Debian Multimedia Audio utilities packages

Aubio gromadzi zestaw funkcji do segmentowania i oznakowywania sygnału dźwiękowego. Biblioteka zawiera wokoder faz, funkcje wykrywania początku i skoku, algorytm śledzenia rytmu oraz inne narzędzia do obróbki dźwięku.

Pakiet zawiera narzędzia wiersza poleceń do uruchamiania różnych algorytmów.

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, MusicBrainz or compatible servers.

Multi-platfrom system for audio measurement through sound card in the PC. Incorporates Generator, Oscilloscope, FFT, Sweep frequency characteristic. Useful for measuring stereo and speaker performance.

The command line tool bs1770gain is a loudness scanner compliant with ITU-R BS.1770 and its flavors EBU R128, ATSC A/85, and ReplayGain 2.0. It helps normalizing the loudness of audio and video files to the same level.

In order to do CDDB queries over the Internet, you must know the DiscID of the CD you are querying. cd-discid provides you with that information. It outputs the discid, the number of tracks, the frame offset of all of the tracks, and the total length of the CD in seconds, on one line in a space-delimited format. cd-discid was designed as a backend tool for cdgrab (now abcde) but will work independently of it.

EasyTAG is an utility for viewing, editing and writing the tags of different audio files, using a GTK+ interface.

Currently EasyTAG supports the following:

• View, edit, write tags of MP3, MP2 files (ID3 tag), FLAC files (FLAC Vorbis tag), Ogg Opus, Ogg Speex and Ogg Vorbis files (Ogg Vorbis tag), MP4/M4A/AAC files (MPEG-4 Part 10 tag), and MusePack, Monkey's Audio files (APE tag);
• Auto tagging: parse file and directory names using masks to automatically fill in tag fields;
• Cover art support for all formats;
• Rename files from the tag fields (using masks) or by loading a text file;
• Process selected files of the selected directory;
• Ability to browse subdirectories;
• Recursion for tagging, removing, renaming, saving, etc;
• Can set a field (artist, title, ...) on all other selected files;
• Read file header information (bitrate, time, ...) and display it;
• Undo and redo last changes;
• Ability to process tag fields and file names (convert letters into uppercase, lowercase, etc);
• Ability to open a directory or a file with an external program;
• CDDB support (from http protocol);
• A tree based browser;
• A list to select files;
• A playlist generator window;
• A file searching window;
• Simple and explicit interface.

EasyTAG to narzędzie do przeglądania, edytowania i zapisywania znaczników różnych plików audio za pomocą interfejsu GTK+.

Jest to rozszerzenie dla menedżera plików Nautilus.

Ecasound is a software package designed for multitrack audio processing. It can be used for simple tasks like audio playback, recording and format conversions, as well as for multitrack effect processing, mixing, recording and signal recycling.

Ecasound supports a wide range of audio inputs, outputs and effect algorithms. Effects and audio objects can be combined in various ways, and their parameters can be controlled by operator objects like oscillators and MIDI-CCs.

This package provides the following tools:

• ecaconvert: tool for converting a set of files to a common target format
• ecafixdc: command-line tool for fixing DC-offset
• ecalength: tool for printing audio file length and other information
• ecamonitor: tool for monitoring an active ecasound session
• ecanormalize: command-line tool for normalizing audio files
• ecaplay: command-line tool for playing audio files

Tagger for whole-album FLAC files with embedded CUE sheets. Data for the tags is retrieved from the MusicBrainz service. Tags are written as VORBIS comments or standalone CUESHEET files, and the application has the ability to rename a file based on the tags using a user-specified template.

flactag can be run either interactively (to allow the user to select which of the results to tag the file with), or in batch mode (allowing files to be checked regularly for changes, and optionally to apply these changes automatically).

Glyr is a library to retrieve music-related metadata from several online providers (Amazon, Discogs, Musicbrainz...).

It can be used in media player programs to download cover art, lyrics, biography and much more.

This package contains a command-line interface to the library.

gmusicbrowser is a powerful graphical browser which supports libraries with a very large number of songs (over 10,000).

