https://wiki.hydrogenaud.io/api.php?action=feedcontributions&user=Gottkaiser&feedformat=atomHydrogenaudio Knowledgebase - User contributions [en]2024-03-29T13:07:11ZUser contributionsMediaWiki 1.22.7https://wiki.hydrogenaud.io/index.php?title=LossyWAVLossyWAV2015-05-19T19:00:19Z<p>Gottkaiser: added 1.5.0 Development thread</p>
<hr />
<div>{{Software Infobox<br />
| name = lossyWAV<br />
| logo =<br />
| screenshot = <br />
| caption = <br />
| maintainer = [http://www.hydrogenaud.io/forums/index.php?showuser=42400 Nick.C]<br />
| stable_release = 1.4.0<br />
| preview_release = 1.4.1a beta<br />
| operating_system = [[Wikipedia:Microsoft Windows|Windows]]<br />
| use = [[Wikipedia:Digital signal processing|Digital signal processing]]<br />
| license = [[Wikipedia:GNU General Public License|GNU GPL]]<br />
| website = [http://www.hydrogenaud.io/forums/index.php?showtopic=107081 1.4.0 release thread]<br />[http://www.hydrogenaud.io/forums/index.php?showtopic=109239 1.5.0 development thread]<br />
}}<br />
lossyWAV is a [[Wikipedia:Free software|free]], [[lossy]] pre-processor for [[PCM]] audio contained in the [[RIFF_WAVE|WAV]] file format. Proposed by [http://www.hydrogenaud.io/forums/index.php?showuser=409 David Robinson], it reduces [[Wikipedia:Audio bit depth|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.<br />
lossyWAV's primary goal is to maintain [[transparency]] with a high degree of confidence when processing any audio data.<br />
<br />
==History==<br />
lossyWAV is based on the lossyFLAC idea proposed by [http://www.hydrogenaud.io/forums/index.php?showuser=409 David Robinson] at Hydrogenaudio, which is a method of carefully reducing the bitdepth of (blocks of) samples which will then allow the FLAC lossless encoder to make use of its wasted bits feature. The aim is to transparently reduce audio bit depth (by making some lower significant bits ([[Wikipedia:Least_significant_bit|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.[http://www.hydrogenaud.io/forums/index.php?s=&showtopic=55522&view=findpost&p=498179] 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.<br />
<br />
[http://www.hydrogenaud.io/forums/index.php?showuser=42400 Nick Currie] ported the original [[Wikipedia:MATLAB|MATLAB]] implementation to [[Wikipedia:Borland Delphi|Delphi]] (Many thanks [[Wikipedia:CodeGear|CodeGear]] for Turbo Explorer!) with a liberal sprinkling of [[Wikipedia:IA-32|IA-32]] and [[Wikipedia:x87|x87]] Assembly Language for speed.<br />
<br />
Subsequently, lossyFLAC proved itself to work with other lossless codecs, so the application name was changed to lossyWAV. <br />
<br />
Since then, Nick has heavily developed and built upon lossyWAV, with valuable tuning performed by [http://www.hydrogenaud.io/forums/index.php?showuser=25015 Horst Albrecht] at Hydrogenaudio. Although the current lossyWAV implementation has built on David's original method, the method itself still very much belongs to its author.<br />
<br />
==Indicative bitrate reduction==<br />
It must be stressed that lossyWAV is a pure variable bit-depth pre-processor in that the overall sample size remains the same after processing but the number of significant bits used for the samples in a codec-block can change on a block-by-block basis. 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 [[Wikipedia:fast Fourier transform|fast Fourier Transform]] (FFT) analyses of at least two lengths (default quality preset (-q 5) = 32, 64 & 1024 [[Wikipedia:Sampling %28signal processing%29|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 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. The added noise is adaptively shaped by default, however the user can select parameters to make the added noise fixed shaped or simply [[Wikipedia:white noise|white noise]]. 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.<br />
<br />
{| class="wikitable" style="text-align:center"<br />
|-<br />
!lossyWAV Test Set (16 bit / 44.1kHz)<br />
!Codec<br />
!lossless<br />
!--insane<br />
!--extreme<br />
!--high<br />
!--standard<br />
!--economic<br />
!--portable<br />
!--extraportable<br />
|-<br />
!10 Album Test Set<br />
| FLAC<br />
| 854 kbit/s<br />
| 627 kbit/s<br />
| 548 kbit/s<br />
| 477 kbit/s<br />
| 442 kbit/s<br />
| 407 kbit/s<br />
| 353 kbit/s<br />
| 311 kbit/s<br />
|-<br />
!Nick.C's Full Collection<br />
| FLAC<br />
| 882 kbit/s<br />
| -<br />
| -<br />
| -<br />
| -<br />
| -<br />
| -<br />
| 307 kbit/s<br />
|}<br />
<br />
==File identification==<br />
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.[http://www.hydrogenaud.io/forums/index.php?s=&showtopic=55522&view=findpost&p=498559]<br />
<br />
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.<br />
<br />
Combinations of lossyWAV with each specific encoder are referred to as lossy'''X''', where '''X''' is an abbreviation of the lossless codec name. Combination names are listed in the "[[LossyWAV#Known supported codecs|known supported codecs]]" section below.<br />
<br />
lossyWAV inserts a variable-length 'fact' chunk into the WAV file immediately after the 'fmt ' chunk. This takes the form:<pre>fact/<size>/lossyWAV x.y.z @ dd/mm/yyyy hh:mm:ss, -q 5</pre>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.<br />
<br />
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.<br />
<br />
==Quality presets==<br />
*--quality insane: (-q I or -q 10) Highest quality preset, generally considered to be excessive;<br />
*--quality extreme: (-q E or -q 7.5) Higher quality preset, disc space-saving alternative to lossless archiving for large audio collections, considered to be suitable for transcoding to other lossy codecs;<br />
*--quality high: (-q H or -q 5.0) High quality preset, midway between extreme and standard;<br />
*--quality standard: (-q S or -q 2.5) Default preset, generally accepted to be transparent;<br />
*--quality economic: (-q C or -q 0.0) Intermediate preset midway between standard and portable;<br />
*--quality portable: (-q P or -q -2.5) DAP quality preset for use on a compatible [[Wikipedia:Digital audio player|DAP]].[http://www.hydrogenaud.io/forums/index.php?s=&showtopic=56129&view=findpost&p=531316]<br />
*--quality extraportable: (-q X or -q -5.0) Lowest quality preset for use on a compatible [[Wikipedia:Digital audio player|DAP]].[http://www.hydrogenaud.io/forums/index.php?s=&showtopic=56129&view=findpost&p=531316]<br />
<br />
All tuning for version 1.0.0 was performed on quality preset --standard with higher presets being more conservative. For versions 1.1.0, 1.2.0 and 1.3.0, tuning effort has been focused on the lowest quality preset in an effort to achieve an effective compromise between resultant bitrate and perceived quality. Quality preset --standard is generally accepted to be (and from testing so far is) transparent. If you find a track which --standard fails to achieve transparency after processing, please post a sample (no more than 30 seconds) in the development thread.<br />
<br />
The upper frequency limit used in the calculation of minimum signal power varies, dependent on quality preset, in the range 15.159kHz to 16.682kHz<br />
<br />
==Supported input formats==<br />
*[[WAV]]: 9-bit to 32-bit integer; 1 to 8 channels; sample rate &ge; 32kHz [[Pulse Code Modulation|PCM]]. Very high sample rates (&gt;48kHz) have not been extensively tested. Tunings have been focussed on 16-bit, 44.1kHz samples (i.e. [[Wikipedia:Red Book (audio CD standard)|CD]] PCM).<br />
<br />
==Codec compatibility==<br />
{| class="wikitable" style="text-align:center"<br />
|-<br />
!Codec<br />
!Supported<br />
!Encoder parameters<br />
!Combination name<br />
|-<br />
! [[Free Lossless Audio Codec|FLAC]]<br />
| '''Yes'''<br />
| -'''5''' -'''b''' 512 --'''keep-foreign-metadata'''<br />
| lossy'''FLAC'''<br />
|-<br />
! [[Lossless Predictive Audio Compression|LPAC]]<br />
| '''Yes'''<br />
| -'''b'''512<br />
| lossy'''LPAC'''<br />
|-<br />
! [[Wikipedia:Audio Lossless Coding|MPEG-4 ALS]]<br />
| '''Yes'''<br />
| -'''l''' -'''n'''512<br />
| lossy'''ALS'''<br />
|-<br />
! [[TAK]]<br />
| '''Yes'''<br />
| -'''fsl'''512<br />
| lossy'''TAK'''<br />
|-<br />
! [[WavPack]]<br />
| '''Yes'''<br />
| --'''blocksize'''=512 --'''merge-blocks'''<br />
| lossy'''WV'''<br />
|-<br />
! [[Windows Media Audio#Windows Media Audio Lossless|WMA Lossless]]<br />
| '''Yes'''<br />
| &mdash;<br />
| lossy'''WMALSL'''<br />
|-<br />
! [[Apple Lossless]]<br />
| No<br />
| &mdash;<br />
| &mdash;<br />
|-<br />
! [[Lossless Audio|LA]]<br />
| No<br />
| &mdash;<br />
| &mdash;<br />
|-<br />
! [[Monkey's Audio]]<br />
| No<br />
| &mdash;<br />
| &mdash;<br />
|-<br />
! [[OptimFROG]]<br />
| No<br />
| &mdash;<br />
| &mdash;<br />
|-<br />
! [[Wikipedia:TTA (codec)|TTA]]<br />
| No<br />
| &mdash;<br />
| &mdash;<br />
|}<br />
<br />
* Combinations of lossyWAV with each specific encoder are referred to as lossy'''X''', where '''X''' is an abbreviation of the lossless codec name.<br />
<br />
<br />
There is also [http://www.hometheaterhifi.com/volume_8_4/dvd-benchmark-part-6-dvd-audio-11-2001.html#Meridian%20Lossless%20Packing%20(MLP)%20in%20a%20Nutshell evidence] &mdash; so-called "Bit Shifting" &mdash; to suggest that lossyWAV may work with [[Wikipedia:Meridian Lossless Packing|MLP]], but this remains untested due to prohibitive prices of encoders. At least one [http://www.hydrogenaud.io/forums/index.php?showtopic=98609&hl= commercial DVD-A] uses constant bit-depth reduction with lower bit-depth on rear channels.<br />
<br />
A comparison of portable media players is [[Wikipedia:Comparison of portable media players#Audio Formats|here]], which shows FLAC and WMA Lossless compatibility among listed players.<br />
Any player supported by [http://www.rockbox.org Rockbox] can use FLAC or WavPack files after installing Rockbox.<br />
===Important note===<br />
'''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.'''<br />
===Bonus feature===<br />
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.<br />
<br />
==Using lossyWAV==<br />
===Application settings===<br />
<pre><br />
lossyWAV 1.4.0, Copyright (C) 2007-2014 Nick Currie. Copyleft.<br />
<br />
This program is free software: you can redistribute it and/or modify it under<br />
the terms of the GNU General Public License as published by the Free Software<br />
Foundation, either version 3 of the License, or (at your option) any later<br />
version.<br />
<br />
This program is distributed in the hope that it will be useful,but WITHOUT ANY<br />
WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR A<br />
PARTICULAR PURPOSE. See the GNU General Public License for more details.<br />
<br />
You should have received a copy of the GNU General Public License along with<br />
this program. If not, see <http://www.gnu.org/licenses/>.<br />
<br />
Process Description:<br />
<br />
lossyWAV is a near lossless audio processor which dynamically reduces the<br />
bitdepth of the signal on a block-by-block basis. Bitdepth reduction adds noise<br />
to the processed output. The amount of permissible added noise is based on<br />
analysis of the signal levels in the default frequency range 20Hz to 16kHz.<br />
<br />
If signals above the upper limiting frequency are at an even lower level, they<br />
can be swamped by the added noise. This is usually inaudible, but the behaviour<br />
can be changed by specifying a different --limit (in the range 10kHz to 20kHz).<br />
<br />
For many audio signals there is little content at very high frequencies and<br />
forcing lossyWAV to keep the added noise level lower than the content at these<br />
frequencies can increase the bitrate dramatically for no perceptible benefit.<br />
<br />
The noise added by the process is shaped using an adaptive method provided by<br />
Sebastian Gesemann. This method, as implemented in lossyWAV, aims to use the<br />
signal itself as the basis of the filter used for noise shaping. Adaptive noise<br />
shaping is enabled by default.<br />
<br />
Usage : lossyWAV <input wav file> <options><br />
<br />
Example : lossyWAV musicfile.wav<br />
<br />
Quality Options:<br />
<br />
-q, --quality <t> where t is one of the following (default = standard):<br />
I, insane highest quality output, suitable for transcoding;<br />
E, extreme higher quality output, suitable for transcoding;<br />
H, high high quality output, suitable for transcoding;<br />
S, standard default quality output, considered to be transparent;<br />
C, economic intermediate quality output, likely to be transparent;<br />
P, portable good quality output for DAP use, may not be transparent;<br />
X, extraportable lowest quality output, probably not transparent.<br />
<br />
Standard Options:<br />
<br />
-C, --correction write correction file for processed WAV file; default=off.<br />
-f, --force forcibly over-write output file if it exists; default=off.<br />
-h, --help display help.<br />
-L, --longhelp display extended help.<br />
-M, --merge merge existing lossy.wav and lwcdf.wav files.<br />
-o, --outdir <t> destination directory for the output file(s).<br />
-v, --version display the lossyWAV version number.<br />
-w, --writetolog create (or add to) lossyWAV.log in the output directory.<br />
<br />
Advanced Options:<br />
<br />
- take WAV input from STDIN.<br />
-c, --check check if WAV file has already been processed; default=off.<br />
errorlevel=16 if already processed, 0 if not.<br />
-q, --quality <n> quality preset (-5.0<=n<=10.0); (-5=lowest, 10=highest;<br />
default=2.5; I=10.0; E=7.5; H=5.0; S=2.5; C=0.0; P=-2.5;<br />
X=-5.0.<br />
--, --stdout write WAV output to STDOUT.<br />
--stdinname <t> pseudo filename to use when input from STDIN.<br />
<br />
Advanced Quality Options:<br />
<br />
-a, --analyses <n> set number of FFT analysis lengths, (2<=n<=7; default=3,<br />
i.e. 32, 64 & 1024 samples. n = 2, remove 32 sample FFT;<br />
n > 3 add 16; n > 4, add 128; n > 5, add 256, n > 6, add<br />
512) n.b. FFT lengths stated are for 44.1/48kHz audio,<br />
higher sample rates will automatically increase all FFT<br />
lengths as required.<br />
--feedback [n] enable experimental bit removal / adaptive noise shaping<br />
noise limiter. Tuning has been carried out at -q X and<br />
should have a negligible effect at -q S. Optional setting<br />
(0.0 <= n <= 10.0, default = 0.0) automatically selects<br />
the following parameters (0 = least effect, 10 = most):<br />
r, round <n> limit deviation from expected added noise due to rounding<br />
(-2.0 <= n <= 2.0, default = 0.0).<br />
n, noise <n> limit added noise due to adaptive noise shaping<br />
(-2.5 <= n <= 2.5, default = 0.0).<br />
a, aclips <n> number of permissible exceedences of adaptive noise<br />
shaping level limit (0 <= n <= 64, default = 32).<br />
A, alevel <n> adaptive noise shaping level limit (-2.0 <= n <= 2.5,<br />
default = 0.0).<br />
V, verbose enable more detailed feedback information in output.<br />
-I, --ignore-chunk-sizes.<br />
ignore 'RIFF' and 'data' chunk sizes in input.<br />
-l, --limit <n> set upper frequency limit to be used in analyses to n Hz;<br />
(12500 <= n <= 20000; default=16000).<br />
--linkchannels revert to original single bits-to-remove value for all<br />
channels rather than channel dependent bits-to-remove.<br />
--maxclips <n> set max. number of acceptable clips per channel per block;<br />
(0 <= n <= 16; default = 3,3,3,3,3,2,2,2,2,2,1,1,1,0,0,0).<br />
-m, --midside analyse 2 channel audio for mid/side content.<br />
--nodccorrect disable DC correction of audio data prior to FFT analysis,<br />
default=on; (DC offset calculated per FFT data set).<br />
-n, --noskew disable application of low frequency level reduction prior<br />
to determination of bits-to-remove.<br />
--scale <n> factor to scale audio by; (0.03125 < n <= 8.0; default=1).<br />
-s, --shaping modify settings for noise shaping used in bit-removal:<br />
a, altfilter enable alternative adaptive shaping filter method.<br />
c, cubic enable cubic interpolation when defining filter shape<br />
e, extra additional white noise to add during creation of filter<br />
f, fixed disable adaptive noise shaping (use fixed shaping)<br />
n, nowarp disable warped noise shaping (use linear frequency shaping)<br />
o, off disable noise shaping altogether (use simple rounding)<br />
s, scale <n> change effectiveness of noise shaping (0 < n <= 2; default<br />
= 1.0)<br />
t, taps <n> select number of taps to use in FIR filter (32 <= n <= 256;<br />
default = 64)<br />
w, warp enable cubic interpolation when creating warped filter<br />
-U, --underlap <n> enable underlap mode to increase number of FFT analyses<br />
performed at each FFT length, (n = 2, 4 or 8, default=2).<br />
<br />
Output Options:<br />
<br />
--bitdist show distrubution of bits to remove.<br />
--blockdist show distribution of lowest / highest significant bit of<br />
input codec-blocks and bit-removed codec-blocks.<br />
-d, --detail enable per block per channel bits-to-remove data display.<br />
-F, --freqdist enable frequency analysis display of input data.<br />
-H, --histogram show sample value histogram (input, lossy and correction).<br />
--longdist show long frequency distribution data (input/lossy/lwcdf).<br />
--perchannel show selected distribution data per channel.<br />
-p, --postanalyse enable frequency analysis display of output and<br />
correction data in addition to input data.<br />
--sampledist show distribution of lowest / highest significant bit of<br />
input samples and bit-removed samples.<br />
--spread [full] show detailed [more detailed] results from the spreading/<br />
averaging algorithm.<br />
-W, --width <n> select width of output options (79<=n<=255).<br />
<br />
System Options:<br />
<br />
-B, --below set process priority to below normal.<br />
--low set process priority to low.<br />
-N, --nowarnings suppress lossyWAV warnings.<br />
-Q, --quiet significantly reduce screen output.<br />
-S, --silent no screen output.<br />
<br />
Special thanks go to:<br />
<br />
David Robinson for the publication of his lossyFLAC method, guidance, and<br />
the motivation to implement his method as lossyWAV.<br />
<br />
Horst Albrecht for ABX testing, valuable support in tuning the internal<br />
presets, constructive criticism and all the feedback.<br />
<br />
Sebastian Gesemann for the adaptive noise shaping method and the amount of<br />
help received in implementing it and also for the basis of<br />
the fixed noise shaping method.<br />
<br />
Tyge Lovset for the C++ translation initiative.<br />
<br />
Matteo Frigo and for libfftw3-3.dll contained in the FFTW distribution<br />
Steven G Johnson (v3.2.1 or v3.2.2).<br />
<br />
Mark G Beckett for the Delphi unit that provides an interface to the<br />
(Univ. of Edinburgh) relevant fftw routines in libfftw3-3.dll.<br />
<br />
Don Cross for the Complex-FFT algorithm originally used.</pre><br />
<br />
===Example drag 'n' drop batch file===<br />
Simply drag the FLAC files onto this batch file and it will process, recode in FLAC and copy ALL of the tags from the input FLAC file, placing the output lossyFLAC file in the same directory as the input FLAC file. Requires flac.exe and [http://www.synthetic-soul.co.uk/tag/ tag.exe] to be somewhere on the path. <br />
<pre>@echo off<br />
:repeat<br />
if %1.==. goto end<br />
if exist "%1" flac -d "%1" --stdout --silent|lossywav - --stdout --standard --stdinname "%1"|flac - -b 512 -o "%~dpn1.lossy.flac" --silent && tag --fromfile "%1" "%~dpn1.lossy.flac"<br />
shift<br />
goto repeat<br />
:end</pre><br />
<br />
===lossyWAV and FFTW===<br />
Since version 1.2.0, lossyWAV has been compatible with [[Wikipedia:FFTW|FFTW]] although not dependent on it. Should the user wish to take advantage of the increased processing speed available when using FFTW (from superior FFT implementations), libfftw3-3.dll should be placed in a directory on the host computer which features on the path.<br />
<br />
===Linux / OS X support: lossyWAV and WINE===<br />
The cause of lossyWAV's WINE incompatibility was found and removed during the development of 1.2.0 and retrospectively amended for 1.1.0b in a maintenance release (1.1.0c). The latest stable version (1.3.0 at the time of writing) is fully supported.<br />
<br />
[http://caudec.net/ caudec] is a command-line tool that can encode and decode lossyWAV files (lossyFLAC, lossyWV, lossyTAK), using the official binary (lossyWAV.exe) with Wine (see: [http://caudec.net/documentation/windowscodecs/ installation instructions]). Caudec can also test file integrity and compute (and tag) Replaygain data. While it hasn't been tested at the time of writing, it is possible that lossyWAV support in caudec works on OS X as well.<br />
<br />
===lossyWAV and [[foobar2000]]===<br />
Example [[foobar2000]] converter settings:<br />
<br />
lossyFLAC settings:<pre>Encoder: c:\windows\system32\cmd.exe<br />
Extension: lossy.flac<br />
Parameters: /d /c c:\"program files"\bin\lossywav - --quality standard --silent --stdout|c:\"program files"\bin\flac - -b 512 -5 -f -o%d --ignore-chunk-sizes<br />
