Free Open Source Audio Conversion Software

Create data discs with advanced data settings. Create video and audio discs, Burn will convert if needed. Create DVD-Video discs. Recreate discs. Burn doesn't reinvent the wheel, it uses many powerful open source Unix utilities and is also open source.

SoX is the Swiss Army Knife of sound processing utilities. It can convert audio files to other popular audio file types and also apply sound effects and filters during the conversion.

LAME is an educational tool to be used for learning about MP3 encoding. The goal of the LAME project is to improve the psycho acoustics, quality and speed of MP3 encoding. Note: we provide source code only!

FAAC is an Advanced Audio Coder (MPEG2-AAC, MPEG4-AAC). The goal of FAAC is to explore the possibilities of AAC and exceed the quality of the currently best MP3 encoders. You can submit pull requests via GitHub: https://github.com/knik0/faac https://github.com/knik0/faad2

MAD is a high-quality fixed-point MPEG audio decoder with 24-bit output. The implementation is entirely new, based on the ISO/IEC standards, and performs especially well on systems without native floating-point support.

Usage: - Install the Audible Manager and open a file of your account. - Sign in into your audible account (in the application). Now the program can convert your audible files. - Start Aax2Mp3 (AaxToMp3GUI.exe) and select the file you want to convert. - Click the Convert button. - The buttons are grayed out and it shows how far the conversion has progressed. - - The time should always continue to increase. - If the buttons are clickable again, the conversion is completed. The mp3 file is now to be found in the same folder as the AAX file. The conversion need (depending on the computing power) for 1 minute of audio approximately 2 seconds. The conversion of Aax to MP3 is fast, free, and requires no I-Tunes. Have fun while listening! If you want to help developing this software, please contact me. I do not have enough time to develop alone.

fre:ac is a free audio converter and CD ripper for various formats and encoders. It features MP3, MP4/M4A, WMA, Ogg Vorbis, FLAC, AAC, and Bonk format support, integrates freedb/CDDB, CDText and ID3v2 tagging and is available in several languages.

The SoX Resampler library `libsoxr' performs one-dimensional sample-rate conversion—it may be used, for example, to resample PCM-encoded audio. It aims to give fast and high quality results for any constant (rational or irrational) resampling ratio. Phase-response, preserved bandwidth, aliasing, and rejection level parameters are all configurable; alternatively, simple `preset' configurations may be selected. An experimental, variable-rate resampling mode of operation is also included.

FFmpeg Batch AV Converter is a free universal audio and video encoder, that allows to use the full potential of ffmpeg command line with a few mouse clicks in a convenient GUI with drag and drop, progress information. Some fancy wizards make things easy for non-experts. Thanks to its multi-file encoding feature, it may be the fastest a/v batch encoder available, since it maximizes system resources usage by launching as many simultaneous processes up to user cpu thread count. You can change encoding priority, pause and resume, set automatic shutdown. It is good for seasoned ffmpeg users as well as beginners. It provides unlimited single or multi-file batch encoding for almost any audio/video format. You can use any set of parameters and try them before starting encoding. You can manipulate and multiplex streams, batch subtitle videos (as track and hardcoded), trim, concatenate, record screen, capture M3u8 or YouTube URLs. You can also access useful multimedia file information.

mp3splt is a free utility to split mp3, ogg vorbis and FLAC files without decoding, selecting begin/end time; if file is an album, you can get splitpoints automatically from internet or a local cue, cddb file. It also splits Mp3Wrap and AlbumWrap archives. Supports splitting and trimming by silence detection. For mp3 files, both ID3v1 and ID3v2 are supported using original tags or user defined tags.

FLAC is a free lossless compressed audio format which supports streaming and archival. The FLAC project maintains the format and provides a reference encoder/decoder and input plugins for several popular audio players.

Lossless compressor for multichannel 8,16 and 24 bits audio data, with the ability of password data protection. Being 'lossless' means that no data/quality is lost in the compression - when uncompressed, the data will be identical to the original.