It can use multiple inputs and has native support for MP3, Ogg, and FLAC files. It also supports mass-renaming and mass-retagging of a song library, multiple genres per song, ratings, and customizable labels.

gmusicbrowser has a customizable window layout, and comes with plugins to use Last.fm, retrieve lyrics, or find album pictures and WebContext (using the Mozilla or WebKit engines to display the artist's page on Wikipedia and search for lyrics with Google).

Metronome is a GNOME application to help you keep the tempo while practising your instrument.

Features:

• Select the tempo by changing the beats per minute, or by tapping it
• Select the time signature
• Play the metronome

isrcsubmit is a command line utility to extract International Standard Recording Codes (ISRC) from audio CDs. It allows one to submit the extracted data to MusicBrainz. ISRCs are used to uniquely identify sound and music video recordings.

isrcsubmit supports multiple backends to extract ISRCs: libdiscid (the default) and cdrdao for discs that do not store ISRCs in the subchannel information.

A valid MusicBrainz account is required to submit ISRCs.

libambix is a library of C routines for reading and writing files following the "ambix" (AMBIsonics eXchange) conventions.

Ambisonics is a periphonic (3D) surround sound technique, with a scalable spatial resolution (put simply: "the more audio channels you use, the better"). The ambix convention defines an extensible format for exchanging soundfiles containing Higher Order Ambisonics data.

This package contains additional commandline utilities to work with ambix-files:

• ambix-info: Print information on ambix files
• ambix-interleave: Merges several audio files into an ambix file
• ambix-deinterleave: Split an ambix file into several mono files
• ambix-jplay: Play back an ambix file via JACK
• ambix-jrecord: Record an ambix file via JACK

libmysofa is a light weight C-library intended to read SOFA (Spatially Oriented Format for Acoustics) files for spatial rendering. It hardly has any library dependencies and is suitable for embedded devices.

It reads SOFA files and checks whether the data complies to the "SimpleFreeFieldHRIR" conventions. In addition, it provides functions to look-up and interpolate the filters for a given orientation and to normalize the HRTFs (Head-Related Transfer Functions) to a reference level.

This package contains the mysofa2json utility.

MediaConch is an extensible, open source software project consisting of an implementation checker, policy checker, reporter, and fixer that targets preservation-level audiovisual files (specifically Matroska, Linear Pulse Code Modulation (LPCM) and FF Video Codec 1 (FFV1)) for use in memory institutions, providing detailed and batch-level conformance checking via an adaptable and flexible application program interface accessible by the command line, a graphical user interface, or a web-based shell.

This package contains the command-line utiltiy.

mhWaveEdit is a graphical program for editing sound files. It is intended to be user-friendly and robust. It does not require a fast computer.

Supports JACK, ALSA, OSS and PulseAudio.

MP3FS is a read-only FUSE filesystem which transcodes audio formats (currently FLAC) to MP3 on the fly when opened and read.

It is also a novel alternative to traditional mp3 encoder applications. Users can use their favorite file browser to select the files they want encoded and copy them somewhere.

A customizable program to scan a list of (sub)directories, creating a report from an HTML template. Also calculates various statistics and each song's playing time. Supports ID3 and ID3v2 tags.

Perl Audio Converter is a tool for converting multiple audio types from one format to another using various external encoders/decoders.

It supports the following audio formats (but the backend program that handles a given format might not be packaged in Debian): 3G2, 3GP, 8SVX, AAC, AC3, ADTS, AIFF, AL, AMB, AMR, APE, AU, AVR, BONK, CAF, CDR, CVU, DAT, DTS, DVMS, F32, F64, FAP, FLA, FLAC, FSSD, GSRT, HCOM, IMA, IRCAM, LA, MAT, MAUD, MAT4, MAT5, M4A, MP2, MP3, MP4, MPC, MPP, NIST, OFF, OFR, OFS, OPUS, OGA, OGG, PAF, PRC, PVF, RA, RAM, RAW, RF64, SD2, SF, SHN, SMP, SND, SOU, SPX, SRN, TAK, TTA, TXW, VOC, VMS, VQF, W64, WAV, WMA, and WV.