Format is: lossless or hybrid<br />
Highest BPS mode supported: 24</pre><br />
<br />
lossyTAK settings:<pre>Encoder: c:\windows\system32\cmd.exe<br />
Extension: lossy.tak<br />
Parameters: /d /c c:\"program files"\bin\lossywav - --quality standard --silent --stdout|c:\"program files"\bin\takc -e -p2m -fsl512 -ihs - %d<br />
Format is: lossless or hybrid<br />
Highest BPS mode supported: 24</pre><br />
<br />
lossyWV settings:<pre>Encoder: c:\windows\system32\cmd.exe<br />
Extension: lossy.wv<br />
Parameters: /d /c c:\"program files"\bin\lossywav - --quality standard --silent --stdout|c:\"program files"\bin\wavpack -hm --blocksize=512 --merge-blocks -i - %d<br />
Format is: lossless or hybrid<br />
Highest BPS mode supported: 24</pre><br />
<br />
lossyWMALSL* settings:<pre>Encoder: c:\windows\system32\cmd.exe<br />
Extension: lossy.wma<br />
Parameters: /d /c c:\"program files"\bin\lossywav - --quality standard --silent --stdout|c:\"program files"\bin\wmaencode.exe - %d --codec lsl --ignorelength<br />
Format is: lossless or hybrid<br />
Highest BPS mode supported: 24</pre><br />
<br />
Enclose the element of the path containing spaces within double quotation marks ("), e.g. C:\"Program Files"\directory_where_executable_is\executable_name. This is a Windows limitation.<br />
<br />
lossyWMALSL conversion uses WMAEncode.exe by lvqcl found [http://www.hydrogenaud.io/forums/index.php?s=&showtopic=90519&view=findpost&p=767754 here].<br />
<br />
===lossyWAV and EAC===<br />
:''For example settings, see [[EAC and LossyWAV]].''<br />
<br />
==Frequently asked questions==<br />
*'''Question:''' Why is the ".wav" file extension used?<br />
*'''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.<br />
<br />
*'''Question:''' Why create a processor which means that I cannot be sure that a lossless file is truly lossless?<br />
*'''Answer:''' Unless one creates the lossless file personally, one can '''never''' be completely sure that the file is indeed lossless. E.g. a lossless file you receive could be transcoded from [[MP3]] without your knowledge. To distinguish a lossyWAV file from lossless files it is recommended to use the extension .lossy.EXT where EXT is the original extension e.g. .lossy.flac<br />
<br />
*'''Question:''' Is it [[Variable Bitrate|VBR]]?<br />
*'''Short answer:''' Yes.<br />
<br />
*'''Question:''' Do I need to re-process to change lossless codecs?<br />
*'''Short answer:''' No.<br />
<br />
*'''Question:''' Is it [[transparency|transparent]]?<br />
*'''Short answer:''' At preset --standard, almost certainly.<br />
<br />
*'''Question:''' Is it [[lossless]]?<br />
*'''Short answer:''' No.<br />
<br />
*'''Question:''' Will it ever have a [[Constant Bitrate|CBR]] mode?<br />
*'''Short answer:''' No.<br />
<br />
*'''Question:''' Will it low-pass filter my audio?<br />
*'''Short answer:''' No. The frequency limit is for the analysis only. LossyWAV cannot low-pass filter your audio.<br />
<br />
*'''Question:''' Why should I use this?<br />
*'''Answer:'''<br />
:*high quality<br />
:*extremely low chance of audible [[artifact]]s<br />
:*reasonable [[bitrate]]s<br />
:*usable with unmodified, established lossless formats.<br />
<br />
==External links==<br />
*[http://www.hydrogenaud.io/forums/index.php?showtopic=55522 Original lossyFLAC thread] - Introduction of the concept by David Robinson (Replay Gain developer) and initial development<br />
----<br />
*[http://www.hydrogenaud.io/forums/index.php?showtopic=109239 lossyWAV 1.5.0 development thread]<br />
*[http://www.hydrogenaud.io/forums/index.php?showtopic=107081 lossyWAV 1.4.0 release thread] - Release of version 1.4.0 on 02 Oktober 2014<br />
----<br />
*[http://www.hydrogenaud.io/forums/index.php?showtopic=96635 lossyWAV 1.3.1 Delphi to C++ translation thread]<br />
----<br />
*[http://www.hydrogenaud.io/forums/index.php?showtopic=81002 lossyWAV 1.3.0 development thread]<br />
*[http://www.hydrogenaud.io/forums/index.php?showtopic=90104 lossyWAV 1.3.0 release thread] - Release of version 1.3.0 on 06 August 2011<br />
----<br />
*[http://www.hydrogenaud.io/forums/index.php?showtopic=65499 lossyWAV 1.2.0 development thread]<br />
*[http://www.hydrogenaud.io/forums/index.php?showtopic=77042 lossyWAV 1.2.0 release thread] - Release of version 1.2.0 on 16 December 2009<br />
----<br />
*[http://www.hydrogenaud.io/forums/index.php?showtopic=63254 lossyWAV 1.1.0 development thread]<br />
*[http://www.hydrogenaud.io/forums/index.php?showtopic=64617 lossyWAV 1.1.0 release thread] - Release of version 1.1.0 on 12 July 2008<br />
----<br />
*[http://www.hydrogenaud.io/forums/index.php?showtopic=56129 lossyWAV Development thread] - Conversion of the original MATLAB script to Delphi and evolution of the method<br />
*[http://www.hydrogenaud.io/forums/index.php?showtopic=63225 lossyWAV 1.0.0 release thread] - Release of version 1.0.0b on 12 May 2008<br />
<br />
[[Category:Software]]</div>Gottkaiserhttps://wiki.hydrogenaud.io/index.php?title=Mp3tagMp3tag2015-05-17T14:03:21Z<p>Gottkaiser: update and smal additions</p>
<hr />
<div>{{Software Infobox<br />
| name = MP3tag<br />
| logo =<br />
| screenshot = [[Image:MP3tag-screenshot.png|250px|MP3tag screenshot]]<br />
| caption = powerful and easy-to-use tool to edit metadata <br />
| maintainer = Florian Heidenreich<br />
| stable_release = [http://www.mp3tag.de/en/download.html v2.70] (2015-05-01)<br />
| preview_release = [http://forums.mp3tag.de/index.php?showtopic=57 here]<br />
| operating_system = Windows<br />
| use = Metadata<br />
| license = Freeware <br />
| website = [http://www.mp3tag.de/en/ http://www.mp3tag.de]<br />
}}<br />
<br />
=Introduction=<br />
'''MP3tag''' is an powerful and mature [[Metadata|metadata]] editor for lots of common audio formats. Its development was started in 1999 by Florian Heidenreich.<br />
<br />
You can rename files based on the [[Tagging|tag]] information, replace characters or words from tags and filenames, import/export tag information, create playlists. The program supports online database lookups for selected files, allowing you to automatically gather proper tag information for select files or CDs. Batch processing to edit lots of ifles is supported.<br />
<br />
==Features==<br />
<br />
* Write [[ID3v1.1]]-, [[ID3v2]]-, [[APEv2]]-Tags and [[Vorbis_Comment|Vorbis Comments]] to multiple files at once<br />
* Full Unicode support<br />
* Support for embedded cover art<br />
* Automatically create playlists<br />
* Recursive subfolders support<br />
* Remove parts or the entire tag of multiple files<br />
* Rename files based on the tag information<br />
* Import tags from filenames<br />
* Format tags and filenames<br />
* Replace characters or words from tags and filenames<br />
* Regular Expressions<br />
* Export tag information to user-defined formats (like html, rtf, csv, xml)<br />
* Import tag information from online databases like Amazon, discogs, freedb, MusicBrain (also by text-search)<br />
* Import tag information from local freedb databases<br />
* Support for [[ID3v2|ID3v2.3]] (ISO-8859-1 and UTF-16) and [[ID3v2|ID3v2.4]] with [[UTF-8]]<br />
<br />
==Supported formats==<br />
<br />
* Advanced Audio Coding ([[AAC|aac]])<br />
* Free Lossless Audio Codec ([[FLAC|flac]])<br />
* Monkey's Audio ([[APE|ape]])<br />
* Mpeg Layer 3 ([[MP3|mp3]])<br />
* MPEG-4 ([[MP4|mp4]] / [[M4A|m4a]] / m4b / [[iTunes]] compatible)<br />
* Musepack ([[MPC|mpc]])<br />
* Ogg Vorbis ([[Ogg_Vorbis|ogg]])<br />
* OptimFROG ([[OptimFROG|ofr]])<br />
* OptimFROG DualStream (ofs)<br />
* Opus ([[Opus|opus]])<br />
* Speex ([[Speex|spx]])<br />
* TAK ([[TAK]])<br />
* True Audio (tta)<br />
* Windows Media Audio ([[WMA|wma]])<br />
* WavPack ([[WavPack|wv]])<br />
<br />
==Operating Systems==<br />
* Win XP SP3<br />
* Win 2003<br />
* Win Vista<br />
* Win 2008<br />
* Win 2012<br />
* Win 7<br />
* Win 8<br />
* Win 8.1<br />
<br />
Windows 2000 is no longer supported as of version 2.40. Version 2.39 is still available on the download page at the MP3tag website.<br />
<br />
==External links==<br />
* [http://www.mp3tag.de/en/ Mp3tag: Homepage]<br />
* [http://www.mp3tag.de/en/download.html Mp3tag: Download]<br />
* [http://www.anytag.de/forums/ Mp3tag: official forum]<br />
* [http://www.mp3tag.de/en/changelog.html Mp3tag: changelog]<br />
* [http://wiki.slimdevices.com/index.php/Mp3tagGuide Mp3tagGuide - Slim Devices]<br />
* [http://www.anytag.de/forums/index.php?showtopic=1794 Mp3tag: additional Web Sources]<br />
<br />
[[Category:Software]]<br />
[[Category:Tag editors]]</div>Gottkaiserhttps://wiki.hydrogenaud.io/index.php?title=Free_Lossless_Audio_CodecFree Lossless Audio Codec2015-05-17T13:29:21Z<p>Gottkaiser: add licence link and update one old link</p>
<hr />
<div>{{Codec Infobox<br />
| name = FLAC<br />
| logo = [[Image:FLAC logo.gif]]<br />
| type = lossless<br />
| purpose = Popular open source patent free lossless compression scheme.<br />
| maintainer = Josh Coalson, Xiph Community <br />
| recommended_encoder = FLAC encoder<br />
| recommended_text = FLAC v1.3.1 (25-11-2014)<br />
| website = http://xiph.org/flac/<br />
}}<br />
'''Free Lossless Audio Codec''' ('''FLAC''') is a codec for lossless audio compression.<br />
Grossly oversimplified, FLAC is similar to [[MP3]], but [[lossless]], meaning that audio is compressed in FLAC without any loss in quality. This is similar to how Zip works, except with FLAC you will get much better compression because it is designed specifically for audio, and you can play back compressed FLAC files in your favorite player (or your car or home stereo, if supported) just like you would an MP3 file.<br />
<br />
== General aspects of the format ==<br />
FLAC is freely available and supported on most operating systems, including Windows, UNIX (Linux, *BSD, Solaris, OS X, IRIX), BeOS, OS/2, and Amiga. There are build systems for autotools, MSVC, Watcom C, and Project Builder.<br />
<br />
The FLAC project consists of:<br />
* the stream format<br />
* reference encoders and decoders in library form<br />
* flac, a command-line program to encode and decode FLAC files<br />
* metaflac, a command-line metadata editor for FLAC files<br />
* input plugins for various music players<br />
<br />
