WavPack: Difference between revisions
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'''WavPack''' is a free, open source [[lossless]] audio compression format developed by David Bryant. | '''WavPack''' is a free, open source [[lossless]] audio compression format developed by David Bryant. | ||
== Description == | |||
==Description== | |||
WavPack (pronounced "wave-pack") allows users to compress (and restore) all [[PCM]] audio formats including 8, 16, and 24-bit ints; 32-bit floats; [[mono]], [[stereo]], and [[multichannel]]; [[sampling rate]]s from 6 to 192 kHz. 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 (pronounced "wave-pack") allows users to compress (and restore) all [[PCM]] audio formats including 8, 16, and 24-bit ints; 32-bit floats; [[mono]], [[stereo]], and [[multichannel]]; [[sampling rate]]s from 6 to 192 kHz. 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! | 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! | ||
==Feature Summary== | == Feature Summary == | ||
* Fast and efficient encoding and decoding | * Fast and efficient encoding and decoding | ||
* [[Open source]], released under a BSDish license | * [[Open source]], released under a BSDish license | ||
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* [[ReplayGain]] compatible | * [[ReplayGain]] compatible | ||
== History == | |||
==History== | |||
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. | 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. | ||
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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. | 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. | ||
== Software support == | |||
==Software support== | === Players === | ||
===Players=== | |||
* NullSoft [[Winamp]] (plugin w/ ReplayGain & Media Library support) and Winamp-compatible players | * NullSoft [[Winamp]] (plugin w/ ReplayGain & Media Library support) and Winamp-compatible players | ||
* [[foobar2000]] Advanced Audio Player (official encoding/decoding addon, w/ ReplayGain & Cuesheets support) | * [[foobar2000]] Advanced Audio Player (official encoding/decoding addon, w/ ReplayGain & Cuesheets support) | ||
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* [http://fondriest.frederic.free.fr/realisations/lamip/ LAMIP] (official plugin) | * [http://fondriest.frederic.free.fr/realisations/lamip/ LAMIP] (official plugin) | ||
* [http://mpxplay.net/ MPXplay] for DOS! | * [http://mpxplay.net/ MPXplay] for DOS! | ||
* [http://aqualung.sourceforge.net/ Aqualung] for GNU/Linux | |||
=== Frontends === | |||
===Frontends=== | |||
* Custom [http://members.home.nl/w.speek/wavpack.htm Windows Frontend] (by Speek) | * Custom [http://members.home.nl/w.speek/wavpack.htm Windows Frontend] (by Speek) | ||
* [http://www.unifront.boereck.de/ UniversalFront] by Böreck | * [http://www.unifront.boereck.de/ UniversalFront] by Böreck | ||
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* [http://www.webearce.com.ar/ MAREO] by Kwambis | * [http://www.webearce.com.ar/ MAREO] by Kwambis | ||
===Converters=== | === Converters === | ||
'''Note:''' ''Several players, like foobar2000 and VUplayer, can also convert from other formats to WavPack)'' | '''Note:''' ''Several players, like foobar2000 and VUplayer, can also convert from other formats to WavPack)'' | ||
* [http://www.dbpoweramp.com/ dBpowerAMP] Music Converter / Audio Player / CD Writer (official addon) | * [http://www.dbpoweramp.com/ dBpowerAMP] Music Converter / Audio Player / CD Writer (official addon) | ||
* [http://www.board-24.de/ GX:Transcoder] Music converter | * [http://www.board-24.de/ GX:Transcoder] Music converter | ||
=== Editors === | |||
===Editors=== | |||
* [[Adobe Audition]] and Cool Edit (filter w/ 32-bit floats & extra info save support) | * [[Adobe Audition]] and Cool Edit (filter w/ 32-bit floats & extra info save support) | ||
=== CD writers/rippers === | |||
===CD writers/rippers=== | |||
* [http://www.ahead.de Ahead Nero Burning Rom] | * [http://www.ahead.de Ahead Nero Burning Rom] | ||
* [http://www.burrrn.net Burrrn] Audio CD burner | * [http://www.burrrn.net Burrrn] Audio CD burner | ||
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* [http://cdexos.sourceforge.net CDex] CD ripper | * [http://cdexos.sourceforge.net CDex] CD ripper | ||
=== Taggers === | |||
===Taggers=== | |||
* [http://www.mp3tag.de/en/index.html Mp3tag] Universal Tag Editor | * [http://www.mp3tag.de/en/index.html Mp3tag] Universal Tag Editor | ||
