WavPack: Difference between revisions
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== 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! | ||
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* Includes MD5 hashes for quick integrity checking | * Includes MD5 hashes for quick integrity checking | ||
* Ability to create self extracting files for Win32 platform | * Ability to create self extracting files for Win32 platform | ||
* [[ | * [[Replay Gain]] 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 | 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 versus 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 | 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. | 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. | ||
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== Software support == | == Software support == | ||
=== Players === | === Players === | ||
* NullSoft [[Winamp]] (plugin | * NullSoft [[Winamp]] (plugin with Replay Gain & Media Library support) and Winamp-compatible players | ||
* [[foobar2000]] Advanced Audio Player (official encoding/decoding addon, | * [[foobar2000]] Advanced Audio Player (official encoding/decoding addon, with Replay Gain & Cuesheets support) | ||
* [http://www.vuplayer.com/vuplayer.htm VUPlayer] (official plugin, supports encoding) | * [http://www.vuplayer.com/vuplayer.htm VUPlayer] (official plugin, supports encoding) | ||
* [[Windows Media Player]] and other directshow-based players (MPC, TCMP, RadLight) (with [http://corewavpack.corecodec.org/ CoreWavPack] directshow filter) | * [[Windows Media Player]] and other directshow-based players (MPC, TCMP, RadLight) (with [http://corewavpack.corecodec.org/ CoreWavPack] directshow filter) | ||
* [http://koti.welho.com/hylinen/apollo/ Apollo] Audio Player (plugin | * [http://koti.welho.com/hylinen/apollo/ Apollo] Audio Player (plugin with Replay Gain support) | ||
* [http://www.un4seen.com/xmplay.html XMplay] (official plugin) | * [http://www.un4seen.com/xmplay.html XMplay] (official plugin) | ||
* [http://cogosx.sourceforge.net/ Cog] Audio player for MacOS X. | * [http://cogosx.sourceforge.net/ Cog] Audio player for MacOS X. | ||
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=== Editors === | === Editors === | ||
* [[Adobe Audition]] and Cool Edit (filter | * [[Adobe Audition]] and Cool Edit (filter with 32-bit floats & extra info save support) | ||
=== CD writers/rippers === | === CD writers/rippers === | ||
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* [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] | ||
* [http://www.bunkus.org/videotools/mkvtoolnix/ mkvtoolnix] | * [http://www.bunkus.org/videotools/mkvtoolnix/ 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 == | == 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] ( | * [http://www.rokulabs.com/products/photobridge/features.php Roku PhotoBridge HD] (with [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". | ||
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. | 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. | 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). | 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). | ||
Revision as of 19:53, 13 June 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
- Replay Gain 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 versus 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 with Replay Gain & Media Library support) and Winamp-compatible players
- foobar2000 Advanced Audio Player (official encoding/decoding addon, with Replay Gain & 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 with Replay Gain 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 with 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 (with 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