When it's said that FLAC is ''free'', it means more than just that it is available at no cost. It means that the specification of the format is fully open to the public to be used for any purpose, although the FLAC project reserves the right to set the FLAC specification and certify compliance. It also means that neither the FLAC format nor any of the implemented encoding/decoding methods are covered by any known patent. And it means that all the source code is available under [http://xiph.org/flac/license.html open-source licenses]. It is the first truly open and free lossless audio format.<br />
<br />
== Features ==<br />
* '''Lossless:''' The encoding of audio (PCM) data incurs no loss of information, and the decoded audio is bit-for-bit identical to what went into the encoder. Each frame contains a 16-bit CRC of the frame data for detecting transmission errors. The integrity of the audio data is further insured by storing an MD5 signature of the original unencoded audio data in the file header, which can be compared against later during decoding or testing.<br />
* '''Fast:''' FLAC is asymmetric in favor of decode speed. Decoding requires only integer arithmetic, and is much less compute-intensive than for most perceptual codecs. Real-time decode performance is easily achievable on even modest hardware.<br />
* '''Hardware support:''' Because of FLAC's free reference implementation, low decoding complexity and popularity, FLAC has relatively widespread hardware support compared to other lossless formats.<br />
* '''Streamable:''' Each FLAC frame contains enough data to decode that frame. FLAC does not even rely on previous or following frames. FLAC uses sync codes and CRCs (similar to MPEG and other formats), which, along with framing, allow decoders to pick up in the middle of a stream with a minimum of delay.<br />
* '''Seekable:''' FLAC supports fast sample-accurate seeking. Not only is this useful for playback, it makes FLAC files suitable for use in editing applications.<br />
* '''Flexible metadata:''' New metadata blocks can be defined and implemented in future versions of FLAC without breaking older streams or decoders. Currently there are metadata types for tags, cue sheets, and seek tables. Applications can write their own APPLICATION metadata once they register an ID.<br />
* '''Suitable for archiving:''' FLAC is an open format, and there is no generation loss if you need to convert your data to another format in the future. In addition to the frame CRCs and MD5 signature, FLAC has a verify option that decodes the encoded stream in parallel with the encoding process and compares the result to the original, aborting with an error if there is a mismatch.<br />
* '''Convenient CD archiving:''' FLAC has a ''cue sheet'' metadata block for storing a CD table of contents and all track and index points. For instance, you can rip a CD to a single file, then import the CD's extracted cue sheet while encoding to yield a single file representation of the entire CD. If your original CD is damaged, the cue sheet can be exported later in order to burn an exact copy.<br />
* '''Error resistant:''' Because of FLAC's framing, stream errors limit the damage to the frame in which the error occurred, typically a small fraction of a second worth of data. Contrast this with some other lossless codecs, in which a single error destroys the remainder of the stream.<br />
<br />
== Pros ==<br />
* Portable to many systems<br />
* Open source and freely licensed<br />
* Hardware support (PhatBox, Kenwood MusicKeg, Rio Karma, etc. See below)<br />
* Streaming support<br />
* Extremely fast decoding<br />
* Supports multichannel and high resolution streams<br />
* Supports [[ReplayGain]]<br />
* Supports cue-sheet (with some limitations)<br />
* Gaining wide use as successor to [[Shorten]]<br />
<br />
== Cons ==<br />
* Compresses less efficiently than other popular modern compressors ([[Monkey's Audio]], [[OptimFROG]])<br />
* Higher compression modes slow, for little gain over the default setting.<br />
<br />
== Hardware and software that support FLAC ==<br />
For a more comprehensive list see the [http://xiph.org/flac/links.html FLAC links page].<br />
<br />
=== Hardware ===<br />
==== Car stereo ====<br />
* Kenwood [http://www.kenwood.com/cs/ce/audiofile/index.php?model=KMM KMM series]<br />
* Pioneer [http://www.pioneer.eu/eur/products/25/121/61/overview.html Car Stereo] (search FLAC)<br />
* JVC [http://mobile.jvc.com/product.jsp?pathId=139 KD-X "Digital Media Receivers" series] (almost all)<br />
* Soundstream [http://soundstream.com/store/car-video/source-units.html Source Units]<br />
* Tesla Model S<br />
* Citroën DS5<br />
<br />
==== Home stereo ====<br />
* Olive's [http://www.olive.us/ Symphony] wireless digital music center<br />
* [http://www.numark.com/ Numark]'s DJ equipment (HDX and CDX turntables, HDMIX mixer)<br />
* [http://www.sonos.com/ Sonos Digital Music System]<br />
* Slim Devices' [http://www.slimdevices.com/pi_squeezebox.html Squeezebox] networked audio players<br />
<br />
==== Portable ====<br />
* [[Apple iPod]] with [[Rockbox]] firmware<br />
* [[iAudio M3]], M5 and X5<br />
* [[iRiver]] iHP-120/iHP-140 with [[Rockbox]] firmware<br />
* [[Iwod G10]]<br />
* [[Rio Karma]]<br />
* [http://en.wikipedia.org/wiki/SanDisk_Sansa SanDisk Sansa]<br />
* TrekStor's [http://www.trekstor.de/en/products/detail_mp3.php?pid=66 Vibez]<br />
* Devices running Android 3.1+<br />
* [[Pono|Pono Player]]<br />
<br />
=== Software ===<br />
==== Players ====<br />
<br />
'''Windows'''<br />
* [[foobar2000]]<br />
* [[MediaMonkey]]<br />
* [[Winamp]]<br />
*[http://mplayerwin.sourceforge.net/ Mplayer] Console player<br />
* [http://www.cyberlink.com/products/powerdvd-ultra/features_en_US.html?&r=1 PowerDVD]<br />
* [http://www.videolan.org/ VLC]<br />
* [http://www.un4seen.com/ XMplay]<br />
<br />
'''Mac'''<br />
* [http://cogx.org/ Cog]<br />
* [http://www.videolan.org/ VLC]<br />
<br />
'''Linux'''<br />
* [http://www.clementine-player.org/ Clementine]<br />
* [http://www.mplayerhq.hu/ MPlayer]<br />
* [http://www.mythtv.org/ MythTV]<br />
* [http://www.videolan.org/ VLC]<br />
* [[XMMS]]<br />
<br />
==== Frontends (Windows) ====<br />
* FLAC frontend - [http://sourceforge.net/projects/flacfrontend/ download] / [http://wiki.hydrogenaud.io/index.php?title=Download_page discussion] (ktf)<br />
* Custom [http://members.home.nl/w.speek/flac.htm Windows Frontend] (by Speek)<br />
<br />
==== Frontends (Mac) ====<br />
* [http://www.sbooth.org/Max/ Max]<br />
<br />
==== Converters ====<br />
* [http://www.dbpoweramp.com/ dBpowerAMP] Music Converter / Audio Player / CD Writer<br />
* [http://www.mediamonkey.com/ MediaMonkey] Music Manager / Audio Player / CD Writer<br />
<br />
==== Editors ====<br />
*[http://audacity.sourceforge.net/ Audacity]<br />
* [[Adobe Audition]]<br />
* [http://www.goldwave.com/ GoldWave]<br />
<br />
==== CD writers/rippers ====<br />
* [http://www.nero.com/eng/ Nero]<br />
* [http://arson.sourceforge.net/ Arson]<br />
* [http://www.burrrn.net Burrrn] Audio CD burner<br />
* [[Exact Audio Copy]] CD Ripper<br />
* [http://cdexos.sourceforge.net CDex] CD ripper<br />
* [http://www.cdwave.com/ CD Wave]<br />
* [http://cdburnerxp.se/ CDburner XP] CD writer<br />
* [http://www.mediamonkey.com/ MediaMonkey] - CD ripper/writer<br />
<br />
==== Taggers ====<br />
* [http://www.mp3tag.de/en/index.html Mp3tag] - Universal Tag Editor<br />
* [http://www.jtclipper.eu/thegodfather/ The GodFather] - Tagger / Music manager<br />
* [http://sbooth.org/Tag/ Tag] - for Mac OS X 10.4 (Tiger)<br />
<br />
* [http://www.synthetic-soul.co.uk/tag/ Case's Tag] - Command line tagger<br />
* [https://xiph.org/flac/documentation_tools_metaflac.html metaflac] - for general metadata (including Vorbis comments) maintenance<br />
* [[MediaMonkey]] - Tagger / Music manager (Including multiple and linked album art support)<br />
<br />
==== Other tools ====<br />
* [http://www.bunkus.org/videotools/mkvtoolnix/ mkvtoolnix] - tool to multiplex FLAC streams inside the Matroska container<br />
* [https://xiph.org/flac/documentation_tools_metaflac.html metaflac] - for general metadata (including Vorbis comments) maintenance, also to calculate [[ReplayGain]] values for FLAC files lacking such<br />
<br />
for a more comprehensive list use the '''External Links''' bottom of this page to visit flac's download and link page.<br />
<br />
== Frequently asked questions ==<br />
''Question:'' Does the compression level affect decompression speed?<br />
<br />
''Short Answer'': No.<br />
<br />
''Long Answer'': In truth, the compression level does affect the decompression speed, but the difference between the various compress levels can barely be measured and is too small to be noticed, even on low-end machines.<br />
<br />
<br />
''Question:'' What is the best compression level for encoding my music?<br />
<br />
''Short Answer'': The default setting, 5.<br />
<br />
''Long Answer'': Encoding at the default setting will give the best balance between compression and encoding speed. Encoding at 8 can more than quadruple the encoding time, while having an insignificant effect on compression.<br />
<br />
== See also ==<br />
* [[Lossless]]<br />
* [[Lossless comparison]]<br />
<br />
== External links ==<br />
* [https://xiph.org/flac/ FLAC Homepage] | [https://xiph.org/flac/format.html format description] | [https://xiph.org/flac/documentation.html documentation] | [https://xiph.org/flac/faq.html FAQ] | [https://www.xiph.org/flac/download.html download]<br />
* [http://www.hydrogenaud.io/forums/index.php?showforum=67 FLAC discussion]<br />
* [http://www.hydrogenaud.io/forums/index.php?showtopic=107611 FLAC 1.3.1 discussion]<br />
* [http://www.hydrogenaud.io/forums/index.php?showtopic=107913 FLAC 1.3.1 non-SSE2 build for older CPUs]<br />
* ktf's [http://www.hydrogenaud.io/forums/index.php?showtopic=107990 Lossless codec comparison] graphs the influence of the chosen encoding level on the encoding and decoding performance of FLAC 1.3.1 and various other lossless codecs. Omion's older test, "[http://web.archive.org/web/20091108104748/http://people.ucsc.edu/~rswilson/flactest File Size vs. Decoding Speed"], covers the influence of the chosen encoding level on the decoding speed of FLAC 1.2.1.<br />
<br />
[[Category:Lossless]]<br />
[[Category:Encoder/Decoder]]</div>Gottkaiserhttps://wiki.hydrogenaud.io/index.php?title=FLAC_encoderFLAC encoder2015-05-17T13:27:28Z<p>Gottkaiser: add link to official FLAC encoder</p>
<hr />
<div>http://xiph.org/flac/</div>Gottkaiserhttps://wiki.hydrogenaud.io/index.php?title=LossyWAVLossyWAV2015-05-17T12:41:28Z<p>Gottkaiser: Add 1.4.0 release. Update links to "hydrogenaud.io"</p>
<hr />
<div>{{Software Infobox<br />
