF, AC3, Lossless Audio (LA), AU, SND, RAW, VOC, SMP, RealAudio (RA/RAM), WAV, and WMA.
It can also convert audio from the following video formats/extensions: RM, RV, ASF, DivX, MPG, MKV, MPEG, AVI, MOV, OGM, QT, VCD, VOB, and WMV. A CD ripping function with CDDB support, batch and playlist conversion, tag preservation for most supported formats, independent tag reading/writing, and extensions for Konqueror and amaroK is also provided.
Enhancements:
- Added: MP4::Info dependency IO::String to pacpl-install
- Updated: License GPLv3
- Bug Fix: Directory conversions. pacpl was claiming the directory was empty and prompting to use the --recursive option...Thanks to John Meyer for the report.
<<lessIt can also convert audio from the following video formats/extensions: RM, RV, ASF, DivX, MPG, MKV, MPEG, AVI, MOV, OGM, QT, VCD, VOB, and WMV. A CD ripping function with CDDB support, batch and playlist conversion, tag preservation for most supported formats, independent tag reading/writing, and extensions for Konqueror and amaroK is also provided.
Enhancements:
- Added: MP4::Info dependency IO::String to pacpl-install
- Updated: License GPLv3
- Bug Fix: Directory conversions. pacpl was claiming the directory was empty and prompting to use the --recursive option...Thanks to John Meyer for the report.
Download (0.12MB)
Added: 2007-07-07 License: GPL v3 Price:
526 downloads
AudConvert 0.52
AudConvert is an application that is designed to take any audio format and convert it to any other audio format. more>>
AudConvert is an application that is designed to take any audio format and convert it to any other audio format.
The idea for AudConvert came from my need to turn my Ogg Vorbis collection into MP3s for portable devices.
Yes, this process sometimes will result in lower quality, but sometimes it must be done.
Main features:
- Input any directory of files, get out the same directory structure (or flat directory) of newly encoded files.
- Multi-threaded: Encode up to 8 files simultaneously.
This is the first release of this software and it needs a lot of testing.
Supported Inputs:
- Ogg Vorbis (oggdec)
- MP3 (mpg123)
- FLAC (flac)
Supported Outputs:
- Ogg Vorbis (oggenc)
- MP3 (lame)
<<lessThe idea for AudConvert came from my need to turn my Ogg Vorbis collection into MP3s for portable devices.
Yes, this process sometimes will result in lower quality, but sometimes it must be done.
Main features:
- Input any directory of files, get out the same directory structure (or flat directory) of newly encoded files.
- Multi-threaded: Encode up to 8 files simultaneously.
This is the first release of this software and it needs a lot of testing.
Supported Inputs:
- Ogg Vorbis (oggdec)
- MP3 (mpg123)
- FLAC (flac)
Supported Outputs:
- Ogg Vorbis (oggenc)
- MP3 (lame)
Download (0.022MB)
Added: 2006-03-11 License: GPL (GNU General Public License) Price:
1322 downloads
daudio 0.3
Daudio is a set of programs to have fully syncronized, distributed mp3 audio over a local network. more>>
Daudio is a set of programs to have fully syncronized, distributed mp3 audio over a local network. Now you can have all your computers playing mp3s at once!
One must first install a kernel patched with the lowlatency and utime patches, located in lowlat-utime/
This is necessary for the server to have resonable timing resolution.
There are three components to the daudio system: the client, the server, and the provider (streamer). The client is used to play incoming data from any server. The provider sends mp3 data to the server, and the server times and sends out the mp3 data.
To compile and install these programs, cd to the appropriate directory, edit the Makefile (or accept the defaults), and type "make" followed by "make install".
To start the server, simply type "daudiod" at the command prompt.The server accepts no arguments.
To start the client, type "daudioc hostname".
To start streaming data to the local server, type "dstreamer [-z] file.mp3 [file2.mp3 ...]". Where "-z" randomizes the songs. dstreamer is meant to function like a very simple mpg123.
