Debian Multimedia Audio utilities packages

aubio gathers a set of functions for audio signal segmentation and labelling. The library contains a phase vocoder, onset and pitch detection functions, a beat tracking algorithm and other sound processing utilities.

This package provides command line tools to run the different algorithms.

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.

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

This is the extension for the Nautilus file manager.

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).

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.

isrsubmit 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.

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.

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: AAC, AC3, AIFF, APE, AU, AVR, BONK, CAF, CDR, FAP, FLA, FLAC, IRCAM, LA, LPAC, M4A, MAT, MAT4, MAT5, MMF, MP2, MP3, MP4, MPC, MPP, NIST, OFR, OFS, OGG, PAC, PAF, PVF, RA, RAM, RAW, SD2, SF, SHN, SMP, SND, SPX, TTA, VOC, 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, QT, VCD, SVCD, M4V, NSV, NUV, PSP, SMK, VOB, FLV, and WMV.

Pacpl also includes a CD ripping function with CDDB support, batch conversion, tag preservation for most supported formats, independent tag reading/writing, a simple kommander gui and extensions for Dolphin and Konqueror. You can write your own modules in order to add support for new file formats.

See README.Debian for more information about supported 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.

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 is a standalone application to analyze audio files for silence and print ranges of detected signals.

It supports a variety of audio-formats and codecs by making use of libsndfile and ffmpeg/libav for reading audio-data.

Signal threshold and hold-off time can be freely configured. The output can be formatted with samples or seconds as unit or printed as audacity label file.

The aim of Sonic Visualiser is to be the first program you reach for when want to study a musical recording rather than simply listen to it.

Sonic Visualiser could be of particular interest to musicologists, archivists, signal-processing researchers and anyone else looking for a friendly way to take a look at what lies inside the audio file.

SoundConverter is a simple sound converter application for the GNOME environment. It reads sound files in any format supported by GStreamer and outputs them in Ogg Vorbis, FLAC, or WAV format, or MP3 format if you have the GStreamer LAME plugin.

This program can strech 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 libsoundtouch for the actual work.

SoX is a command line utility that can convert various formats of computer audio files in to other formats. It can also apply various effects to these sound files during the conversion. As an added bonus, SoX can play and record audio files on several unix-style platforms.

SoX is able to handle formats like Ogg Vorbis, MP3, WAV, AIFF, VOC, SND, AU, GSM and several more. Any format support requires at least libsox-fmt-base. Some formats have their own package e.g. mp3 read and write support is provided by libsox-fmt-mp3.

SoX supports most common sound architectures i.e. Alsa, Libao, OSS and Pulse (respectively provided by libsox-fmt-alsa, libsox-fmt-ao, libsox-fmt-oss and libsox-fmt-pulse). It also supports LADSPA plugins.

StretchPlayer will play back an audio file and allow you to time stretch (without affecting pitch) and/or pitch shift (without affecting the time) the audio, even while you are listening to it.

Warning: this package contains the development version (potentially unstable) released for testing purpose only.

vorbis-tools contains oggenc (an encoder), ogg123 (a playback tool), ogginfo (displays ogg information), oggdec (decodes ogg files), vcut (ogg file splitter), and vorbiscomment (ogg comment editor).

ogg123 can play both Ogg Vorbis and FLAC audio streams.

This application's purpose in life is to take a wave file and break it up into multiple wave files. It makes a clean break at the correct position to burn the files to an audio cd without any dead air between the tracks.

WavPack is a completely open audio compression format providing lossless, high-quality lossy, and a unique hybrid compression mode. Although the technology is loosely based on previous versions of WavPack, the new version 4 format has been designed from the ground up to offer unparalleled performance and functionality.

This package contains an encoder and decoder for wavpack files.

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.

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: http://www.kokkinizita.net/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 http://ardour.org/.

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 http://ardour.org/.

Audacity is a multi-track audio editor for Linux/Unix, MacOS and Windows. It is designed for easy recording, playing and editing of digital audio. Audacity features digital effects and spectrum analysis tools. Editing is very fast and provides unlimited undo/redo.

Supported file formats include Ogg Vorbis, MP2, MP3, WAV, AIFF, and AU.

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.

Qtractor is a MIDI/Audio multi-track sequencer application written in C++ around the Qt4 toolkit using Qt Designer.

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.

Rosegarden is a Qt application which provides a mixed Audio/MIDI sequencer (for playback and recording), a multi-track editor, music editing using both piano-roll and score notation, MIDI file IO, lilypond and Csound files export, etc.

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.

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 MusicBrainz, FreeDB or FreeDB-compatible servers. Audio formats supported are: WAV AIFF Sun AU FLAC WavPack MP3 MP2 Ogg Vorbis Ogg Speex Ogg FLAC M4A Musepack

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.

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.

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.