| name = lossyWAV<br />
| logo =<br />
| screenshot = <br />
| caption = <br />
| maintainer = [http://www.hydrogenaud.io/forums/index.php?showuser=42400 Nick.C]<br />
| stable_release = 1.4.0<br />
| preview_release = <none><br />
| operating_system = [[Wikipedia:Microsoft Windows|Windows]]<br />
| use = [[Wikipedia:Digital signal processing|Digital signal processing]]<br />
| license = [[Wikipedia:GNU General Public License|GNU GPL]]<br />
| website = [http://www.hydrogenaud.io/forums/index.php?showtopic=107081 1.4.0 release thread]<br />
}}<br />
lossyWAV is a [[Wikipedia:Free software|free]], [[lossy]] pre-processor for [[PCM]] audio contained in the [[RIFF_WAVE|WAV]] file format. Proposed by [http://www.hydrogenaud.io/forums/index.php?showuser=409 David Robinson], it reduces [[Wikipedia:Audio bit depth|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.<br />
lossyWAV's primary goal is to maintain [[transparency]] with a high degree of confidence when processing any audio data.<br />
<br />
==History==<br />
lossyWAV is based on the lossyFLAC idea proposed by [http://www.hydrogenaud.io/forums/index.php?showuser=409 David Robinson] at Hydrogenaudio, which is a method of carefully reducing the bitdepth of (blocks of) samples which will then allow the FLAC lossless encoder to make use of its wasted bits feature. The aim is to transparently reduce audio bit depth (by making some lower significant bits ([[Wikipedia:Least_significant_bit|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.[http://www.hydrogenaud.io/forums/index.php?s=&showtopic=55522&view=findpost&p=498179] 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.<br />
<br />
[http://www.hydrogenaud.io/forums/index.php?showuser=42400 Nick Currie] ported the original [[Wikipedia:MATLAB|MATLAB]] implementation to [[Wikipedia:Borland Delphi|Delphi]] (Many thanks [[Wikipedia:CodeGear|CodeGear]] for Turbo Explorer!) with a liberal sprinkling of [[Wikipedia:IA-32|IA-32]] and [[Wikipedia:x87|x87]] Assembly Language for speed.<br />
<br />
Subsequently, lossyFLAC proved itself to work with other lossless codecs, so the application name was changed to lossyWAV. <br />
<br />
Since then, Nick has heavily developed and built upon lossyWAV, with valuable tuning performed by [http://www.hydrogenaud.io/forums/index.php?showuser=25015 Horst Albrecht] at Hydrogenaudio. Although the current lossyWAV implementation has built on David's original method, the method itself still very much belongs to its author.<br />
<br />
==Indicative bitrate reduction==<br />
It must be stressed that lossyWAV is a pure variable bit-depth pre-processor in that the overall sample size remains the same after processing but the number of significant bits used for the samples in a codec-block can change on a block-by-block basis. 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 [[Wikipedia:fast Fourier transform|fast Fourier Transform]] (FFT) analyses of at least two lengths (default quality preset (-q 5) = 32, 64 & 1024 [[Wikipedia:Sampling %28signal processing%29|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 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. The added noise is adaptively shaped by default, however the user can select parameters to make the added noise fixed shaped or simply [[Wikipedia:white noise|white noise]]. 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.<br />
<br />
{| class="wikitable" style="text-align:center"<br />
|-<br />
!lossyWAV Test Set (16 bit / 44.1kHz)<br />
!Codec<br />
!lossless<br />
!--insane<br />
!--extreme<br />
!--high<br />
!--standard<br />
!--economic<br />
!--portable<br />
!--extraportable<br />
|-<br />
!10 Album Test Set<br />
| FLAC<br />
| 854 kbit/s<br />
| 627 kbit/s<br />
| 548 kbit/s<br />
| 477 kbit/s<br />
| 442 kbit/s<br />
| 407 kbit/s<br />
| 353 kbit/s<br />
| 311 kbit/s<br />
|-<br />
!Nick.C's Full Collection<br />
| FLAC<br />
| 882 kbit/s<br />
| -<br />
| -<br />
| -<br />
| -<br />
| -<br />
| -<br />
| 307 kbit/s<br />
|}<br />
<br />
==File identification==<br />
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.[http://www.hydrogenaud.io/forums/index.php?s=&showtopic=55522&view=findpost&p=498559]<br />
<br />
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.<br />
<br />
Combinations of lossyWAV with each specific encoder are referred to as lossy'''X''', where '''X''' is an abbreviation of the lossless codec name. Combination names are listed in the "[[LossyWAV#Known supported codecs|known supported codecs]]" section below.<br />
<br />
lossyWAV inserts a variable-length 'fact' chunk into the WAV file immediately after the 'fmt ' chunk. This takes the form:<pre>fact/<size>/lossyWAV x.y.z @ dd/mm/yyyy hh:mm:ss, -q 5</pre>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.<br />
<br />
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.<br />
<br />
==Quality presets==<br />
*--quality insane: (-q I or -q 10) Highest quality preset, generally considered to be excessive;<br />
*--quality extreme: (-q E or -q 7.5) Higher quality preset, disc space-saving alternative to lossless archiving for large audio collections, considered to be suitable for transcoding to other lossy codecs;<br />
*--quality high: (-q H or -q 5.0) High quality preset, midway between extreme and standard;<br />
*--quality standard: (-q S or -q 2.5) Default preset, generally accepted to be transparent;<br />
*--quality economic: (-q C or -q 0.0) Intermediate preset midway between standard and portable;<br />
*--quality portable: (-q P or -q -2.5) DAP quality preset for use on a compatible [[Wikipedia:Digital audio player|DAP]].[http://www.hydrogenaud.io/forums/index.php?s=&showtopic=56129&view=findpost&p=531316]<br />
*--quality extraportable: (-q X or -q -5.0) Lowest quality preset for use on a compatible [[Wikipedia:Digital audio player|DAP]].[http://www.hydrogenaud.io/forums/index.php?s=&showtopic=56129&view=findpost&p=531316]<br />
<br />
All tuning for version 1.0.0 was performed on quality preset --standard with higher presets being more conservative. For versions 1.1.0, 1.2.0 and 1.3.0, tuning effort has been focused on the lowest quality preset in an effort to achieve an effective compromise between resultant bitrate and perceived quality. Quality preset --standard is generally accepted to be (and from testing so far is) transparent. If you find a track which --standard fails to achieve transparency after processing, please post a sample (no more than 30 seconds) in the development thread.<br />
<br />
The upper frequency limit used in the calculation of minimum signal power varies, dependent on quality preset, in the range 15.159kHz to 16.682kHz<br />
<br />
==Supported input formats==<br />
*[[WAV]]: 9-bit to 32-bit integer; 1 to 8 channels; sample rate &ge; 32kHz [[Pulse Code Modulation|PCM]]. Very high sample rates (&gt;48kHz) have not been extensively tested. Tunings have been focussed on 16-bit, 44.1kHz samples (i.e. [[Wikipedia:Red Book (audio CD standard)|CD]] PCM).<br />
<br />
==Codec compatibility==<br />
{| class="wikitable" style="text-align:center"<br />
|-<br />
!Codec<br />
!Supported<br />
!Encoder parameters<br />
!Combination name<br />
|-<br />
! [[Free Lossless Audio Codec|FLAC]]<br />
| '''Yes'''<br />
| -'''5''' -'''b''' 512 --'''keep-foreign-metadata'''<br />
| lossy'''FLAC'''<br />
|-<br />
! [[Lossless Predictive Audio Compression|LPAC]]<br />
| '''Yes'''<br />
| -'''b'''512<br />
| lossy'''LPAC'''<br />
|-<br />
! [[Wikipedia:Audio Lossless Coding|MPEG-4 ALS]]<br />
| '''Yes'''<br />
| -'''l''' -'''n'''512<br />
| lossy'''ALS'''<br />
|-<br />
! [[TAK]]<br />
| '''Yes'''<br />
| -'''fsl'''512<br />
| lossy'''TAK'''<br />
|-<br />
! [[WavPack]]<br />
| '''Yes'''<br />
| --'''blocksize'''=512 --'''merge-blocks'''<br />
| lossy'''WV'''<br />
|-<br />
! [[Windows Media Audio#Windows Media Audio Lossless|WMA Lossless]]<br />
| '''Yes'''<br />
| &mdash;<br />
| lossy'''WMALSL'''<br />
|-<br />
! [[Apple Lossless]]<br />
| No<br />
| &mdash;<br />
| &mdash;<br />
|-<br />
! [[Lossless Audio|LA]]<br />
| No<br />
| &mdash;<br />
| &mdash;<br />
|-<br />
! [[Monkey's Audio]]<br />
| No<br />
| &mdash;<br />
| &mdash;<br />
|-<br />
! [[OptimFROG]]<br />
| No<br />
| &mdash;<br />
| &mdash;<br />
|-<br />
! [[Wikipedia:TTA (codec)|TTA]]<br />
| No<br />
| &mdash;<br />
| &mdash;<br />
|}<br />
<br />
* Combinations of lossyWAV with each specific encoder are referred to as lossy'''X''', where '''X''' is an abbreviation of the lossless codec name.<br />
<br />
<br />
There is also [http://www.hometheaterhifi.com/volume_8_4/dvd-benchmark-part-6-dvd-audio-11-2001.html#Meridian%20Lossless%20Packing%20(MLP)%20in%20a%20Nutshell evidence] &mdash; so-called "Bit Shifting" &mdash; to suggest that lossyWAV may work with [[Wikipedia:Meridian Lossless Packing|MLP]], but this remains untested due to prohibitive prices of encoders. At least one [http://www.hydrogenaud.io/forums/index.php?showtopic=98609&hl= commercial DVD-A] uses constant bit-depth reduction with lower bit-depth on rear channels.<br />
<br />
A comparison of portable media players is [[Wikipedia:Comparison of portable media players#Audio Formats|here]], which shows FLAC and WMA Lossless compatibility among listed players.<br />
Any player supported by [http://www.rockbox.org Rockbox] can use FLAC or WavPack files after installing Rockbox.<br />
===Important note===<br />
'''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.'''<br />
===Bonus feature===<br />
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.<br />
<br />
==Using lossyWAV==<br />
===Application settings===<br />
<pre><br />
lossyWAV 1.4.0, Copyright (C) 2007-2014 Nick Currie. Copyleft.<br />
<br />
This program is free software: you can redistribute it and/or modify it under<br />
the terms of the GNU General Public License as published by the Free Software<br />
Foundation, either version 3 of the License, or (at your option) any later<br />
version.<br />
<br />
This program is distributed in the hope that it will be useful,but WITHOUT ANY<br />
WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR A<br />
PARTICULAR PURPOSE. See the GNU General Public License for more details.<br />
<br />
You should have received a copy of the GNU General Public License along with<br />
this program. If not, see <http://www.gnu.org/licenses/>.<br />
<br />
Process Description:<br />
<br />
lossyWAV is a near lossless audio processor which dynamically reduces the<br />
bitdepth of the signal on a block-by-block basis. Bitdepth reduction adds noise<br />