By default, daudioc & daudiod are installed without SUID. If youre brave, you may get slightly better performance by setting them SUID to root. This will cause the critical timing threads to get realtime priority. This is highly dangerous, as a thread with realtime
priority will not be preempted and must yield. If there are any bugs in the code, they could be exploited in a DOS attack.
Enhancements:
- client/Makefile, doc/COPYING, doc/README, lowlat-utime/README, server/Makefile, server/bcast.c, server/data.c, server/local.c, server/main.c, server/timer.c, streamer/streamer.c: Put somefinishing touches on documentation and stuff.
<<lessOne must first install a kernel patched with the lowlatency and utime patches, located in lowlat-utime/
This is necessary for the server to have resonable timing resolution.
There are three components to the daudio system: the client, the server, and the provider (streamer). The client is used to play incoming data from any server. The provider sends mp3 data to the server, and the server times and sends out the mp3 data.
To compile and install these programs, cd to the appropriate directory, edit the Makefile (or accept the defaults), and type "make" followed by "make install".
To start the server, simply type "daudiod" at the command prompt.The server accepts no arguments.
To start the client, type "daudioc hostname".
To start streaming data to the local server, type "dstreamer [-z] file.mp3 [file2.mp3 ...]". Where "-z" randomizes the songs. dstreamer is meant to function like a very simple mpg123.
By default, daudioc & daudiod are installed without SUID. If youre brave, you may get slightly better performance by setting them SUID to root. This will cause the critical timing threads to get realtime priority. This is highly dangerous, as a thread with realtime
priority will not be preempted and must yield. If there are any bugs in the code, they could be exploited in a DOS attack.
Enhancements:
- client/Makefile, doc/COPYING, doc/README, lowlat-utime/README, server/Makefile, server/bcast.c, server/data.c, server/local.c, server/main.c, server/timer.c, streamer/streamer.c: Put somefinishing touches on documentation and stuff.
Download (0.074MB)
Added: 2006-07-26 License: GPL (GNU General Public License) Price:
1186 downloads
Make audio 0.4.1
Make audio is a graphical tool to create audio CDs from MP3 or wav files. more>>
Make audio is a graphical tool to create audio CDs from MP3 or wav files. You can simply arrange a playlist of all songs that you want. The application converts MP3 files to WAV and then burns them on the CD.
<<less Download (0.016MB)
Added: 2006-07-25 License: GPL (GNU General Public License) Price:
1186 downloads
Audio Convert 0.3.1.1
audio convert is a script to convert Wav, Ogg, MP3, MPC, FLAC, APE, or WMA files into Wav, Ogg, MP3, MPC, FLAC, or APE files. more>>
Audio Convert is a script to convert Wav, Ogg, MP3, MPC, FLAC, APE, or WMA files into Wav, Ogg, MP3, MPC, FLAC, or APE files.
It has an easy-to-use interface that makes it possible to fill in the tags for a few formats and choose the quality of compression.
The script was initially designed for the nautilus file browser, and can be easily installed into nautilus by copying it to the nautilus-scripts directory, at which point you can right click on the desired audio file(s) and choose "audio-convert" from the "scripts" menu.
The script is also known to work on rox or directly from a shell.
Enhancements:
- The installer was modified to create the directory it is supposed to create.
<<lessIt has an easy-to-use interface that makes it possible to fill in the tags for a few formats and choose the quality of compression.
The script was initially designed for the nautilus file browser, and can be easily installed into nautilus by copying it to the nautilus-scripts directory, at which point you can right click on the desired audio file(s) and choose "audio-convert" from the "scripts" menu.
The script is also known to work on rox or directly from a shell.
Enhancements:
- The installer was modified to create the directory it is supposed to create.