* [http://users.otenet.gr/~jtcliper/tgf/ The GodFather] Tagger / Music manager | * [http://users.otenet.gr/~jtcliper/tgf/ The GodFather] Tagger / Music manager | ||
* [[Tag.exe|Case's Tag]] command line tagger | * [[Tag.exe|Case's Tag]] command line tagger | ||
=== Other tools === | |||
===Other tools=== | |||
* [http://www.burrrn.net/mrq/ Mr. QuestionMan] | * [http://www.burrrn.net/mrq/ Mr. QuestionMan] | ||
* [http://www.bitattack.ro/ai/ Audio Identifier] | * [http://www.bitattack.ro/ai/ Audio Identifier] | ||
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lossless compressor of choice for their container. Quite an honor... :-) | lossless compressor of choice for their container. Quite an honor... :-) | ||
== Hardware Support == | |||
==Hardware Support== | |||
* iRiver iHP-120/iHP-140 with the open source [http://www.rockbox.org Rockbox firmware] | * iRiver iHP-120/iHP-140 with the open source [http://www.rockbox.org Rockbox firmware] | ||
* [http://www.rokulabs.com/products/photobridge/features.php Roku PhotoBridge HD] (w/ [http://www.wavpack.com/downloads.html plugin]) | * [http://www.rokulabs.com/products/photobridge/features.php Roku PhotoBridge HD] (w/ [http://www.wavpack.com/downloads.html plugin]) | ||
==Technology description== | == Technology description == | ||
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". | 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". | ||
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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. | 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. | ||
WavPack source code is very portable. It has been compiled on several Unices (Linux, Mac OS X, Solaris, FreeBSD, OpenBSD, NetBSD, Compaq Tru64, HP-UX...) as well as Windows, DOS and OpenVMS. It works on architectures such as x86, ARM, PowerPC, SPARC, DEC Alpha, PA-RISC, MIPS, Motorola 68k... | WavPack source code is very portable. It has been compiled on several Unices (Linux, Mac OS X, Solaris, FreeBSD, OpenBSD, NetBSD, Compaq Tru64, HP-UX...) as well as Windows, DOS and OpenVMS. It works on architectures such as x86, ARM, PowerPC, SPARC, DEC Alpha, PA-RISC, MIPS, Motorola 68k... | ||
==External links== | == External links == | ||
* [http://www.wavpack.com/ Official website] | * [http://www.wavpack.com/ Official website] | ||
* [http://www.rarewares.org/lossless.html Unofficial multiplatform versions] at RareWares | * [http://www.rarewares.org/lossless.html Unofficial multiplatform versions] at RareWares |
Revision as of 22:34, 22 May 2007
WavPack is a free, open source lossless audio compression format developed by David Bryant.
Description
WavPack (pronounced "wave-pack") allows users to compress (and restore) all PCM audio formats including 8, 16, and 24-bit ints; 32-bit floats; mono, stereo, and multichannel; sampling rates from 6 to 192 kHz. 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!
Feature Summary
- Fast and efficient encoding and decoding
- Open source, released under a BSDish license
- Multiplatform
- Hardware support
- 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
History
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.
Software support
Players
- 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)
- Cog Audio player for MacOS X.
- XMMS (with Kuniklo's plugin)
- LAMIP (official plugin)
- MPXplay for DOS!
- Aqualung for GNU/Linux
Frontends
- Custom Windows Frontend (by Speek)
- UniversalFront by Böreck
- Frontah by Madah
- MAREO by Kwambis
Converters
Note: 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
Editors
- Adobe Audition and Cool Edit (filter w/ 32-bit floats & extra info save support)
CD writers/rippers
- Ahead Nero Burning Rom
- Burrrn Audio CD burner
- Exact Audio Copy CD Ripper
- CDex CD ripper
Taggers
- Mp3tag Universal Tag Editor
- The GodFather Tagger / Music manager
- Case's Tag command line tagger
Other tools
- 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... :-)
Hardware Support
- iRiver iHP-120/iHP-140 with the open source Rockbox firmware
- Roku PhotoBridge HD (w/ plugin)
Technology description
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.
WavPack source code is very portable. It has been compiled on several Unices (Linux, Mac OS X, Solaris, FreeBSD, OpenBSD, NetBSD, Compaq Tru64, HP-UX...) as well as Windows, DOS and OpenVMS. It works on architectures such as x86, ARM, PowerPC, SPARC, DEC Alpha, PA-RISC, MIPS, Motorola 68k...
External links
- Official website
- Unofficial multiplatform versions at RareWares
- Historical versions at ReallyRareWares
- Lossless Codec Comparison by Rjamorim
- Configuring EAC and Wavpack