wavpack v 5.10 - verlustfreie Audiokompression

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wavpack v 5.10 - verlustfreie Audiokompression

Beitrag von RIP » 27.12.02, 10:07

wavpack v3.96 - verlustfreier audicodec

home: http://www.wavpack.com/

winamp plugin: http://www.wavpack.com/winamp_plugin.zip

wavpack: http://www.wavpack.com/wavpack.zip

frontend: http://home.wanadoo.nl/~w.speek/wavpack.htm

WavPack's Technology

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.

I have developed a unique data encoder for WavPack that I believe 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. My 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 my 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).

I have decided to not use any floating-point arithmetic in WavPack's data path because I believe that integer operations are less susceptible to subtle chip to chip variations that could corrupt the lossless nature of the compression, the recent 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, I have included a 32-bit error detection code.


Summary

To achieve very fast and reliable operation for lossless audio compression, WAVPACK utilizes only integer math and employs no pre-scanning of sample data (sample in --> bits out). A variable "lossy" compression mode is also implemented through the use of a novel encoding scheme having several advantages to the standard Rice method.
Zuletzt geändert von RIP am 31.07.04, 9:42, insgesamt 3-mal geändert.
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Beitrag von kraftian » 05.05.06, 20:41


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Beitrag von Kostenlos » 13.12.06, 20:45

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Beitrag von Stevie » 19.03.07, 18:54

WavPack 4.41 Beta 2

Changelog:
auf manchen Systemen etwas schneller

Download:
http://www.wavpack.com/wavpack-4.41.0-beta2.zip

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Beitrag von Stevie » 10.05.07, 18:21

WavPack 4.41 Final vom 6. Mai 2007

Download: http://www.wavpack.com/wavpack-4.41.zip

Changelog: http://www.wavpack.com/changelog.txt

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Re: wavpack v4.60.1 - verlustfreier Audiocodec

Beitrag von Gordon » 21.10.13, 18:50

What's new in WavPack version 4.70:

Transcoding from existing WavPack files (with tag copy)
Option to verify WavPack file integrity on creation (-v)
Use temporary files for safer overwriting
Detect UTF-16LE encoding for tag text files (mostly a Windows thing)
Transition to Visual Studio 2008 and 64-bit version (Windows)
Option to allow up to 16 MB APEv2 tag data (--allow-huge-tags)
Option for wvunpack to suppress audio decoding (useful for extracting only tags with -cc and -xx) (-n)
experimental, undocumented option to store float16 and float24 audio data as integers (--store-floats-as-ints)
Allow channel-order specification on WAV files with WAVEFORMATEXTENSIBLE headers that contain zeroed channel masks
Add --version and a new command (-f) for getting machine parsable info from wvunpack
Allow read-only access to APEv2 tags that come at the beginning of files
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Gordon
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Re: wavpack v4.70 - verlustfreier Audiocodec

Beitrag von Gordon » 09.12.16, 16:29

wegen des Vorziffernwechsels mal wieder ein Update
Release 5.0.0 - December 6, 2016
--------------------------------
added: multiple input formats, including RF64, Wave64, and CAF
added: lossless DSD audio in Philips DSDIFF and Sony DSF files
fixed: seeking in > 2GB WavPack files (new stream reader)
fixed: accept > 4GB source audio files (all formats)
improved: increase maximum samples from 2^32 to 2^40
added: block checksums for robustness to corruption
added: support for non-standard channel identities
removed: support for legacy WavPack files (< 4.0)
added: block decoder for streaming applications
fixed: many small fixes and improvements
added: all new pdf documentation
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Gordon
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Re: wavpack v 5.00 - verlustfreie Audiokompression

Beitrag von Gordon » 18.01.18, 18:34

Release 5.1.0 - January 18, 2017
--------------------------------

WavPack Library Source Code - 5.1.0
wavpack.exe (command-line encoder) - 5.1.0
wvunpack.exe (command-line decoder) - 5.1.0
wvgain.exe (command-line ReplayGain scanner) - 5.1.0
wvtag.exe (command-line tagging utility) - 5.1.0
----------------------------------------------------
added: all new command-line tagging utility (wvtag)
added: option to import ID3v2.3 tags from Sony DSF files
fixed: fuzz test failures from AFL reported on SourceForge
improved: DSD decimation filter (less HF rolloff & CPU use)
fixed: non-byte audio depths (12-bit, 20-bit) not showing
fixed: rare case of noise-shaping triggering a lossy mute
fixed: recognize UTF-8 BOM when reading text files
fixed: a few portability issues

in_wv.dll (winamp plugin) - 2.8.0.3
CoreWavPack DirectShow Filters - 1.5.1.0
AmioWavpack.amio (Adobe Audition Plugins) - 2.1
cool_wv4.flt (Cool Edit / Audition filter) - 3.1
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