Download (0.010MB)
Added: 2005-11-12 License: GPL (GNU General Public License) Price:
1446 downloads
MPEG::Audio::Frame 0.09
MPEG::Audio::Frame is a class for weeding out MPEG audio frames out of a file handle. more>>
MPEG::Audio::Frame is a class for weeding out MPEG audio frames out of a file handle.
SYNOPSIS
use MPEG::Audio::Frame;
open FILE,"file.mp3";
while(my $frame = MPEG::Audio::Frame->read(*FILE)){
print $frame->offset(), ": ", $frame->bitrate(), "Kbps/", $frame->sample()/1000, "KHzn"; # or something.
}
A very simple, pure Perl module which allows parsing out data from mp3 files, or streams, and chunking them up into different frames. You can use this to accurately determine the length of an mp3, filter nonaudio data, or chunk up the file for streaming via datagram. Virtually anything is possible.
<<lessSYNOPSIS
use MPEG::Audio::Frame;
open FILE,"file.mp3";
while(my $frame = MPEG::Audio::Frame->read(*FILE)){
print $frame->offset(), ": ", $frame->bitrate(), "Kbps/", $frame->sample()/1000, "KHzn"; # or something.
}
A very simple, pure Perl module which allows parsing out data from mp3 files, or streams, and chunking them up into different frames. You can use this to accurately determine the length of an mp3, filter nonaudio data, or chunk up the file for streaming via datagram. Virtually anything is possible.
Download (0.35MB)
Added: 2006-06-19 License: Perl Artistic License Price:
1222 downloads
Audio::TagLib::MPEG::File 1.42
Audio::TagLib::MPEG::File is an MPEG file class with some useful methods specific to MPEG. more>>
Audio::TagLib::MPEG::File is an MPEG file class with some useful methods specific to MPEG.
SYNOPSIS
use Audio::TagLib::MPEG::File;
my $i = Audio::TagLib::MPEG::File->new("sample file.mp3");
print $i->tag()->artist()->toCString(), "n"; # got artist
This implements the generic Audio::TagLib::File API and additionally provides access to properties that are distinct to MPEG files, notably access to the different ID3 tags.
<<lessSYNOPSIS
use Audio::TagLib::MPEG::File;
my $i = Audio::TagLib::MPEG::File->new("sample file.mp3");
print $i->tag()->artist()->toCString(), "n"; # got artist
This implements the generic Audio::TagLib::File API and additionally provides access to properties that are distinct to MPEG files, notably access to the different ID3 tags.
Download (1.4MB)
Added: 2006-11-16 License: Perl Artistic License Price:
1072 downloads
Audio::ConvTools 0.08
Audio::ConvTools is a API to convert audio files from/to mp3 ogg and wav. more>>
Audio::ConvTools is a API to convert audio files from/to mp3 ogg and wav.
SYNOPSIS
use Audio::ConvTools;
use Audio::ConvTools qw/:DEFAULT :Tmp :Log/;
$status = ogg2mp3(file.ogg);
$status = ogg2wav(file.ogg);
$status = ogg2wav(in.ogg, out.wav);
$status = mp32ogg(file.mp3);
$status = mp32wav(file.mp3);
$status = mp32wav(in.mp3, out.wav);
$status = wav2ogg(file.wav);
$status = wav2ogg(in.wav, out.ogg);
$status = wav2mp3(file.wav);
$status = wav2mp3(in.wav, out.mp3);
Audio::ConvTools::logMsg(This is a log message);
Audio::ConvTools::errMsg(This is an error message);
$tmp = Audio::ConvTools::getTmpFile(.wav);
Audio::ConvTools::destroyTmpFile($tmp);
<<lessSYNOPSIS
use Audio::ConvTools;
use Audio::ConvTools qw/:DEFAULT :Tmp :Log/;
$status = ogg2mp3(file.ogg);
$status = ogg2wav(file.ogg);
$status = ogg2wav(in.ogg, out.wav);
$status = mp32ogg(file.mp3);
$status = mp32wav(file.mp3);
$status = mp32wav(in.mp3, out.wav);
$status = wav2ogg(file.wav);
$status = wav2ogg(in.wav, out.ogg);
$status = wav2mp3(file.wav);
$status = wav2mp3(in.wav, out.mp3);
Audio::ConvTools::logMsg(This is a log message);
Audio::ConvTools::errMsg(This is an error message);
$tmp = Audio::ConvTools::getTmpFile(.wav);
Audio::ConvTools::destroyTmpFile($tmp);
Download (0.007MB)
Added: 2006-11-08 License: Perl Artistic License Price:
1088 downloads
FLV::ToMP3 0.17
FLV::ToMP3 can convert audio from a FLV file into an MP3 file. more>>
FLV::ToMP3 can convert audio from a FLV file into an MP3 file.