It can also convert audio from the following video extensions: RM, RV, ASF, DivX, MPG, MKV, MPEG, AVI, MOV, OGM, OGV, QT, VCD, SVCD, M4V, NSV, NUV, PSP, SMK, VOB, FLV, WEBM, and WMV.

Pacpl also features parallel processing, a CD ripping function with CDDB support, batch conversion, tag preservation for most supported formats and independent tag reading & writing. Service menus for KDE (Dolphin/Konqueror), GNOME Nautilus script, and Nemo action script are also provided (see README.Debian).

You can write your own modules in order to add support for new file formats.

This is a program for stretching the audio. It is suitable only for extreme sound stretching of the audio (like 50x) and for applying special effects by "spectral smoothing" the sounds. It can transform any sound/music to a texture.

It produces high quality extreme sound stretching. While most sound stretching software sounds bad when trying to stretch the sounds a lot, this one is optimized for extreme sound stretching.

The stretch amount is unlimited. You can play the stretched sound in real-time (including the possibility to "freeze" the sound) or you can render the whole sound or a part of it to audio files.

It has many post-processing effects, like: filters, pitch/frequency shifters Support for WAV, OGG VORBIS files and MP3 files.

Play it slowly is a piece of software to play back audio files at a different speed or pitch. It also allows you to loop over a certain part of a file.

It is intended to help you learn or transcribe songs. It can also play videos thanks to GStreamer.

Play it slowly works with both ALSA and the JACK Audio Connection Kit.

Pragha is a Lightweight Music Player for GNU/Linux, based on Gtk, sqlite, and completely written in C, constructed to be fast, light, and simultaneously tries to be complete without obstructing the daily work.

ReZound is an advanced audio file editor for the GNU/Linux platform. It supports the JACK audio server and the OSS sound API. It can import several audio file formats such Ogg/Vorbis, mp3, standard wav, FLAC and exports in its own file format (.rez). You can use ReZound to burn your audio files directly to CD. ReZound has unlimited Undo. It supports the LADSPA effect plugin standard and has several builtin effects.

Rubber Band is a library and utility program that permits you to change the tempo and pitch of an audio recording independently of one another.

This package contains a command-line utility that can be used to exploit Rubber Band's capabilities.

Rubber Band is a library and utility program that permits you to change the tempo and pitch of an audio recording independently of one another.

This package contains a LADSPA plugin that can change the pitch of a sound in real-time.

Rubber Band is a library and utility program that permits you to change the tempo and pitch of an audio recording independently of one another.

This package contains the following Vamp plugins:

• increments (Output Increments): Output time increment for each input step

• aggregate_increments (Accumulated Output Increments): Accumulated output time increments

• divergence (Divergence from Linear): Difference between actual output time and the output time for a theoretical linear stretch

• phaseresetdf (Phase Reset Detection Function): Curve whose peaks are used to identify transients for phase reset points

• smoothedphaseresetdf (Smoothed Phase Reset Detection Function): Phase reset curve smoothed for peak picking

• phaseresetpoints (Phase Reset Points): Points estimated as transients at which phase reset occurs

• timesyncpoints (Time Sync Points): Salient points which stretcher aims to place with strictly correct timing

Silan to niezależna aplikacja do analizowania plików dźwiękowych pod kątem występowania w nich obszarów ciszy oraz drukowania przedziałów wykrytych sygnałów.

Obsługuje wiele formatów audio i kodeków, wykorzystując libsndfile i ffmpeg/libav do odczytu danych dźwiękowych.

Próg sygnału i czas wstrzymania można dowolnie konfigurować. Dane wyjściowe mogą być formatowane przy użyciu próbek dźwiękowych lub sekund jako jednostek lub drukowane w formie pliku Audacity.

Snd is a powerful sound file editor that can be customized and extended using the Scheme programming language.

This package depends on a full installation of Snd.

Snd is a powerful sound file editor that can be customized and extended using the Scheme programming language.

Snd no longer supports GTK, so this package now depends on the generic "snd-gui-jack" package.

This is a dummy package and it may be safely removed after installation.

