LossyWAV: Difference between revisions
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:*usable with unmodified, established lossless formats. | :*usable with unmodified, established lossless formats. |
Revision as of 19:20, 15 June 2008
Developer(s) | Nick.C |
Release information | |
---|---|
Stable release | 1.0.0b |
Preview release | 1.0.1t |
Compatibility | |
Operating system | Windows |
Additional information | |
Use | Digital signal processing |
License | GNU GPL |
Website | Hydrogenaudio |
lossyWAV is a new free lossy pre-processor for PCM audio contained in the WAV file format. It reduces bit depth of the input signal, which, when used in conjunction with certain lossless codecs, reduces the bitrate of the encoded file significantly compared to unpreprocessed compression. lossyWAV's primary goal is to maintain transparency with a high degree of confidence when processing any audio data.
History
lossyFLAC is an idea started by 2Bdecided at Hydrogenaudio, utilising the wasted bits feature of the FLAC lossless codec with the aim of transparently reducing audio bit depth (making some lower significant bits (LSB's) zero), consequently taking advantage of FLAC's detection of consistently-zeroed lower significant bits within each single frame and significantly increasing coding efficiency.[1] In this way the user can enjoy audio encoded using the same codec (which may be all important from a hardware compatibility perspective) at a reduced bitrate compared to the lossless version.
Nick.C ported the original MATLAB implementation to Delphi (Many thanks CodeGear for Turbo Explorer!!) with a liberal sprinkling of IA-32 and x87 Assembly Language for speed.
Subsequently, lossyFLAC proved itself to work with other lossless codecs, so the application name was changed to lossyWAV.
Since then, Nick.C has heavily developed and built upon lossyWAV, with valuable tuning performed by halb27 at Hydrogenaudio.
Indicative bitrate reduction
It must be stressed that lossyWAV is a pure variable bitrate pre-processor. Bits-to-remove from the audio data are calculated on a block-by-block basis (codec-block length = 512 samples, 11.6msec @ 44.1kHz) using overlapping fast Fourier Transform (FFT) analyses of at least two lengths (default quality preset (-q 5) = 32, 64 & 1024 samples). After some manipulation, the results of each FFT analysis for a specific codec-block are then grouped and the minimum value used to determine bits-to-remove for the whole codec-block. Bit removal adds white noise to the output, however the level of the added noise associated with the removal of a number of bits has been pre-calculated and the number of bits to remove will depend on the level of the noise floor of the codec-block in question. Each sample in the codec-block is then rounded such that the first <bits-to-remove> lsb's are zero. In this way the wasted bits feature of FLAC et al is exploited.
lossyWAV Test Set | Version | FLAC -8 | -q 10 | -q 9 | -q 8 | -q 7 | -q 6 | -q 5 | -q 4 | -q 3 | -q 2 | -q 1 | -q 0 |
---|---|---|---|---|---|---|---|---|---|---|---|---|---|
53 sample "problem" set | 1.0.0b | 784 kbit/s | 654 kbit/s | 626 kbit/s | 596 kbit/s | 565 kbit/s | 534 kbit/s | 501 kbit/s | 470 kbit/s | 447 kbit/s | 408 kbit/s | 366 kbit/s | 329 kbit/s |
lossyWAV Test Set | Version | FLAC -8 | --insane | --extreme | --standard | --portable | -q 0 |
---|---|---|---|---|---|---|---|
55 sample "problem" set | 1.0.1t | 780 kbit/s | 656 kbit/s | 583 kbit/s | 508 kbit/s | 425 kbit/s | 321 kbit/s |
10 album test set | 1.0.1t | 854 kbit/s | 632 kbit/s | 548 kbit/s | 462 kbit/s | 376 kbit/s | 285 kbit/s |
File identification
lossyWAV-processed WAV files are named with a double filename extension, .lossy.wav, to make them instantly identifiable. e.g. ".lossy.flac" would indicate an audio file which was processed using lossyWAV, and subsequently encoded using FLAC.[2]
The --correction parameter is used when processing to create a correction file which is named with the .lwcdf.wav double filename extension. When "added" to the corresponding .lossy.wav, using the --merge parameter, the original file will be reconstituted.
Combinations of lossyWAV with each specific encoder are referred to as lossyX, where X is an abbreviation of the lossless codec name. Combination names are listed in the "known supported codecs" section below.
lossyWAV inserts a variable-length 'fact' chunk into the WAV file immediately after the 'fmt ' chunk. This takes the form:
fact/<size>/lossyWAV x.y.z @ dd/mm/yyyy hh:mm:ss, -q 5
Where the version, date & time and user settings are copied. Additionally, if a lossyWAV 'fact' chunk is found in a file, the processing will be halted (exit code = 16) to prevent re-processing of an already processed file.
The -check parameter can be used to determine whether a file has previously been processed without trying to process it, exit code = 16 if already processed; exit code = 0 if not.
Quality presets
- -q 10 to -q 8: Highest quality presets, disc space-saving alternative to lossless archiving for large audio collections, considered to be suitable for transcoding to other lossy codecs;
- -q 7 to -q 6: High quality presets, disc space-saving alternative to lossless archiving for large audio collections;
- -q 5: Default preset, generally accepted to be transparent;
- -q 4 to -q 0: DAP quality presets of reducing bitrate with reducing quality preset number, for usage on a compatible DAP. [3]
New quality preset synonyms introduced at 1.0.1j/k (and -q <n> quality parameter moved to advanced settings):
- --insane: (-q 10) Highest quality preset, generally considered to be excessive;
- --extreme: (-q 7.5) High quality preset, disc space-saving alternative to lossless archiving for large audio collections, considered to be suitable for transcoding to other lossy codecs;
- --standard: (-q 5) Default preset, generally accepted to be transparent;
- --portable: (-q 2.5) DAP quality preset for use on a compatible DAP. [4]
All tuning has been performed on quality preset -q 5 with higher presets being more conservative. Quality preset -q 5 is generally accepted to be (and from testing so far is) transparent. If you find a track which -q 5 fails to achieve transparency after processing, please post a sample (no more than 30 seconds) in the development thread.
