A 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.
EdgeSounds’ RatHole (former GenieSys RatHole) is a free unique nondestructive universal compression utility. Its function is based on a principle of self-training neural networks. EdgeSounds RatHole was especially designed for nondestructive compression of any files containing audio data in PCM 8/16/24 bit or IEEE_FLOAT 32 bit format.
A new EdgeSounds compression algorithm makes it possible to efficiently reduce the size of packed audio data and later unpack exactly same bits, with no difference to the original data. The compression algorithm compresses audio data, considering the bit depth of the digital data contained in the audio file (8/16/24/32 bit). The algorithm is proven to be equally effective on compressing the following file types:
Audio files WAV, AU, AIFF, SND;
Ensoniq Paris PAS;
Sample sound banks such as:
AVM Apex bank file APEX
Aureal sound bank ARL
Creative Labs/E-mu System SoundFont banks format 1&2.x SBK, SF2;
Downloadable sounds level 1&2 DLS;
Ensoniq EPS Files EFE, Ensoniq Instruments family files INS, Ensoniq Disk Image GKH;
Gravis Ultrasound /Forte Patch PAT;
Kurzweil 2000-2600 files KRZ;
Seer Systems Reality banks SEERBANKSET;
Turtle Beach WaveFront Bank WFB;
Virtual Sampler bank VSB;
Multi-track audio data as Cakewalk/Sonar BUN, CWB;
Impulse Tracker instrument ITI, Fast Tracker 2 instrument XI;
any other formats containing audio data
The compression ratio of the algorithm depends on the size of audio data, the balance between the tone and noise component, bit depth and other factors, and usually varies from 36% to 78% or even more, with an average of 48-56%. The higher is the bit depth and the fidelity of audio data, the better is the compression ratio.
The RatHole can be successfully used as a common archiving utility for any other file types as well.
The swiss army knife of lossless video / audio editing
LosslessCut aims to be the ultimate cross platform FFmpeg GUI for extremely fast and lossless operations on video, audio, subtitle and other related media files. The main feature is lossless trimming and cutting of video and audio files, which is great for saving space by rough-cutting your large video files taken from a video camera, GoPro, drone, etc. It lets you quickly extract the good parts from your videos and discard many gigabytes of data without doing a slow re-encode and thereby losing quality. Or you can add a music or subtitle track to your video without needing to encode. Everything is extremely fast because it does an almost direct data copy, fueled by the awesome FFmpeg which does all the grunt work.
Lossless cutting of most video and audio formats
Losslessly cut out parts of video / audio (for cutting away commercials etc.)
Losslessly rearrange the order of video / audio segments
Lossless merge / concatenation of arbitrary files (with identical codecs parameters, e.g. from the same camera)
Lossless stream editing: Combine arbitrary tracks from multiple files (ex. add music or subtitle track to a video file)
Losslessly extract all tracks from a file (extract video, audio, subtitle, attachments and other tracks from one file into separate files)
Batch view for fast multi-file workflow
Remux into any compatible output format
Take full-resolution snapshots from videos in JPEG / PNG format
Manual input of cutpoint times
Apply a per-file timecode offset (and auto load timecode from file)
Change rotation / orientation metadata in videos
View technical data about all streams
Timeline zoom and frame / keyframe jumping for accurate cutting around keyframes
Saves per project cut segments to project file
View FFmpeg last command log so you can modify and re-run recent commands on the command line
Give labels to cut segments
View segment details, export / import cut segments as CSV
Import segments from: MP4 / MKV chapters, Text file, YouTube, CSV, CUE, XML (DaVinci, Final Cut Pro)
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.
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.
Internet music dealers currently sell “CD-Quality” tracks, or even better (“Studio-Master”), thanks to lossless audio coding formats (FLAC, ALAC). However, a lossless format does not guarantee that the audio content is what it seems to be. The audio signal may have been upscaled (increasing the resolution), upsampled (increasing the sampling rate) or even transcoded from a lossy to a lossless format. Lossless Audio Checker analyzes lossless audio tracks and detects upscaling, upsampling and transcoding.
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.
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.
FLAC stands for Free Lossless Audio Codec, an audio format similar to MP3, but lossless, meaning that audio is compressed in FLAC without any loss in quality. This is similar to how Zip works, except with FLAC you will get much better compression because it is designed specifically for audio, and you can play back compressed FLAC files in your favorite player (or your car or home stereo) just like you would a MP3 file.
FLAC stands out as the fastest and most widely supported lossless audio codec, and the only one that at once is non-proprietary, is unencumbered by patents, has an open-source reference implementation, has a well documented format and API and has several other independent implementations.