Kwave ~ KDE Audio Editor


Kwave is a sound editor for the KDE environment. It is written with KDE/QT and is extendable through a powerful plugin interface. For the moment it supports .wav files and many other formats, recording/playback via PulseAudio, Qt Multimedia, OSS and ALSA and some simple effects.

Features:
  • user interface based on KDE Frameworks 5
  • editing of multi track sound files
  • playback via ALSA, OSS, Phonon, PulseAudio
  • recording via ALSA
  • supported file formats: WAV audio, Ogg/Vorbis, FLAC, AIFF,Amiga,Mac-Audio, ULAW (Sun)-Audio, Berkeley,IRCAM,Carl Sound Format, MP3-Audio
  • volume control
  • sample rate conversion
  • pitch shift
  • reverse
  • low pass, notch filter
  • normalizing

kwave.sourceforge.net/

sourceforge.net/projects/kwave/

AudiQ ~ Format mp3 File Names & Tags


All the mp3 files of the music albums that you can download have different name formats or the information tags are empty. AudiQ is a very easy to use application that allows you to format the mp3 file names the way you decide. This application also fills the ID3 tags with the information of each song from the filename. Normalize your audio library.

https://sourceforge.net/projects/audiq/

Demlo ~ Music Library Organiser


Demlo is a music library organizer. It can encode, fix case, change folder
hierarchy according to tags or file properties, tag from an online database,
copy covers while ignoring duplicates or those below a quality threshold, and
much more. It makes it possible to manage your libraries uniformly and
dynamically. You can write your own rules to fit your needs best.
Demlo can address any of these recurring music library issues (and much more):

  • Fix the lack of folder structure.
  • Normalize tags, fix their case, chose which tags to keep and which to discard.
  • Handle lossy and lossless audio differently.
  • Handle mp3 id3tags hell…
  • Handle multiple covers, whether embedded and/or external, resize covers,
    discard bad quality ones.

gitlab.com/ambrevar/demlo

MAnalyzer ~ Spectral Analyzer & Sonogram


MAnalyzer is an advanced spectral analyzer and sonogram containing unique features such as smoothing, normalization, super-resolution, prefiltering and deharmonization. The included meters provide a peak meter and EBU R128 and ITU-R BS 1770-3 compliant loudness meter.

MAnalyzer00.v1

www.meldaproduction.com/MAnalyzer

Dynamic Audio Normalizer ~ Advanced Normalization


Dynamic Audio Normalizer is a library for advanced audio normalization purposes. It applies a certain amount of gain to the input audio in order to bring its peak magnitude to a target level (e.g. 0 dBFS). However, in contrast to more “simple” normalization algorithms, the Dynamic Audio Normalizer dynamically re-adjusts the gain factor to the input audio. This allows for applying extra gain to the “quiet” sections of the audio while avoiding distortions or clipping the “loud” sections. In other words: The Dynamic Audio Normalizer will “even out” the volume of quiet and loud sections, in the sense that the volume of each section is brought to the same target level. Note, however, that the Dynamic Audio Normalizer achieves this goal without applying “dynamic range compression”. It will retain 100% of the dynamic range within each “local” region of the audio file.

The Dynamic Audio Normalizer is available as a small standalone command-line utility and also as an effect in the SoX audio processor as well as in the FFmpeg audio/video converter. Furthermore, it can be integrated into your favourite DAW (digital audio workstation), as a VST plug-in, or into your favourite media player, as a Winamp plug-in. Last but not least, the “core” library can be integrated into custom applications easily, thanks to a straightforward API (application programming interface). The “native” API is written in C++, but language bindings for C99 Microsoft.NETJavaPython and Pascal are provided.

muldersoft.com/#dyauno
github.com/lordmulder/DynamicAudioNormalizer

ANMP ~ Another Nameless Music Player


ANMP aims to be a versatile but lightweight audio player, just as the other hundred thousands out there. It is written in C++11. As being only a frontend, ANMP itself doesn’t know anything about audio formats and how to decode them. That’s why it uses 3rd party libraries to decode them. By using VgmStream, GameMusicEmu, LazyUSF and supporting looped songs natively, ANMP is esp. suited to play various audio formats from video games. Moreover it supports Looped Midi Tracks.