to the processed output. The amount of permissible added noise is based on<br />
analysis of the signal levels in the default frequency range 20Hz to 16kHz.<br />
<br />
If signals above the upper limiting frequency are at an even lower level, they<br />
can be swamped by the added noise. This is usually inaudible, but the behaviour<br />
can be changed by specifying a different --limit (in the range 10kHz to 20kHz).<br />
<br />
For many audio signals there is little content at very high frequencies and<br />
forcing lossyWAV to keep the added noise level lower than the content at these<br />
frequencies can increase the bitrate dramatically for no perceptible benefit.<br />
<br />
The noise added by the process is shaped using an adaptive method provided by<br />
Sebastian Gesemann. This method, as implemented in lossyWAV, aims to use the<br />
signal itself as the basis of the filter used for noise shaping. Adaptive noise<br />
shaping is enabled by default.<br />
<br />
Usage : lossyWAV <input wav file> <options><br />
<br />
Example : lossyWAV musicfile.wav<br />
<br />
Quality Options:<br />
<br />
-q, --quality <t> where t is one of the following (default = standard):<br />
I, insane highest quality output, suitable for transcoding;<br />
E, extreme higher quality output, suitable for transcoding;<br />
H, high high quality output, suitable for transcoding;<br />
S, standard default quality output, considered to be transparent;<br />
C, economic intermediate quality output, likely to be transparent;<br />
P, portable good quality output for DAP use, may not be transparent;<br />
X, extraportable lowest quality output, probably not transparent.<br />
<br />
Standard Options:<br />
<br />
-C, --correction write correction file for processed WAV file; default=off.<br />
-f, --force forcibly over-write output file if it exists; default=off.<br />
-h, --help display help.<br />
-L, --longhelp display extended help.<br />
-M, --merge merge existing lossy.wav and lwcdf.wav files.<br />
-o, --outdir <t> destination directory for the output file(s).<br />
-v, --version display the lossyWAV version number.<br />
-w, --writetolog create (or add to) lossyWAV.log in the output directory.<br />
<br />
Advanced Options:<br />
<br />
- take WAV input from STDIN.<br />
-c, --check check if WAV file has already been processed; default=off.<br />
errorlevel=16 if already processed, 0 if not.<br />
-q, --quality <n> quality preset (-5.0<=n<=10.0); (-5=lowest, 10=highest;<br />
default=2.5; I=10.0; E=7.5; H=5.0; S=2.5; C=0.0; P=-2.5;<br />
X=-5.0.<br />
--, --stdout write WAV output to STDOUT.<br />
--stdinname <t> pseudo filename to use when input from STDIN.<br />
<br />
Advanced Quality Options:<br />
<br />
-a, --analyses <n> set number of FFT analysis lengths, (2<=n<=7; default=3,<br />
i.e. 32, 64 & 1024 samples. n = 2, remove 32 sample FFT;<br />
n > 3 add 16; n > 4, add 128; n > 5, add 256, n > 6, add<br />
512) n.b. FFT lengths stated are for 44.1/48kHz audio,<br />
higher sample rates will automatically increase all FFT<br />
lengths as required.<br />
--feedback [n] enable experimental bit removal / adaptive noise shaping<br />
noise limiter. Tuning has been carried out at -q X and<br />
should have a negligible effect at -q S. Optional setting<br />
(0.0 <= n <= 10.0, default = 0.0) automatically selects<br />
the following parameters (0 = least effect, 10 = most):<br />
r, round <n> limit deviation from expected added noise due to rounding<br />
(-2.0 <= n <= 2.0, default = 0.0).<br />
n, noise <n> limit added noise due to adaptive noise shaping<br />
(-2.5 <= n <= 2.5, default = 0.0).<br />
a, aclips <n> number of permissible exceedences of adaptive noise<br />
shaping level limit (0 <= n <= 64, default = 32).<br />
A, alevel <n> adaptive noise shaping level limit (-2.0 <= n <= 2.5,<br />
default = 0.0).<br />
V, verbose enable more detailed feedback information in output.<br />
-I, --ignore-chunk-sizes.<br />
ignore 'RIFF' and 'data' chunk sizes in input.<br />
-l, --limit <n> set upper frequency limit to be used in analyses to n Hz;<br />
(12500 <= n <= 20000; default=16000).<br />
--linkchannels revert to original single bits-to-remove value for all<br />
channels rather than channel dependent bits-to-remove.<br />
--maxclips <n> set max. number of acceptable clips per channel per block;<br />
(0 <= n <= 16; default = 3,3,3,3,3,2,2,2,2,2,1,1,1,0,0,0).<br />
-m, --midside analyse 2 channel audio for mid/side content.<br />
--nodccorrect disable DC correction of audio data prior to FFT analysis,<br />
default=on; (DC offset calculated per FFT data set).<br />
-n, --noskew disable application of low frequency level reduction prior<br />
to determination of bits-to-remove.<br />
--scale <n> factor to scale audio by; (0.03125 < n <= 8.0; default=1).<br />
-s, --shaping modify settings for noise shaping used in bit-removal:<br />
a, altfilter enable alternative adaptive shaping filter method.<br />
c, cubic enable cubic interpolation when defining filter shape<br />
e, extra additional white noise to add during creation of filter<br />
f, fixed disable adaptive noise shaping (use fixed shaping)<br />
n, nowarp disable warped noise shaping (use linear frequency shaping)<br />
o, off disable noise shaping altogether (use simple rounding)<br />
s, scale <n> change effectiveness of noise shaping (0 < n <= 2; default<br />
= 1.0)<br />
t, taps <n> select number of taps to use in FIR filter (32 <= n <= 256;<br />
default = 64)<br />
w, warp enable cubic interpolation when creating warped filter<br />
-U, --underlap <n> enable underlap mode to increase number of FFT analyses<br />
performed at each FFT length, (n = 2, 4 or 8, default=2).<br />
<br />
Output Options:<br />
<br />
--bitdist show distrubution of bits to remove.<br />
--blockdist show distribution of lowest / highest significant bit of<br />
input codec-blocks and bit-removed codec-blocks.<br />
-d, --detail enable per block per channel bits-to-remove data display.<br />
-F, --freqdist enable frequency analysis display of input data.<br />
-H, --histogram show sample value histogram (input, lossy and correction).<br />
--longdist show long frequency distribution data (input/lossy/lwcdf).<br />
--perchannel show selected distribution data per channel.<br />
-p, --postanalyse enable frequency analysis display of output and<br />
correction data in addition to input data.<br />
--sampledist show distribution of lowest / highest significant bit of<br />
input samples and bit-removed samples.<br />
--spread [full] show detailed [more detailed] results from the spreading/<br />
averaging algorithm.<br />
-W, --width <n> select width of output options (79<=n<=255).<br />
<br />
System Options:<br />
<br />
-B, --below set process priority to below normal.<br />
--low set process priority to low.<br />
-N, --nowarnings suppress lossyWAV warnings.<br />
-Q, --quiet significantly reduce screen output.<br />
-S, --silent no screen output.<br />
<br />
Special thanks go to:<br />
<br />
David Robinson for the publication of his lossyFLAC method, guidance, and<br />
the motivation to implement his method as lossyWAV.<br />
<br />
Horst Albrecht for ABX testing, valuable support in tuning the internal<br />
presets, constructive criticism and all the feedback.<br />
<br />
Sebastian Gesemann for the adaptive noise shaping method and the amount of<br />
help received in implementing it and also for the basis of<br />
the fixed noise shaping method.<br />
<br />
Tyge Lovset for the C++ translation initiative.<br />
<br />
Matteo Frigo and for libfftw3-3.dll contained in the FFTW distribution<br />
Steven G Johnson (v3.2.1 or v3.2.2).<br />
<br />
Mark G Beckett for the Delphi unit that provides an interface to the<br />
(Univ. of Edinburgh) relevant fftw routines in libfftw3-3.dll.<br />
<br />
Don Cross for the Complex-FFT algorithm originally used.</pre><br />
<br />
===Example drag 'n' drop batch file===<br />
Simply drag the FLAC files onto this batch file and it will process, recode in FLAC and copy ALL of the tags from the input FLAC file, placing the output lossyFLAC file in the same directory as the input FLAC file. Requires flac.exe and [http://www.synthetic-soul.co.uk/tag/ tag.exe] to be somewhere on the path. <br />
<pre>@echo off<br />
:repeat<br />
if %1.==. goto end<br />
if exist "%1" flac -d "%1" --stdout --silent|lossywav - --stdout --standard --stdinname "%1"|flac - -b 512 -o "%~dpn1.lossy.flac" --silent && tag --fromfile "%1" "%~dpn1.lossy.flac"<br />
shift<br />
goto repeat<br />
:end</pre><br />
<br />
===lossyWAV and FFTW===<br />
Since version 1.2.0, lossyWAV has been compatible with [[Wikipedia:FFTW|FFTW]] although not dependent on it. Should the user wish to take advantage of the increased processing speed available when using FFTW (from superior FFT implementations), libfftw3-3.dll should be placed in a directory on the host computer which features on the path.<br />
<br />
===Linux / OS X support: lossyWAV and WINE===<br />
The cause of lossyWAV's WINE incompatibility was found and removed during the development of 1.2.0 and retrospectively amended for 1.1.0b in a maintenance release (1.1.0c). The latest stable version (1.3.0 at the time of writing) is fully supported.<br />
<br />
[http://caudec.net/ caudec] is a command-line tool that can encode and decode lossyWAV files (lossyFLAC, lossyWV, lossyTAK), using the official binary (lossyWAV.exe) with Wine (see: [http://caudec.net/documentation/windowscodecs/ installation instructions]). Caudec can also test file integrity and compute (and tag) Replaygain data. While it hasn't been tested at the time of writing, it is possible that lossyWAV support in caudec works on OS X as well.<br />
<br />
===lossyWAV and [[foobar2000]]===<br />
Example [[foobar2000]] converter settings:<br />
<br />
lossyFLAC settings:<pre>Encoder: c:\windows\system32\cmd.exe<br />
Extension: lossy.flac<br />
Parameters: /d /c c:\"program files"\bin\lossywav - --quality standard --silent --stdout|c:\"program files"\bin\flac - -b 512 -5 -f -o%d --ignore-chunk-sizes<br />
Format is: lossless or hybrid<br />
Highest BPS mode supported: 24</pre><br />
<br />
lossyTAK settings:<pre>Encoder: c:\windows\system32\cmd.exe<br />
Extension: lossy.tak<br />
Parameters: /d /c c:\"program files"\bin\lossywav - --quality standard --silent --stdout|c:\"program files"\bin\takc -e -p2m -fsl512 -ihs - %d<br />
Format is: lossless or hybrid<br />
Highest BPS mode supported: 24</pre><br />
<br />