Supported input formats
- WAV: 9-bit to 32-bit integer; 1 to 8 channels; sample rate ≥ 32kHz PCM. Very high sample rates (>48kHz) have not been extensively tested. Tunings have been focussed on 16-bit, 44.1kHz samples (i.e. CD PCM).
Codec compatibility
Codec | Supported | Encoder parameters | Combination name |
---|---|---|---|
Apple Lossless | No | — | — |
FLAC | Yes | -5 -b 512 --keep-foreign-metadata | lossyFLAC |
LA | No | — | — |
LPAC | Yes | -b512 | lossyLPAC |
Monkey's Audio | No | — | — |
MPEG-4 ALS | Yes | -l -n512 | lossyALS |
OptimFROG | No | — | — |
TAK | Yes | -fsl512 | lossyTAK |
TTA | No | — | — |
WavPack | Yes | --blocksize=512 | lossyWV |
WMA Lossless | Yes | — | lossyWMALSL |
There is also evidence — so-called "Bit Shifting" — to suggest that lossyWAV may work with MLP, but this remains untested due to prohibitive prices of encoders.
A comparison of portable media players is here, which shows FLAC and WMA Lossless compatibility among listed players. Any player supported by Rockbox can use FLAC or WavPack files after installing Rockbox.
Important note
NB: when encoding using a lossless codec, please ensure that the block size of the lossless codec matches that of lossyWAV (default = 512 samples). If this is not done then the lossless encoding of the processed WAV file will (almost certainly) be larger than it would otherwise have been. This is achieved by adding the "Encoder Parameters" in the table above to the command line of the lossless codec in question.
Bonus feature
Another, possibly not obvious, feature of lossyWAV is that the processed output can be "transcoded" from one lossless codec to another lossless codec with absolutely no loss of quality whatsoever. This is solely due to the fact that lossyWAV output is designed to be losslessly encoded - something that lossless codecs do very well indeed.
Using lossyWAV
Application settings
lossyWAV 1.0.0b, Copyright (C) 2007,2008 Nick Currie. Copyleft. This program is free software: you can redistribute it and/or modify it under the terms of the GNU General Public License as published by the Free Software Foundation, either version 3 of the License, or (at your option) any later version. This program is distributed in the hope that it will be useful,but WITHOUT ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for more details. You should have received a copy of the GNU General Public License along with this program. If not, see <http://www.gnu.org/licenses/>. Usage : lossyWAV <input wav file> <options> Example : lossyWAV musicfile.wav Quality Options: -q, --quality <n> quality preset (10=highest quality, 0=lowest bitrate; -q 5 is generally accepted to be transparent) default=-q 5. Standard Options: -c, --check check if WAV file has already been processed; default=off. errorlevel=16 if already processed, 0 if not. -C, --correction write correction file for processed WAV file; default=off. -f, --force forcibly over-write output file if it exists; default=off. -h, --help display help. -L, --longhelp display extended help. -M, --merge merge existing lossy.wav and lwcdf.wav files. -N, --noclips set allowable number of clips / channel / codec block to 0; default=3,3,3,3,2,1,0,0,0,0,0 (-q 0 to -q 10) -o, --outdir <dir> destination directory for the output file(s). -v, --version display the lossyWAV version number. Special thanks: David Robinson for the method itself and motivation to implement it. Don Cross for the Complex-FFT algorithm used. Horst Albrecht for valuable tuning input and feedback.
Example Foobar2000 converter settings
Example flossy.bat file called from Foobar2000
@echo off z:\bin\lossyWAV %1 %3 %4 %5 %6 %7 %8 %9 --below --nowarnings --quiet z:\bin\flac -5 -f -b 512 "%~N1.lossy.wav" -o"%~N2.flac" del "%~N1.lossy.wav"
Frequently asked questions
- Question: Why is the ".wav" file extension used?
- Answer: The ".wav" file extension is used because lossyWAV is a digital signal processor and not a codec. No decoding is required for any program to play a WAV file which has been processed with lossyWAV as it remains compliant with the RIFF WAVE format.
- Question: Why create a processor which means that I cannot be sure that a lossless file is truly lossless?
- Answer: Unless one creates the lossless file personally, one can never be completely sure that the file is indeed lossless. e.g. If a WAV file is encoded to mp3 and then transcoded to a lossless codec, how can this pre-processing be easily determined?
- Question: Is it VBR?
- Short answer: Yes.
- Question: Do I need to re-process to change lossless codecs?
- Short answer: No.
- Question: Is it transparent?
- Short answer: At preset --standard, almost certainly.
- Question: Is it lossless?
- Short answer: No.
- Question: Will it ever have a CBR mode?
- Short answer: No.
- Question: Why should I use this?
- Answer:
External links
- Original lossyFLAC thread Where David Robinson (Replay Gain developer) introduces the method and a MATLAB implementation.
- Original development thread up to 1.0.0b release
- lossyWAV 1.0.0b release thread
- Current development thread You will find the latest release candidate and latest beta version in post #1 of this thread.