SYNOPSIS
use FLV::ToMP3;
my $converter = FLV::ToMP3->new();
$converter->parse_flv($flv_filename);
$converter->save($mp3_filename);
See also flv2mp3.
Extracts audio data from an FLV file and constructs an MP3 file. See the flv2mp3 command-line program for a nice interface and a detailed list of caveats and limitations.
<<lessSYNOPSIS
use FLV::ToMP3;
my $converter = FLV::ToMP3->new();
$converter->parse_flv($flv_filename);
$converter->save($mp3_filename);
See also flv2mp3.
Extracts audio data from an FLV file and constructs an MP3 file. See the flv2mp3 command-line program for a nice interface and a detailed list of caveats and limitations.
Download (1.2MB)
Added: 2006-11-08 License: Perl Artistic License Price:
1156 downloads
Apache::Audio::DB 0.01
Apache::Audio::DB is a Perl module that can generate a database of your tunes complete with searchable interface. more>>
Apache::Audio::DB is a Perl module that can generate a database of your tunes complete with searchable interface and nifty statistical analyses!
SYNOPSIS
# httpd.conf or srm.conf AddType audio/mpeg mp3 MP3
# httpd.conf or access.conf
< Location /songs >
SetHandler perl-script
PerlHandler Apache::MP3::Sorted
PerlSetVar SortFields Album,Title,-Duration
PerlSetVar Fields Title,Artist,Album,Duration
< /Location >
<<lessSYNOPSIS
# httpd.conf or srm.conf AddType audio/mpeg mp3 MP3
# httpd.conf or access.conf
< Location /songs >
SetHandler perl-script
PerlHandler Apache::MP3::Sorted
PerlSetVar SortFields Album,Title,-Duration
PerlSetVar Fields Title,Artist,Album,Duration
< /Location >
Download (0.061MB)
Added: 2006-10-11 License: Perl Artistic License Price:
1108 downloads
Audio::MPEG 0.04
Audio::MPEG is a Perl module for encoding and decoding of MPEG Audio (MP3). more>>
Audio::MPEG is a Perl module for encoding and decoding of MPEG Audio (MP3).
SYNOPSIS
use Audio::MPEG;
Audio::MPEG is a Perl interface to the LAME and MAD MPEG audio Layers I, II, and III encoding and decoding libraries.
Rationale
I have been building a fairly extensive MP3 library, and decided to write some software to help manage the collection. Its turned out to be a rather cool piece of software (incidentally, I will be releasing it under the GPL shortly), with both a web and command line interface, good searching, integrated ripping, archive statistics, etc.
However, I also wanted to be able to stream audio, and verify the integrity of files in the archive. It is certainly possible to stream audio (even with re-encoding at a different bitrate) without resorting to writing interface glue like this module, but verification of the files was clumsy at best (e.g. scanning stdout/err for strings), and useless at worst.
Thus, Audio::MPEG was born.
LAME
This is arguably the best quality MPEG encoder available (certainly the best GPL encoder). Portions of the code have been optimized to take advantage of some of the advanced features for Intel/AMD processors, but even on non-optimized machines, such as the PowerPC, it performs quite well (faster than real-time on late 90s (and later) machines).