Snd is a powerful sound file editor that can be customized and extended using the Scheme programming language.

Snd no longer supports GTK, so this package now depends on the generic "snd-gui-pulse" package.

This is a dummy package and it may be safely removed after installation.

Snd is a powerful sound file editor that can be customized and extended using the Scheme programming language.

This package contains a version of snd compiled with both JACK and ALSA support, but without support for X.

Celem projektu Sonic Visualiser jest stworzenie programu, służącego do studiowania nagrania muzycznego zamiast jego odsłuchu.

Sonic Visualiser może być szczególnie interesujący dla muzykologów, archiwistów, badaczy zajmujących się przetwarzaniem sygnału i każdego, kto szuka przyjaznego sposobu, aby przyjrzeć się temu, co znajduje się w pliku muzycznym.

SoundConverter jest prostym konwerterem dźwięku do środowiska GNOME. Czyta pliki dźwiękowe w dowolnym formacie obsługiwanym przez GStreamer i przekształca je w Ogg Vorbis, FLAC, WAV, bądź MP3 jeśli użytkownik posiada wtyczkę GStreamer LAME.

This program can stretch and pitch-shift .wav-files independently. This is especially useful for musicians, who want to listen to a song played slowly, without having the pitch altered.

This program uses the SoundTouch shared library for the actual work.

SoX jest narzędziem wiersza poleceń, które może konwertować różne formaty komputerowych plików audio do innych formatów. Może również dodawać różne efekty do tych plików dźwiękowych podczas konwersji. Jako dodatkowy bonus, SoX może odtwarzać i nagrywać pliki audio na wielu uniksowych platformach.

SoX jest w stanie obsłużyć formaty, takie jak: Ogg Vorbis, MP3, WAV, AIFF, VOC, SND, AU, GSM oraz kilka innych. Każdy obsługiwany format wymaga przynajmniej zainstalowania pakietu libsox-fmt-base. Niektóre formaty posiadają własne pakiety, np. obsługa odczytu i zapisu plików MP3 jest zapewniona przez libsox-fmt-mp3.

SoX obsługuje większość popularnych architektur dźwięku tj. ALSA, Libao, OSS i Pulse (odpowiednio dostarczane przez pakiety libsox-fmt-alsa, libsox-fmt-ao, libsox-fmt-oss i libsox-fmt pulse). Obsługuje również wtyczki LADSPA.

StretchPlayer służy do odtwarzania plików audio umożliwiając rozciąganie w czasie (bez wpływu na wysokość dźwięku) i/lub zmianę wysokości dźwięku (bez wpływu na czas odtwarzania dźwięku), nawet podczas słuchania.

Ostrzeżenie: pakiet zawiera wersję rozwojową (potencjalnie niestabilną) wydaną wyłącznie w celach testowych.

vorbis-tools zawierają: oggenc (koder), ogg123 (narzędzie odtwarzania), ogginfo (wyświetla informacje ogg), oggdec (dekoduje pliki ogg), vcut (narzędzie do dzielenia plików ogg) i vorbiscomment (edytor komentarza ogg).

ogg123 może odtwarzać zarówno strumienie audio FLAC jak i Ogg Vorbis.

Zasadniczym celem aplikacji jest pobranie pliku w formacie wave i podzielenie go na wiele tego typu plików. Wykonuje czystą przerwę w odpowiednim miejscu, aby nagrać pliki na płytę audio CD bez przestojów między ścieżkami.

WavPack jest całkowicie otwartym formatem kompresji dźwięku, zapewniającym bezstratny, wysokojakościowy stratny, oraz unikalny hybrydowy tryb kompresji. Chociaż technologia ta jest luźno oparta na poprzednich wersjach WavPack, to nowa 4 wersja formatu została zaprojektowana od podstaw, aby zaoferować niezrównaną wydajność i funkcjonalność.

Pakiet zawiera koder i dekoder do plików WavPack.

XCFA is a tool to extract the contens of Audio-CDs and convert musical audio files conversion to FLAC, WAV, OGG, M4A, MPC, MP3, WavPack and many other formats.