FlacSquisher converts a library of Flac files to MP3, Opus, or Ogg Vorbis format, maintaining the directory structure in the original library. That way, you can maintain your Flac files for home listening, and easily convert them to MP3, Opus, or Ogg format for mobile use, where storage space is more often tightly constrained.

LameXP is a free multi-format audio file converter that supports a variety of output formats, including MP3, AAC/MP4, Ogg Vorbis, Opus, as well as FLAC, and an even higher number of input formats. It also supports batch processing and can utilize multiple processor cores.

Dragon UnPACKer is a game archive (Quake PAK, etc..) unpacking tool. It is plugin based making easier to add new archive file formats. It has convert ability and raw search function for known material (audio, video and pictures).

The Bauer stereophonic-to-binaural DSP (bs2b) library and plugins is designed to improve headphone listening of stereo audio records. Recommended for headphone prolonged listening to disable superstereo fatigue without essential distortions.

winLAME is an easy to use encoder for many audio formats, including MP3, Opus, Ogg Vorbis and more. winLAME lets you read in audio tracks from CDs or encode audio files from your hard drive. The intuitive wizard-style user interface makes it easy to set up encoding settings. Multicore CPUs are used to parallelize encoding tasks. Official GitHub project webpage: https://github.com/vividos/winLAME

HEVC was developed with the goal of providing twice the compression efficiency of the previous standard, H.264 / AVC. Although compression efficiency results vary depending on the type of content and the encoder settings, at typical consumer video distribution bit rates HEVC is typically able to compress video twice as efficiently as AVC. End-users can take advantage of improved compression efficiency in one of two ways (or some combination of both); IFME helps you to organized, manage queue, manage plugins, and choose compiled x265 and more! Support AviSynth natively with build-in awesome library like InterFrame & Script Editor

Media Converter is a plugin based video and audio converter. It uses FFmpeg as its engine, which allows it to convert to a lot of formats.

Multimedia architecture consisting of libraries and applications. Libraries include gavl for low level audio/video support and gmerlin_avdecoder, a multiformat decoding library. Applications include a GUI player with mediatree and a GUI transcoder.

As Speek's celebrated FLAC Frontend hasn't been updated for 10 years, things started to get a little buggy with new Windows versions. This remake fixes most issues. FLAC Frontend is a convenient way for Windows users not used to working with command lines to use the official FLAC tools. It accepts WAVE, W64, AIFF and RAW files for encoding and outputs FLAC or OGG-FLAC files. It is able to decode FLAC files, test them, fingerprint them and re-encode them. It has drag-and-drop support too.

Exports Rocksmith 2014 arrangements to Guitar Pro tabs (.gp5, .gpx). It parses Rocksmith's .psarc archives and converts any songs found within to tablature. All the Rocksmith techniques are supported and translated, and you get all of the chord diagrams and fingering hints that Rocksmith shows you. It's a great way to figure out specific parts of a song, and Guitar Pro's speed trainer is a very good alternative to the riff repeater. All arrangement types (bass, lead, rhythm) can be exported, either together in a single tab or into separate tabs.

Fast batch audio processor for automatic loudness adjustment and audio files splitting. The software automatically normalizes a batch of files to a given RMS volume, and can split tracks in several ones by detecting low level ("noise") chunks. It embeds : - a multi-band limiter, - an Automatic Gain Control, - an automated 10 bands equalizer, - a classic 3 bands semi-parametric equalizer, - a noise gate, - a trimming function, - user preset management. Portable version needs VC 2015 RunTime to be installed https://www.microsoft.com/fr-fr/download/details.aspx?id=48145 You use LASTAR for your mobile music library ? Try now the new Android player with its real-time embedded algorithm ! https://play.google.com/store/apps/details?id=com.arthelion.loudplayer&hl=fr

LSP (Linux Studio Plugins) is a collection of open-source plugins currently compatible with LADSPA, LV2 and LinuxVST formats. Standalone plugins for JACK are provided since version 1.0.8. Experimental support of ARMv7 added since version 1.1.4 Experimental support of AArch64 added since version 1.1.9 The basic idea is to fill the lack of good and useful plugins under the GNU/Linux platform. After some contributions to other open source projects the decision was made to implement separate and independent plugin distribution.