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.
You have the need for professional and state-of-the-art audio plug-ins – but don’t have thousands of bucks to spend on it? Then we believe you will love Calf Studio Gear! Focused on high-quality sound processing and a highly usable interface Calf Studio Gear is designed to give you a professional production environment for your open source operating system. Play your SF2 sample banks, create filthy organs, fatten your sounds with phasers, delays, reverbs and other FX, process your recordings with gates, compressors, deesser and finally master your stuff with multiband dynamics – for free! Calf Studio Gear is available exclusively for LINUX-based operating systems and runs as a stand-alone effect rack connectable through Jack sound server or as plug-ins in every audio host that is able to fire up LV2 compliant devices, e.g. the highly recommended Ardour Audio Workstation
Our RoughRider compressor is one of the most popular dynamics processors on the planet, with well over a half a million downloads over its decade-plus lifespan, and is in heavy daily use by producers the world over. With RoughRider 3, we’ve expanded the original to include an external sidechain input, the ability to turn off the built-in “warming” filter (the FULL BANDWIDTH button), and much more accurate metering.
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, 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.
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).
WavPack is a completely open audio compression format providing lossless, high-quality lossy, and a unique hybrid compression mode. For version 5.0.0, several new file formats and lossless DSD audio compression were added, making WavPack a universal audio archiving solution.
In the default lossless mode WavPack acts just like a WinZip compressor for audio files. However, unlike MP3 or WMA encoding which can affect the sound quality, not a single bit of the original information is lost, so there’s no chance of degradation. This makes lossless mode ideal for archiving audio material or any other situation where quality is paramount. The compression ratio depends on the source material, but generally is between 30% and 70%.
The hybrid mode provides all the advantages of lossless compression with an additional bonus. Instead of creating a single file, this mode creates both a relatively small, high-quality lossy file that can be used all by itself, and a “correction” file that (when combined with the lossy file) provides full lossless restoration. For some users this means never having to choose between lossless and lossy compression!
WavPack employs only well known, public domain techniques (i.e., linear prediction with LMS adaptation, Elias and Golomb codes) in its implementation. Methods and algorithms that have ever been patented (e.g., arithmetic coding, LZW compression) are specifically avoided. This ensures that WavPack encoders and decoders will remain open and royalty-free.