Some features follow:

• Handling of frequency, track and bits.
• Handling of files shared volume before a recording.
• Web retrieval of Audio-CD information during treatment.
• Small cover creation for Audio-CDs.

Dynamic range compression reduces the volume of loud sounds or amplifies quiet sounds thus reducing or compressing an audio signal's dynamic range. DC1 provides a dynamics compressor for use with JACK Audio Connection Kit. It can be used as master effect or as an effect on individual instrument tracks as well.

Input format is any file readable by libsndfile, output is either WAV (WAVEX for more than 2 channels) or CAF. Apart from resampling, users can change the sample format to 16-bit, 24-bit or float, and for 16-bit output, add dithering.

Available dithering types are rectangular, triangular and Lipschitz' optimised error feedback filter. Some examples of dithering can be seen here: https://kokkinizita.linuxaudio.org/linuxaudio/dithering.html

Ardour is a multichannel hard disk recorder (HDR) and digital audio workstation (DAW). It can be used to control, record, edit and run complex audio setups.

Ardour supports pro-audio interfaces through the ALSA project, which provides high quality, well designed device drivers and API's for audio I/O under Linux. Any interface supported by ALSA can be used with Ardour. This includes the all-digital 26 channel RME Hammerfall, the Midiman Delta 1010 and many others.

Ardour has support for 24 bit samples using floating point internally, non-linear editing with unlimited undo, a user-configurable mixer, MTC master/slave capabilities, MIDI hardware control surface compatibility.

It supports MIDI Machine Control, and so can be controlled from any MMC controller and many modern digital mixers.

Ardour contains a powerful multitrack audio editor/arranger that is completely non-destructive and capable of all standard non-linear editing operations (insert, replace, delete, move, trim, select, cut/copy/paste). The editor has unlimited undo/redo capabilities and can save independent "versions" of a track or an entire piece

Ardour's editor supports the community-developed LADSPA plugin standard. Arbitrary chains of plugins can be attached to any portion of a track. Every mixer strip can have any number of inputs and outputs, not just mono, stereo or 5.1. An N-way panner is included, with support for various panning models. Pre- and post-fader sends exist, each with their own gain and pan controls. Every mixer strip acts as its own bus, and thus the bus count in Ardour is unlimited. You can submix any number of strips into another strip.

Ardour's channel capacity is limited only by the number on your audio interface and the ability of your disk subsystem to stream the data back and forth.

JACK (the JACK Audio Connection Kit) is used for all audio I/O, permitting data to be exchanged in perfect samplesync with other applications and/or hardware audio interfaces.

Ardour is sample rate and size neutral - any hardware formats from 8 to 32 bits, and rates from 8kHz to 192kHz. Internal processing in 32/64 bit IEEE floating point format.

Further information can be found at https://ardour.org/.

Ardour jest wielokanałową nagrywarką (Hard Disk Recorder - HDR) i stacją roboczą do obsługi dźwięku cyfrowego (Digital Audio Workstation - DAW). Może być używany do kontrolowania, nagrywania, edytowania i uruchamiania złożonych zestawów audio.

Jest to pakiet przejściowy. Można go bezpiecznie usunąć.

Audacity jest wielościeżkowym edytorem audio do systemów Linux/Unix, MacOS oraz Windows. Jest zaprojektowany z myślą o łatwym nagrywaniu, odtwarzaniu i edytowaniu cyfrowych plików audio. Audacity zawiera cyfrowe efekty oraz narzędzia do analizy widma. Edytowanie jest bardzo szybkie i umożliwia wykonywanie nieograniczonej liczby operacji cofnij/ponów.

Do wspieranych plików należą: Ogg Vorbis, MP3, WAV, AIFF oraz AU.

AudioLink is a tool that makes searching for music on your local storage media easier and faster. Your searches can include a variety of criteria, like male artists, female artists, band, genre, etc.

It works with MP3 and Ogg Vorbis files and creates a MySQL database in which it stores the information about the music files. It creates symbolic links to the actual music files based on the search results. You can search for multiple fields, like artist, band, composer, lyricist, etc.

cdrdao nagrywa dane lub ścieżki dźwiękowe na płyty CD-R w trybie Disk-At-Once (DAO), bazując na tekstowym opisie zawartości CD.

