FSLAC ~ Free Semi-Lossless Audio Codec

constrained VBR (CVBR) version of the publicly available open-source lossless audio coder FLAC.

FLAC, being a mathematically lossless audio codec, inevitably creates VBR streams as compressed files. Depending on the «difficulty» of coding each segment of the audio signal, the instantaneous coding bit-rate can be quite high. However, one can observe that, during passages of high FLAC bit-rate, the coded audio also exhibits the greatest ability of psychoacoustic masking. FSLAC exploits this property to limit the maximum instantaneous bit-rate of the compressed file. It does so by detecting the difficult audio blocks (by measuring their predictability via linear-prediction error energy calculations) and requantizing each of the detected blocks to a lower bit-depth, thereby reducing the bit-rate needed for lossless coding of that block. To prevent the quantization error from becoming audible (or visible in a spectrogram), simple adaptive noise shaping is used.

This approach is similar to the one used by LossyWAV, but differs in two important aspects. First, FSLAC is not a stand-alone pre-processor but instead is coupled with a FLAC encoder and, hence, directly creates FLAC compatible compressed files. Second, FSLAC only alters the high-bit-rate audio segments, not (almost) all parts of the audio input as LossyWAV does. The coded audio, therefore, remains perceptually lossless. In addition, it is worth noting that, due to its simplicity, FSLAC encoding is very fast. All of these features make FSLAC attractive for audio production and archival applications.


fdkaac ~ Command Line Frontend For libfdk-aac Encoder

dkaac reads linear PCM audio in either WAV, raw PCM, or CAF format,
and encodes it into either M4A / AAC file.

If the input file is "-", data is read from stdin. Likewise, if the
output file is "-", data is written to stdout if one of streamable AAC
transport formats are selected by **-f**.

When CAF input and M4A output is used, tags in CAF file are copied into
the resulting M4A.


TAK ~ Tom’s Lossless Audio Kompressor

TAK stands for (T)om’s lossless (A)audio (k)ompressor. Besides, it’s a throwback to a (not very philanthropic) character from Stephen King’s “Regulators”. Early semi-public evaluation versions operated under the working title YALAC .

  • High compression . The strongest mode is on a par with Monkey’s Audio High and OptimFrog’s Normal; for specific files such as classical music or voice recordings, it often outperforms both. This classification is based on the evaluation of hundreds of files of various styles; it definitely does not apply to every single file.
  • High compression speed . I am currently not aware of any other compressor that works faster than TAK’s Turbo or Fast mode and achieves similar compression rates.
  • Multi-core compressor . The compressor optionally generates up to four threads in order to take advantage of multi-core cpus.
  • Very high decompression speed . It is at the level of FLAC and therefore significantly higher than most symmetrical compressors.
  • Support for every popular audio format (not yet fully implemented).
  • Streaming support . An info frame, which contains all the information required for decoding, is inserted into the compressed audio data every 2 seconds.
  • Fault tolerance . A single bit error never damages the audio data for more than a maximum of 250 ms, since it is stored in completely independent frames of a maximum of this duration. The decoder processes even extremely damaged files, optionally replacing or removing the affected data with silence.
  • Error detection . Every single frame is protected by a 24-bit checksum (CRC).
  • MD5 checksums for quick identification of audio material (e.g. for searching for duplicates).
  • Fast, sample-accurate access to any playback position . The file header contains a look-up table with index positions every second. Even without this table, efficient random access is possible; for this purpose, the synchronization codes of the frame headers and / or the offset values ​​optionally recorded in the frame header, which refer to the beginning of the previous and next frame, can be used.
  • Metadata . A flexible and expandable structure allows the recording of non-audio data such as images or cuesheets.
  • Playback plugins for Winamp and Foobar are currently available.
  • An SDK provides other developers with decoding functions for integration into their applications. An extension to include coding functions is planned.


dsfTAKSource ~ TAK  DirectShow  Source  Filter
  • Playback TAK audio files in any DirectShow Player (Windows Media Player, MediaPlayerClassic, …)
  • Now supporting TAK 2.2.0 (multi-channel audio, …)
  • Support UNICODE filenames and Sample Rates > 44.1 KHz
  • Upgrade: now correctly works in Windows 7 64bit !  (and other 64bit Windows versions)


Monkey’s Audio ~ Free Lossless CODEC

Monkey’s Audio is a fast and easy way to compress digital music.  Unlike traditional methods such as mp3, ogg, or wma that permanently discard quality to save space, Monkey’s Audio only makes perfect, bit-for-bit copies of your music.  That means it always sounds perfect – exactly the same as the original.  Even though the sound is perfect, it still saves a lot of space (think of it as a beefed-up Winzip™ your music).  The other great thing is that you can always decompress your Monkey’s Audio files back to the exact, original files.  That way, you’ll never have to recopy your CD collection to switch formats, and you’ll always be able to perfectly recreate the original music CD.







exhale ~ MPEG-4 Audio Encoder

exhale, which is an acronym for “Ecodis eXtended High-efficiency And Low-complexity Encoder”, is a lightweight library and application to encode uncompressed WAVE-format audio files into MPEG-4 format files complying with the ISO/IEC 23003-3 (MPEG-D) Unified Speech and Audio Coding (USAC, also known as Extended High-Efficiency AAC) standard. In addition, exhale writes program peak-level and loudness data into the generated MPEG-4 files according to the ISO/IEC 23003-4, Dynamic Range Control (DRC) specification for use by decoders providing DRC. exhale currently makes use of all frequency-domain (FD) coding tools in the scale factor based MDCT processing path, except for predictive joint stereo, which is still being integrated. Its objective is high quality mono, stereo, and multichannel coding at medium and high bit rates, so the lower-rate USAC coding tools (ACELP, TCX, Enhanced SBR and MPEG Surround with Unified Stereo coding) won’t be integrated.


opencore-amr ~ Android Audio CODECS

Library of OpenCORE Framework implementation of Adaptive Multi Rate Narrowband and Wideband (AMR-NB and AMR-WB) speech codec. Library of VisualOn implementation of Adaptive Multi Rate Wideband (AMR-WB) encoder and Advanced Audio Coding (AAC) encoder. Modified library of Fraunhofer AAC decoder and encoder.


FLAC Frontend ~ Lossless Encoding GUI

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. It is tested on Windows XP SP3 and Windows 7, but should work with Windows XP SP2 or newer. It requires .NET 2.0 or later.