Main Features:

  • muting multichannel audio files
  • gapless playback (for most streamed audio formats)
  • arbitrary (forward) looping of songs (i.e. even nested loops)
  • synthesizing MIDI files using fluidsynth
  • multi-channel audio rendering
  • overlapping notes do not kill each other
  • unrolling looped MIDI tracks
  • volume response compatible to N64’s software synth
  • supports an IIR lowpass filter compatible to Rareware‘s N64 games (Conkers Bad Fur Day, Jet Force Gemini)
  • support for cue sheets
  • support for audio normalization based on EBU R 128
  • exposes basic commands via D-Bus

aimp

github.com/derselbst/ANMP

LASTAR ~ Batch File Splitting & Normalization


LASTAR is a batch (non-interactive) audio processor for loudness adjustment and file splitting of a batch of audio recordings, using an audio transparent maximizer.

At the opposite of usual available software, loudness normalization is done on signal power, which leads to a louder and more homogeneous result than the usual “peak” normalization, in particular on live recordings.

Its purpose is:

  • to split, equalize and normalize digitized analog tapes or vinyl
  • to split, equalize and normalize live recordings from microphones (ex ZOOM H2 recordings)
  • fast and homogeneous normalization of a group of files (compilation for instance)
  • loudness and dynamic reduction for listening in a noisy environment (car…)
  • and so on…

This software aims to be very fast and easy to use : the most efficient computing techniques have been implemented, and there are very few parameters to set (most of them are automatically adjusted by analyzing the file).

It embeds:

  • an automatic RMS normalizer,
  • an Automatic Gain Controller (compressor)
  • a 10 bands automatic equalizer
  • a 3 bands semi-parametric equalizer
  • a file splitter
  • a noise gate
  • preset management
  • a preview function.

www.arthelion.com/index.php/en/windows-en/lastar
sourceforge.net/projects/lastar

WaveShop ~ Windows Audio Editor


WaveShop is a free, open-source audio editor for Windows XP/Vista/7/8 32-bit and 64-bit. WaveShop is fast, lightweight, and bit-perfect, meaning samples aren’t altered unless they need to be. Editing a portion of an audio file only affects that portion; the rest of the file is untouched. Blocks of audio can be cut and pasted without changing their contents at all. This is especially useful for patching a finished master without corrupting its dither. Waveshop’s features include peak, RMS and spectral analysis, normalizing, fading, sample rate conversion, audio generation, plug-ins, and more, all with unlimited undo and comprehensive help.

waveshop-main-window

waveshop.sourceforge.net/
sourceforge.net/projects/waveshop/

loudgain ~ Linux Loudness Normalizer


loudgain is a versatile ReplayGain 2.0 loudness normalizer, based on the EBU R128 / ITU BS.1770 standard (-18 LUFS) and supports FLAC / Ogg / MP2 / MP3 / MP4 / M4A / ALAC & Opus audio files. It uses the well-known mp3gain command line syntax but will never modify the actual audio data.

Just what you ever wanted: the best of mp3gain, ReplayGain 2.0 and Linux combined. Spread the word!

github.com/Moonbase59/loudgain

TAudioConverter ~ Audio Converter & Extractor


TAudioConverter is a audio converter and extractor. It can convert almost any audio format to aac, ac3, mp3, flac, wav, ogg, opus or mpc. SoX is used to apply effects such as normalization, volume change etc. It can also extract audio streams from video files and encode them. Video files with more than one audio stream is supported. It is possible to run up to 8 simultaneous processes to shorten encoding time.

TAudioConverter

ozok.users.sourceforge.net
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