lossyWV settings:<pre>Encoder: c:\windows\system32\cmd.exe<br />
Extension: lossy.wv<br />
Parameters: /d /c c:\"program files"\bin\lossywav - --quality standard --silent --stdout|c:\"program files"\bin\wavpack -hm --blocksize=512 --merge-blocks -i - %d<br />
Format is: lossless or hybrid<br />
Highest BPS mode supported: 24</pre><br />
<br />
lossyWMALSL* settings:<pre>Encoder: c:\windows\system32\cmd.exe<br />
Extension: lossy.wma<br />
Parameters: /d /c c:\"program files"\bin\lossywav - --quality standard --silent --stdout|c:\"program files"\bin\wmaencode.exe - %d --codec lsl --ignorelength<br />
Format is: lossless or hybrid<br />
Highest BPS mode supported: 24</pre><br />
<br />
Enclose the element of the path containing spaces within double quotation marks ("), e.g. C:\"Program Files"\directory_where_executable_is\executable_name. This is a Windows limitation.<br />
<br />
lossyWMALSL conversion uses WMAEncode.exe by lvqcl found [http://www.hydrogenaud.io/forums/index.php?s=&showtopic=90519&view=findpost&p=767754 here].<br />
<br />
===lossyWAV and EAC===<br />
:''For example settings, see [[EAC and LossyWAV]].''<br />
<br />
==Frequently asked questions==<br />
*'''Question:''' Why is the ".wav" file extension used?<br />
*'''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.<br />
<br />
*'''Question:''' Why create a processor which means that I cannot be sure that a lossless file is truly lossless?<br />
*'''Answer:''' Unless one creates the lossless file personally, one can '''never''' be completely sure that the file is indeed lossless. E.g. a lossless file you receive could be transcoded from [[MP3]] without your knowledge. To distinguish a lossyWAV file from lossless files it is recommended to use the extension .lossy.EXT where EXT is the original extension e.g. .lossy.flac<br />
<br />
*'''Question:''' Is it [[Variable Bitrate|VBR]]?<br />
*'''Short answer:''' Yes.<br />
<br />
*'''Question:''' Do I need to re-process to change lossless codecs?<br />
*'''Short answer:''' No.<br />
<br />
*'''Question:''' Is it [[transparency|transparent]]?<br />
*'''Short answer:''' At preset --standard, almost certainly.<br />
<br />
*'''Question:''' Is it [[lossless]]?<br />
*'''Short answer:''' No.<br />
<br />
*'''Question:''' Will it ever have a [[Constant Bitrate|CBR]] mode?<br />
*'''Short answer:''' No.<br />
<br />
*'''Question:''' Will it low-pass filter my audio?<br />
*'''Short answer:''' No. The frequency limit is for the analysis only. LossyWAV cannot low-pass filter your audio.<br />
<br />
*'''Question:''' Why should I use this?<br />
*'''Answer:'''<br />
:*high quality<br />
:*extremely low chance of audible [[artifact]]s<br />
:*reasonable [[bitrate]]s<br />
:*usable with unmodified, established lossless formats.<br />
<br />
==External links==<br />
*[http://www.hydrogenaud.io/forums/index.php?showtopic=55522 Original lossyFLAC thread] - Introduction of the concept by David Robinson (Replay Gain developer) and initial development<br />
----<br />
*[http://www.hydrogenaud.io/forums/index.php?showtopic=107081 lossyWAV 1.4.0 release thread] - Release of version 1.4.0 on 02 Oktober 2014<br />
----<br />
*[http://www.hydrogenaud.io/forums/index.php?showtopic=96635 lossyWAV 1.3.1 Delphi to C++ translation thread]<br />
----<br />
*[http://www.hydrogenaud.io/forums/index.php?showtopic=81002 lossyWAV 1.3.0 development thread]<br />
*[http://www.hydrogenaud.io/forums/index.php?showtopic=90104 lossyWAV 1.3.0 release thread] - Release of version 1.3.0 on 06 August 2011<br />
----<br />
*[http://www.hydrogenaud.io/forums/index.php?showtopic=65499 lossyWAV 1.2.0 development thread]<br />
*[http://www.hydrogenaud.io/forums/index.php?showtopic=77042 lossyWAV 1.2.0 release thread] - Release of version 1.2.0 on 16 December 2009<br />
----<br />
*[http://www.hydrogenaud.io/forums/index.php?showtopic=63254 lossyWAV 1.1.0 development thread]<br />
*[http://www.hydrogenaud.io/forums/index.php?showtopic=64617 lossyWAV 1.1.0 release thread] - Release of version 1.1.0 on 12 July 2008<br />
----<br />
*[http://www.hydrogenaud.io/forums/index.php?showtopic=56129 lossyWAV Development thread] - Conversion of the original MATLAB script to Delphi and evolution of the method<br />
*[http://www.hydrogenaud.io/forums/index.php?showtopic=63225 lossyWAV 1.0.0 release thread] - Release of version 1.0.0b on 12 May 2008<br />
<br />
[[Category:Software]]</div>Gottkaiserhttps://wiki.hydrogenaud.io/index.php?title=PowerShell_AudioPowerShell Audio2014-12-19T10:51:34Z<p>Gottkaiser: initial publication</p>
<hr />
<div>{{title|PowerShell Audio}}<br />
<br />
{{Software Infobox<br />
| name = PowerShell Audio<br />
| logo =<br />
| screenshot = [[Image:PowerShellAudio-logo.png|200px|PowerShellAudio Logo]]<br />
| caption = audio conversion and tagging module for Windows PowerShell<br />
| maintainer = [https://github.com/jherby2k/ Jebus]<br />
| stable_release = [https://github.com/jherby2k/PowerShellAudio/releases/latest 1.0.0] (12 Dezember 2014)<br />
| operating_system = Windows<br />
| use = Digital Audio conversion, tagging<br />
| license = GNU GPLv3<br />
| website = [https://github.com/jherby2k/PowerShellAudio GitHub project]<br />
}}<br />
<br />
=Introduction= <br />
PowerShell Audio is a PowerShell-driven interface for converting, tagging and analyzing audio / music files. It is available under a GPLv3 license and is for Windows use only.<br />
<br />
== Features ==<br />
* A unified interface to popular codecs (see Supported Formats).<br />
* Fast and highly concurrent ("multi-threaded") for modern, multi-core systems.<br />
* ReplayGain integration.<br />
** Batch-analyze thousands of files quickly,<br />
** save the results to disk,<br />
** convert to Apple SoundCheck format,<br />
** or apply the changes directly during encoding.<br />
* Metadata preservation between formats.<br />
* PowerShell interface brings powerful integration and scripting capabilities.<br />
* API is open-source and extensible.<br />
<br />
== Supported Formats ==<br />
* [[FLAC]]<br />
* [[Ogg_Vorbis|Ogg Vorbis]]<br />
* [[MP3]] ([[LAME]])<br />
* [[Apple AAC]]<br />
* [[WAV]]<br />
<br />
== Supported languages ==<br />
* English<br />
<br />
== Example Usage ==<br />
'''simple Examples'''<br /><br />
Convert a folder full of FLAC files into Lame MP3s:<br />
Get-AudioFile *.flac | Export-AudioFile "Lame MP3" -Directory C:\Output<br />
<br /><br />
Lame uses VBR -q 3 by default. To list the default switches:<br />
Get-AudioEncoderDefaultSettingList "Lame MP3"<br />
<br /><br />
To get all the available settings for Lame:<br />
Get-AudioEncoderAvailableSettingList "Lame MP3"<br />
<br /><br />
All the available cmdlets are documented. For examples and a list of parameters use:<br />
Get-Help Export-AudioFile -Full<br />
<br /><br />
'''advanced Examples'''<br /><br />
Add ReplayGain to a entire FLAC library and treat each directory as a separate album:<br />
Get-ChildItem C:\Users\Myself\Music -Directory -Recurse | % { $_ | Get-ChildItem -File -Filter *.flac | Measure-AudioFile ReplayGain -PassThru | Save-AudioMetadata }<br />
<br /><br />
Convert a whole FLAC library to VBR AAC, with SoundCheck tags calculated from album ReplayGain information:<br />
Get-ChildItem C:\Users\Myself\Music -Filter *.flac -Recurse | Get-AudioFile | Export-AudioFile "Apple AAC" -Directory "C:\Output\{Artist}\{Album}" -Setting @{AddSoundCheck = "Album"} -Name "{TrackNumber} - {Title}"<br />
<br /><br />
Convert a whole FLAC library to VBR MP3, with ReplayGain directly applied to the resulting volume levels:<br />
Get-ChildItem C:\Users\Myself\Music -Filter *.flac -Recurse | Get-AudioFile | Export-AudioFile "Lame MP3" -Directory "C:\Output\{Artist}\{Album}" -Setting @{ApplyGain = "Album"} -Name "{TrackNumber} - {Title}"<br />
<br />
== Externals links ==<br />
* [https://github.com/jherby2k/PowerShellAudio/wiki GitHub Wiki]<br />
* [https://wikipedia.org/wiki/Windows_PowerShell PowerShell Overview]<br />
<br />
<br />
[[Category:Software]]</div>Gottkaiserhttps://wiki.hydrogenaud.io/index.php?title=File:PowerShellAudio-logo.pngFile:PowerShellAudio-logo.png2014-12-19T09:53:57Z<p>Gottkaiser: </p>
<hr />
<div></div>Gottkaiserhttps://wiki.hydrogenaud.io/index.php?title=Fre:acFre:ac2013-06-19T16:15:53Z<p>Gottkaiser: updated and deleted outdated information</p>
<hr />
<div>{{title|fre:ac}}<br />
<br />
{{Software Infobox<br />
| name = fre:ac<br />
| logo =<br />
| screenshot = [[Image:BonkEnc-screenshot.png|250px|fre:ac screenshot]]<br />
| caption = Slim CD ripper, audio encoder and converter <br />
| maintainer = Robert Kausch<br />
| stable_release = 1.0.20a (26 March 2013)<br />
| preview_release = snapshot-20130430 (30 April 2013)<br />
| operating_system = Windows<br />
| use = Digital Audio Extraction, transcoding<br />
| license = GPL <br />
| website = [http://www.freac.org http://www.freac.org]<br />
}}<br />
<br />
=Introduction=<br />
'''fre:ac''' ('''former''' BonkEnc ) is a CD ripper, audio encoder and converter for various formats. It's a very slim program and can be run from a USB stick. Fre:ac is available under GPL license and has native support for numerous languages.<br />
<br />
== Features ==<br />
* CD ripping<br />
** [[Cdparanoia]] mode<br />
** jitter correction<br />
** CD Text support<br />
* [[Transcoding]] from on to another format<br />
* using Compact Disc Database (CDDB)<br />
* support [[ID3v1]], [[ID3v2]], MP4-Metadata and [[Vorbis_comment|Vorbis comment]] [[Tags]]<br />
* keeps image tags when converting from FLAC to MP3<br />
* creating cue sheets and playlists<br />
* full UTF-8 Unicode support<br />
* additional command line interface (CD Ripping/Encoding)<br />
<br />
== Supported Formats ==<br />
* [[MP3]] ([[LAME]])<br />
* [[MP4/M4A AAC]] ([[FAAC]]/[[FAAD|FAAD2]])<br />
* [[Ogg_Vorbis|Ogg Vorbis]]<br />
* [[FLAC]]<br />
* [http://www.logarithmic.net/pfh/bonk Bonk] v0.11<br />
Supported by the development snapshot:<br />
* [[Opus]]<br />
* [[Speex]]<br />
* Apple Lossless ([[ALAC]])<br />
* [[Musepack]]<br />
* [[WavPack]]<br />
* [[Monkey%27s_Audio|Monkey's Audio]]<br />
* [[Tak|TAK]]<br />
<br />
== Supported languages ==<br />
{| border="0" -valign="top"<br />
|-valign="top"<br />
||<br />
* Catalan<br />
* Chinese<br />
* Czech<br />
* Danish<br />
* Dutch<br />
* English<br />
* Esperanto<br />
* Finnish<br />
||<br />
* French<br />
* German<br />
* Greek<br />
* Hungarian<br />
* Italian<br />
* Japanese<br />
* Korean<br />
* Lithuanian<br />
||<br />
* Polish<br />
* Portuguese<br />
* Russian<br />
* Serbian<br />
