Audiodope is an audio editor. You can load and listen to music files of various formats such as wave, MP3, WMA, etc. You can also edit part or the whole stream with functions like copy, cut, paste, insert and delete. You may modify any part of the stream by applying audio effects and synthesize sound files.
Sound editing functions such as copy, cut, paste, delete, insert, trim.
WaveGain is an application of the ReplayGain algorithms to standard PCM wave files. Calculated gain adjustments are applied directly to the audio data, instead of just writing metadata as traditionally done for other formats like MP3, FLAC and Ogg Vorbis. The replaygain values can also be added as metadata in a custom RIFF chunk named ‘gain’. This could theoretically allow WAV files to have same lossless functionality as other formats where audio data is not altered. But since no current players are aware of this “standard”, the metadata is used only by WaveGain for the “–undo-gain” feature, which is lossy.
NorQualizer is a smart audio equalizer / normalizer which corrects audio files to get them all similar in terms of bandwidth and volume level. Especially useful for preparing audio CDs with files from different sources.
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.
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.
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.
Dynamic Audio Normalizer is a library for advancedaudio 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 C99Microsoft.NET, Java, Python and Pascal are provided.
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.
muting multichannel audio files
gapless playback (for most streamed audio formats)
arbitrary (forward) looping of songs (i.e. even nested loops)
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).