WavPack is a free, open source lossless audio compression format developed by David Bryant.
WavPack allows users to compress (and restore) both 16 and 24-bit audio files in the .WAV format. It also supports multichannel streams and high frequency sampling rates. Like other lossless compression schemes the data reduction varies with the source, but it is generally between 25% and 50% for typical popular music and somewhat better than that for classical music and other sources with greater dynamic range.
WavPack also incorporates a unique "hybrid" mode that 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!
- Fast and efficient encoding and decoding
- Open source, released under a BSDish license
- Error robustness
- Streaming support
- Supports multichannel audio and high resolutions
- Hybrid/lossy mode
- Tagging support (ID3v1, APE)
- Supports RIFF chunks
- Supports embedded CUE sheets
- Includes MD5 hashes for quick integrity checking
- Ability to create self extracting files for Win32 platform
- ReplayGain compatible
David Bryant started development on WavPack in mid-1998, with the release of version 1.0. This first version compressed and decompressed audio losslessly, nothing else, but by then it already featured one of the best efficiency vs. speed ratio among lossless encoders.
Very soon after the release of version 1.0, Bryant released v. 2.0, which featured lossy encoding (using only quantization for data reduction - no psychoacoustic process was applied to the stream).
In 1999, the developer released version 3.0, which featured novelties such as a fast mode (with reduced compression ratio), compression of RAW files and error detection using CRC checksums.
WavPack development is still going on, and a major feature added in late 3.x versions is the hybrid mode, where the encoder generates a lossy file + a correction file, so that both can be decompressed back to the original PCM stream.
WavPack 4 has been recently released. It included important changes, such as fast seeking, multichannel support, high resolution audio support, etc. turning it into one of the most full featured and modern lossless audio compressors.
- NullSoft Winamp (plugin w/ ReplayGain & Media Library support) and Winamp-compatible players
- foobar2000 Advanced Audio Player (official encoding/decoding addon, w/ ReplayGain & Cuesheets support)
- VUPlayer (official plugin, supports encoding)
- Windows Media Player and other directshow-based players (MPC, TCMP, RadLight) (with CoreWavPack directshow filter)
- Apollo Audio Player (plugin w/ ReplayGain support)
- XMplay (official plugin)
(Several players, like foobar2000 and VUplayer, can also convert from other formats to WavPack)
- dBpowerAMP Music Converter / Audio Player / CD Writer (official addon)
- GX:Transcoder Music converter
- Adobe Audition and CoolEdit (filter w/ 32-bit floats & extra info save support)
- Mr. QuestionMan
- Audio Identifier
- mkvtoolnix - tool to multiplex WavPack streams inside the Matroska container
It's worth mentioning the Matroska guys decided to concentrate on WavPack as the lossless compressor of choice for their container. Quite an honor... :-)
To ensure high-speed operation, WavPack uses a very simple predictor that is implemented entirely in integer math. In its "fast" mode the prediction is simply the arithmetic extrapolation of the previous two samples. For example, if the previous two samples were -10 and 20, then the prediction would be 50. For the default mode a simple adaptive factor is added to weigh the influence of the earlier sample on the prediction. In our example the resulting prediction could then vary between 20 for no influence to 50 for full influence. This weight factor is constantly updated based on the audio data's changing spectral characteristics, which is why it is called "adaptive".
The prediction generated is then subtracted from the actual sample to be encoded to generate the error value. In mono mode this value is sent directly to the coder. However, stereo signals tend to have some correlation between the two channels that can be further exploited. Therefore, two error values are calculated that represent the difference and average of the left and right error values. In the "fast" mode of operation these two new values are simply sent to the coder instead of the left and right values. In the default mode, the difference value is always sent to the coder along with one of the other three values (average, left, or right). An adaptive algorithm continuously determines the most efficient of the three to send based on the changing balance of the channels.
The developer has developed a unique data encoder for WavPack that he believes is better than Rice coding in two different areas. It is impossible to encode more efficiently than Rice coding because it represents the optimal bit coding (sometimes known as the Huffman code) for this type of data. WavPack's encoder is slightly less efficient than this, but only by about 0.15 bits/sample (or less than 1% for 16-bit data). The first advantage of WavPack's coder is that it does not require the data to be buffered ahead of encoding, instead it converts each sample directly to bitcodes. This is more computationally efficient and it is better in some applications where coding delay is critical. The second advantage is that it is easily adaptable to lossy encoding because all significant bits (except the implied "one" MSB) are transmitted directly. In this way it is possible to only transmit, for example, the 3 most significant bits (with sign) of each sample. In fact, it is possible to transmit only the sign and implied MSB for each sample with an average of only 3.65 bits/sample.
This coding scheme is used to implement the "lossy" mode of WavPack. In the "fast" mode the output of the non-adaptive decorrelator is simply rounded to the nearest codable value for the specified number of bits. In the default mode the adaptive decorrelator is used (which reduces the average noise about 1 dB) and also both the current and the next sample are considered in choosing the better of the two available codes (which reduces noise another 1 dB).
The developer has decided to not use any floating-point arithmetic in WavPack's data path because he believes that integer operations are less susceptible to subtle chip to chip variations that could corrupt the lossless nature of the compression, the Pentium floating point bug being a blatant example of this. It is possible that a lossless compressor that used floating-point math could generate different output when running on that faulty Pentium. Even disregarding actual bugs, floating-point math is complicated enough that there could be subtle differences between "correct" implementations that could cause trouble for this type of application. To further ensure confidence in the integrity of WavPack's compression, the encoder includes a 32-bit error detection code to the generated streams.