MAD
This is a relatively new MPEG decoding library. I chose it after struggling to clean up the MPEG decoding library included with LAME (which is based on Michael Hipps mpg123(1) implementation). In the end, I was very pleased with the results. MAD performs its decoding with an internal precision of 24 bits (pro-level quality) with fixed-point arithmetic. The code is very clean, and seems rock-solid. Although it may seem that it should be faster than the mpg123(1) library due to the use of fixed-point arithmetic, it is in fact about 60% or so of the speed (due to the higher resolution audio). However, the ease of coding against MAD, and the higher precision of the output more than makes up for the slower decoding.
Audio::MPEG can export the data at its highest precision for programs that wish to manipulate the data at the higher resolution.
Operating System Environment
I have only tested this on a Linux 2.4.x system so far, but I see no reason why it should not work on any Un*x variant. In fact, it may actually even work on a Windoze box (the underlying LAME and MAD libraries apparently compile somehow on them). I am doing no special magic with the interface, so presumably it will work under Windows. As you can probably tell, I dont really care if it does (Ill may start caring if M$ releases the source code to Windows under GPL, BSD, or Artistic licenses...). But, for you poor, misguided souls that insist upon running Windows, I expect that there should be little problem getting it to work.
Performance
You would think that with encoding/decoding audio, which is quite a compute-intensive task, Perl would be much slower than the equivalent pure C programs. Surprise... it is only about 3% slower (!) Even with the mechanism I use here (Perl->C->Perl for every frame, Perl 5.6.1 and Linux 2.4.4 (PowerPC 7500) performs just fantastic. So, the moral of this paragraph is to run your own performance tests, but theres no need to think of your own Perl encoder/decoder will be inferior to a pure C/C++ implementation. The only drawback is that, depending upon how much buffer space you use for reading, memory usage will be at least 3 times as much (eh... RAM is cheap...)
<<lessSYNOPSIS
use Audio::MPEG;
Audio::MPEG is a Perl interface to the LAME and MAD MPEG audio Layers I, II, and III encoding and decoding libraries.
Rationale
I have been building a fairly extensive MP3 library, and decided to write some software to help manage the collection. Its turned out to be a rather cool piece of software (incidentally, I will be releasing it under the GPL shortly), with both a web and command line interface, good searching, integrated ripping, archive statistics, etc.
However, I also wanted to be able to stream audio, and verify the integrity of files in the archive. It is certainly possible to stream audio (even with re-encoding at a different bitrate) without resorting to writing interface glue like this module, but verification of the files was clumsy at best (e.g. scanning stdout/err for strings), and useless at worst.
Thus, Audio::MPEG was born.
LAME
This is arguably the best quality MPEG encoder available (certainly the best GPL encoder). Portions of the code have been optimized to take advantage of some of the advanced features for Intel/AMD processors, but even on non-optimized machines, such as the PowerPC, it performs quite well (faster than real-time on late 90s (and later) machines).
MAD
This is a relatively new MPEG decoding library. I chose it after struggling to clean up the MPEG decoding library included with LAME (which is based on Michael Hipps mpg123(1) implementation). In the end, I was very pleased with the results. MAD performs its decoding with an internal precision of 24 bits (pro-level quality) with fixed-point arithmetic. The code is very clean, and seems rock-solid. Although it may seem that it should be faster than the mpg123(1) library due to the use of fixed-point arithmetic, it is in fact about 60% or so of the speed (due to the higher resolution audio). However, the ease of coding against MAD, and the higher precision of the output more than makes up for the slower decoding.
Audio::MPEG can export the data at its highest precision for programs that wish to manipulate the data at the higher resolution.
Operating System Environment
I have only tested this on a Linux 2.4.x system so far, but I see no reason why it should not work on any Un*x variant. In fact, it may actually even work on a Windoze box (the underlying LAME and MAD libraries apparently compile somehow on them). I am doing no special magic with the interface, so presumably it will work under Windows. As you can probably tell, I dont really care if it does (Ill may start caring if M$ releases the source code to Windows under GPL, BSD, or Artistic licenses...). But, for you poor, misguided souls that insist upon running Windows, I expect that there should be little problem getting it to work.