Nagrywanie w trybie Disk-At-Once zapisuje cały dysk, tzn. obszar lead-in, jedną lub więcej ścieżek i lead-out w jednym kroku. Często używany tryb Track-At-Once (TAO) zapisuje ścieżki niezależnie, co wymaga powiązania bloków pomiędzy dwoma ścieżkami. Dobrym przykładem użycia tej metody jest kopiowanie płyty CD z wieloma ścieżkami np. większości płyt audio.

cdrdao może obsłużyć format bin/cue, często używany do VCD lub płyt z danymi subkanałowymi.

Do nagrania zwykłej płyty CD z danymi, lepszym wyborem będzie prawdopodobnie użycie programu wodim.

Ecasound is a software package designed for multitrack audio processing. It can be used for simple tasks like audio playback, recording and format conversions, as well as for multitrack effect processing, mixing, recording and signal recycling.

Ecasound supports a wide range of audio inputs, outputs and effect algorithms. Effects and audio objects can be combined in various ways, and their parameters can be controlled by operator objects like oscillators and MIDI-CCs.

This package provides the ecasound executable.

This application based on libsdl and on the alsa audio driver provides visualisation to the sound reproduced by a sound device creating waves and particular effects.

Plays audio sources to transcribe them in your favourite text application. Speed of playback can be changed without changing the pitch and on pause it will rewind for a few seconds according to your preferences. Parlatype can be controlled with the "Play" button from your keyboard even if its window is not focused.

This is a MIDI/Audio multi-track sequencer application written in C++.

The Jack Audio Connection Kit (JACK) for audio, and the Advanced Linux Sound Architecture (ALSA) for MIDI, are the main infrastructures to evolve as a fairly-featured Linux Desktop Audio Workstation GUI, especially dedicated to the personal home-studio.

This is a free and open source media converter frontend to FFmpeg. FFmpeg is a powerful command-line utility to convert audio and video file into numerous formats. QWinFF features a rich set of presets to help users use FFmpeg easily without having to manually input command-line flags. Average users can convert multiple media files in just a few clicks, while advanced users can still adjust conversion parameters in detail.

Rosegarden jest aplikacją Qt będącą sekwencerem miksującym audio/MIDI (odtwarzanie i nagrywanie) oraz edytorem wielościeżkowym do edycji muzyki, który używa zarówno notacji taśmy perforowanej, jak i partytury, obsługuje wejście/wyjście plików MIDI, eksport do plików lilypond i Csound itp.

Spek helps to analyse your audio files by showing their spectrogram. It supports all popular lossy and lossless audio file formats.

Features:

• Ultra-fast signal processing, uses multiple threads to further speed up the analysis
• Shows the codec name and the audio signal parameters
• Can save the spectrogram as an image file
• Drag-and-drop support; associates with common audio file formats
• Auto-fitting time, frequency and spectral density rulers
• Adjustable spectral density range

hit enter key with style an you'll get the corresponding number of beats per minute (bpm). This tool is very useful to quickly find the tempo of a song without launching a big digital workstation like Ardour or LMMS. Just type "taptempo" in a terminal and hit the enter key while your hearing a song to print the tempo.

A simple AC3-compatible audio encoder based on FFmpeg.

AudioSeg is a toolkit dedicated to audio segmentation and classification of audio streams. The toolkit implements standard reference algorithms such as energy-based silence detection, BIC segmentation and clustering as well as GMM/HMM classification.

cutmp3 is a small and fast command line MP3 editor. It lets you select sections of an MP3 interactively or via a timetable and save them to separate files without quality loss. It uses mpg123 for playback and works with VBR files and even with files bigger than 2GB. Other features are configurable silence seeking and ID3 tag seeking, which are useful for concatenated mp3s.

A free DTS Coherent Acoustics decoder with support for HD extensions.

Supported features:

 Decoding of standard DTS core streams with up to 5.1 channels
 Decoding of DTS-ES streams with discrete back channel
 Decoding of High Resolution streams with up to 7.1 channels and extended
 bitrate
 Decoding of 96/24 core streams
 Lossless decoding of Master Audio streams with up to 7.1 channels, 192 kHz
 Downmixing to stereo and 5.1 using embedded coefficients

fdmf finds pairs of music files in a collection that are likely to contain the same music. It works on the music itself, not on the filename, tags, or headers. It uses an audio fingerprint, or perceptual hash to recognise the duplicate files.

FF Multi Converter is a simple graphical application which enables the user to convert audio, video, image and document files between all popular formats by using and combining other programs. It uses avconv for audio/video files, unoconv for document files and ImageMagick for image file conversions.

The goal of FF Multi Converter is to gather the most popular multimedia types in one application and provide conversion options for them easily through a user-friendly interface.

FFT Spectra is a tool for visualization of frequency spectra of an audio signal. What is it good for? For instance, you can verify whether you are able to sing and whistle in-tune or tune your guitar.

Fueco is usefull to understand ffmpeg and mencoder. It can convert to .ogg, to .wav, to ac3 video, etc...

The Coveramazon plugin is a a metadata provider for gmpc which downloads cover art and album information.

mp3cat can read a stream from standard input and write to standard output. Instead of standard output, mp3cat can store MP3 data across multiple timestamped files in a directory (an "mp3dir") and another instance can follow the stream from there.

Also, mp3cat only outputs MP3 frames with valid headers, even if there is extra garbage in its input stream. This fixes up time/frame estimates in a stream created by simply concatenating mp3 files or by other utilities that add app-specific frames.

A fork of the dormant Grip package, MP3Mesa, like Grip enables ripping from CD's to a variety of formats, and includes freedb.org lookup and also entry/update of freedb.org entries.

MP3packer is a program which attempts to save space of MP3 files by storing frame data in the smallest possible frame.

By default, the program generates the smallest MP3 possible (with the least padding), transforming Constant Bitrate (CBR) files in Variable Bitrate (VBR) files. With files generated by the LAME MP3 encoder with "--preset insane", it can result in saving of up to 10% in size, losslessly.

Furthermore, it can turn VBR files into CBR for use with players which don't support VBR.

A graphical editor that allows you to replay or remove selected parts (frames) of an MP3 file.

Nautilus extension to convert audio files to a different format, using the GStreamer framework.

odvr is a user-space driver for Olympus digital voice recorders that do not support USB Mass Storage. There is no GUI at this time, and functionality is limited, but basic download and listing capabilities are implemented.

Playdar is a music content resolver service - run it on every computer you use, and you'll be able to listen to all the songs you would otherwise be able to find manually by searching though all your computers, hard disks, online services, and more.

I've created a bash shell script to convert between CD Audio, flac, ogg and mp3. It works with multiple input/output files and allows for complete preservation of audio tagging data including ReplayGain.

It is available from my personal webspace http://www.fortnight1.plus.com/software/ and is currently at version 0.5b (18th March 2008). Since it is undergoing rapid develeopment I've not fully bug-tested it and some newer features may be partly broken. It is architecture independent being a single script file.

License is LGPL, debian package contains fuller description and the README installed to /usr/share/doc/playlist has full instructions for using the current version.

Qrest is a toolbox aimed at music production people. It is a GUI application written in C++ and using the Qt libraries.

It helps guessing the tempo of a musical piece, and getting the period in millisecond (and related frequency) of the most common notes durations. It also handles common note variations like dotted notes and triplets.

Spleeter is the Deezer source separation library with pretrained models written in Python and uses Tensorflow. It makes it easy to train source separation model (assuming you have a dataset of isolated sources), and provides already trained state of the art model for performing various flavour of separation : Vocals (singing voice) / accompaniment separation (2 stems) Vocals / drums / bass / other separation (4 stems) * Vocals / drums / bass / piano / other separation (5 stems) 2 stems and 4 stems models have state of the art performances on the musdb dataset. Spleeter is also very fast as it can perform separation of audio files to 4 stems 100x faster than real-time when run on a GPU. We designed Spleeter so you can use it straight from command line as well as directly in your own development pipeline as a Python library.