Performance
You would think that with encoding/decoding audio, which is quite a compute-intensive task, Perl would be much slower than the equivalent pure C programs. Surprise... it is only about 3% slower (!) Even with the mechanism I use here (Perl->C->Perl for every frame, Perl 5.6.1 and Linux 2.4.4 (PowerPC 7500) performs just fantastic. So, the moral of this paragraph is to run your own performance tests, but theres no need to think of your own Perl encoder/decoder will be inferior to a pure C/C++ implementation. The only drawback is that, depending upon how much buffer space you use for reading, memory usage will be at least 3 times as much (eh... RAM is cheap...)
Download (00057MB)
Added: 2006-06-19 License: GPL (GNU General Public License) Price:
1225 downloads
Audio Format Converter 0.7.0
Audio Format Converter is a an Amarok script that allows you to convert the audio format of the tracks in the playlist. more>>
Audio Format Converter is a an Amarok script that allows you to convert the audio format of the tracks in the playlist. The script takes a snapshot of the current playlist. Once the conversion has started, you can go back to using amaroK normally.
Supported files for both input and output include .flac, .ogg, .mp3, .mpc, .m4a, and .wav. Additionally, .wma, .ra, and are supported for input. (See below for the required packages.)
I dont know how to do anything with metatags from .wma or .ra. All I know is tags will get transferred to the output file if and only if the tags show up in the amaroK playlist. If they arent there, I suggest converting the files, then using a real tag editor like Easy Tag to put them in to the converted files.
This has only been tested on amaroK 1.2.4, but it should work on any amaroK>=1.2
The original files are NOT deleted. If the output file will overwrite an existing file, a dialog will appear to ask you whether to go ahead and overwrite or to skip the track. If you try to convert a file into the format its already in, a dialog will ask you how to handle this situation.
Depending on the number of tracks and the speed of your computer, this can take a long time to run. With each track, a passive popup will let you know what track its processing. At any time, you can stop the script by clicking stop in the amaroK script GUI.
<<lessSupported files for both input and output include .flac, .ogg, .mp3, .mpc, .m4a, and .wav. Additionally, .wma, .ra, and are supported for input. (See below for the required packages.)
I dont know how to do anything with metatags from .wma or .ra. All I know is tags will get transferred to the output file if and only if the tags show up in the amaroK playlist. If they arent there, I suggest converting the files, then using a real tag editor like Easy Tag to put them in to the converted files.
This has only been tested on amaroK 1.2.4, but it should work on any amaroK>=1.2
The original files are NOT deleted. If the output file will overwrite an existing file, a dialog will appear to ask you whether to go ahead and overwrite or to skip the track. If you try to convert a file into the format its already in, a dialog will ask you how to handle this situation.
Depending on the number of tracks and the speed of your computer, this can take a long time to run. With each track, a passive popup will let you know what track its processing. At any time, you can stop the script by clicking stop in the amaroK script GUI.
Download (0.006MB)
Added: 2007-04-30 License: GPL (GNU General Public License) Price:
914 downloads
Rainbyte Video Converter 0.01 Beta
Rainbyte Video Converter is a Kommander script which helps you convert video and audio in an easy way. more>>
Rainbyte Video Converter is a Kommander script which helps you convert video and audio in an easy way.
Its based on Kommander, mencoder and ffmpeg and you can convert from/to any format supported by those programs.
Well, I hope that it will be usefull for you, thanks for all.
<<lessIts based on Kommander, mencoder and ffmpeg and you can convert from/to any format supported by those programs.
Well, I hope that it will be usefull for you, thanks for all.
Download (0.012MB)
Added: 2007-02-27 License: GPL (GNU General Public License) Price:
985 downloads
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