* Slovak<br />
* Spanish<br />
* Swedish<br />
* Turkish<br />
||<br />
* Ukrainian<br />
* Romanian<br />
* ...<br />
|}<br />
<br />
[[Category:CD Rippers]]<br />
[[Category:Software]]</div>Gottkaiserhttps://wiki.hydrogenaud.io/index.php?title=OpusOpus2013-06-19T15:32:45Z<p>Gottkaiser: /* Player software */</p>
<hr />
<div>{{Software Infobox<br />
| name = Opus<br />
| logo = [[Image:opus-logo.png|250px|Official Opus logo]]<br />
| screenshot =<br />
| caption = Opus Interactive Audio Codec<br />
| maintainer = [http://xiph.org/ Xiph.Org Foundation]<br />
| stable_release = 1.0.2<br />
| preview_release = exp_analysis7<br />
| operating_system = Windows, Mac OS/X, Linux/BSD<br />
| use = Encoder/Decoder<br />
| license = 3-clause BSD license<br />
| website = [http://www.opus-codec.org/ opus-codec.org]<br />
}}<br />
<br />
'''Opus''' is a [[lossy]] audio compression format developed by the Internet Engineering Task Force (IETF) designed to be suitable for interactive real-time applications over the Internet,{{ref|homepage|a}} including music as well as speech, yet it is also very competitive for use as a storage and playback format, being a [http://people.xiph.org/~greg/opus/ha2011/ class leader at around 64 kbps]. As an open format standardised through [http://tools.ietf.org/html/rfc6716 Request for Comments (RFC) 6716],{{ref|RFC|c}} a high quality reference implementation is provided under the 3-clause BSD license{{ref|homepage|a}} which compiles and runs on the vast majority of general purpose and embedded (fixed point) processors. Many Software patents which cover Opus are licensed under royalty-free terms.{{ref|FAQ|b}} Opus is also a Mandatory To Implement (MTI) codec for the upcoming WebRTC (Web Real Time Communication) specification of the World Wide Web Consortium (W3C).<br />
<br />
Opus incorporates technology from two codecs, the speech-oriented SILK codec developed by Skype and the multi-purpose low-latency CELT codec developed by Xiph.org with significant changes to each to ensure they can work together.{{ref|RFC|c}} Opus can seamlessly transition among high and low bitrates, using a linear prediction codec (the SILK layer) at lower bitrates and a lapped transform codec (the CELT layer) at higher bitrates, as well as a hybrid of the two for a short overlap in which SILK encodes the 0-8kHz spectrum and the CELT layer encodes only the frequencies above 8kHz.{{ref|RFC|c}} Opus has very low algorithmic delay (typ 22.5 ms) compared to popular music formats such as [[MP3]], [[Vorbis |Ogg Vorbis]], [[AAC | LC-AAC and HE-AAC]] (all over 100 ms), yet performs very competitively with them in terms of quality per bitrate, making it comparably viable as a storage & playback format. Also unlike Vorbis, Opus does not require the definition of large codebooks for each individual file, making it also preferable for short clips of audio, such as those often used by game developers, a field where patent-free Vorbis is commonly used.{{ref|RFC|c}}<br />
<br />
Considerably more details of the history and potential applications for Opus are included in the ''Wikipedia'' page for '''[http://en.wikipedia.org/wiki/Opus_%28audio_format%29 Opus (audio format)]'''<br />
<br />
==Characteristics==<br />
Opus supports bitrates from 6kbps to 510kbps for typical stereo audio sources (and a maximum of around 255 kbps per channel for multichannel audio), with the 'sweet spot' for music and general audio around 30kbps (mono) and 40-100 kbps (stereo). It is intrinsically [[VBR | variable bitrate]], though constrained VBR and [[CBR | constant bitrate]] modes are possible where required. In the case of the reference release, libopus, the target bitrate is calibrated against the internal constant quality targets so that over a typical music collection, something very close to the target bitrate will be achieved. This bitrate-calibrated approach differs from most VBR encoders (e.g. LAME, helix mp3, qaac, Nero aacenc, Ogg Vorbis, Musepack) where a setting on some 'constant quality' scale (which differs between encoders) is used and the bitrate will fall where it may. Improved future versions can be expected to offer improved quality at the same setting. Independent implementations may adopt a different approach.<br />
<br />
Opus is able to seamlessly adapt its mode of operation without glitches or sound interruption (an illustrative demonstration of [http://opus-codec.org/examples/#gauge bitrate scalability] is on the Opus Examples page), which can be particularly useful for mixed-content audio or varying network conditions, making the unified Opus codec superior to a suite of different codecs that might otherwise cover the same range of bitrate and quality settings and would require out-of-band signalling to instigate codec switching. The switching includes the choice of mono, stereo and other channel mappings, the use of the speech-oriented SILK layer, the general-purpose CELT layer or the hybrid of both, and the use of different audio bandwidths (4kHz, 6kHz, 8kHz, 12kHz, 20kHz) as well as the quality adjustments within the same operating mode that are available in most VBR-capable codecs.<br />
<br />
Of importance mainly to interactive uses, but potentially useful in time-delayed audio streaming also, Opus includes packet loss concealment (PLC) in all modes and, in the speech-oriented modes where the SILK layer is active it also supports Forward Error Correction (FEC) where the expected rate of packet loss can be indicated to the encoder by the user or by application software and critical frames (e.g. consonant sounds) can be retransmitted at low bitrate to preserve intelligibility.<br />
<br />
For music and general audio, the CELT layer of Opus builds on knowledge gained during xiph.org's Vorbis development and ensures as a primary goal that the total energy in each spectral band is preserved while requiring only a modest bitrate overhead to achieve this, thereby eliminating a lot of bitrate-starvation artifacts such as 'birdies' that are common in low-bitrate MP3, especially during transients, applause and cymbal sounds. This technique likewise increases coding efficiency at bitrates targetting transparent music reproduction. Short blocks (2.5 ms) are also possible for efficient transient handling. Short blocks can also be used exclusively, if very low algorithmic delay (5.0ms) is required to enable very low-latency interative audio (e.g. live networked music performances such as remote jam sessions), though greater bitrate is then required to maintain the same quality (illustrated in [http://people.xiph.org/~xiphmont/demo/celt/demo.html#demo Monty's CELT demo page] under Constant PEAQ value, varying latency). CELT uses a number of additional techniques and provides additional advanced tools to enable encoder tuning.<br />
<br />
Opus natively supports [[gapless playback]] (though [[Gapless_playback#Poorly_designed_playback_systems | poor player design]] might itself induce interruptions during playback). Playback gain is also required, making some form of [[ReplayGain]] or [[ReplayGain_2.0_specification | similar]] volume control possible in any compliant player.<br />
<br />
==Bitrate performance==<br />
For mono speech, Opus ranges from intelligible narrowband speech reproduction starting at 6 kbps to medium-band, wideband and superwideband speech, reaching full-band speech by around 32 kbps. Above about 32 kbps, the SILK layer is no longer used at all, as CELT alone gives superior quality.<br />
<br />
For music, the SILK modes are quite tolerable and better than CELT at very low bitrates. The hybrid mode is adopted as bitrate increases, extending bandwidth first to 12kHz (comparable with compact cassette) then to the full 20kHz and CELT then takes over. Assuming the source is stereo, the transition from mono to stereo typically happens between the transition from 12kHz to 20kHz.<br />
<br />
==Indicative bitrate and quality==<br />
The table below gives illustrative, indicative quality guidance based on typical modes used internally by Opus and a range of listening tests.<br />
<br />
In the experimental libopus version 1.1-alpha, automatic detection of speech/music and bandwidth detection have been introduced to improve mode decisions, and VBR is less constrained, all with the aim of maximizing the quality/bitrate tradeoff. Thus changes are likely, and this table is likely to require small updates as the encoder is improved.<br />
<br />
===Speech encoding quality===<br />
This table assumes a '''monophonic''' source sampled at CD quality or above (typ 48 kHz sampling rate) but mentions stereo compatibility for 40kbps+. The default 20ms frame size (22.5ms latency) is assumed.<br />
<br />
{| class="wikitable" style="text-align:center"<br />
|-<br />
!Bitrate target<br />
!Bandwidth<br />
!typ SILK/CELT use<br />
!Speech quality notes<br />
!Use cases/notes/competitive codecs<br />
|-<br />
!1 to 5 kbps<br />
| -<br />
| -<br />
| <6kbps bitrate not supported<br />
| Try [http://codec2.org/ codec2] for 1.2-2.4 kbps speech<br />
|-<br />
!6 kbps<br />
|4 kHz<br />
|SILK<br />
|Fair, intelligible<br />
|AMR-NB may be a little better, but higher latency & proprietary, Speex also competitive<br />
|-<br />
!8 kbps<br />
|4 kHz narrowband<br />
|SILK<br />
|Close to telephone quality<br />
|AMR-NB & AMR-WB similar quality, but higher latency & proprietary. Speex competitive.<br />
|-<br />
!12 kbps<br />
|6 kHz medium-band<br />
|SILK<br />
|Medium bandwidth, better than telephone quality<br />
|Similar quality to AMR-WB<br />
|-<br />
!16 kbps<br />
|8 kHz wideband<br />
|SILK<br />
|Wideband speech quality<br />
|Similar to/better than AMR-WB<br />
|-<br />
!24 kbps<br />
|12 kHz super-wideband<br />
|hybrid<br />
|Near transparent speech<br />
|Better than AMR-WB. Podcasts/audiobooks/talk-radio.<br />
|-<br />
!32 kbps<br />
|20 kHz<br />
|hybrid / possibly CELT<br />
|Essentially transparent speech plus moderately good mono music<br />
|Much better than AMR-WB. Podcasts/audiobooks/talk-radio.<br />
|-<br />
!40 kbps<br />
|20 kHz<br />
|CELT<br />
|Essentially transparent mono or stereo speech, fairly good stereo music<br />
|Stereo podcasts/audiobooks/talk radio with some music<br />
|-<br />
!48 kbps+<br />
|20 kHz<br />
|CELT<br />
|Essentially transparent mono or stereo speech, reasonable music<br />
|Flexible general purpose modes to suit mixed music and speech<br />
|-<br />
|}<br />
<br />
===Music encoding quality===<br />
This table assumes a '''stereophonic''' source sampled at CD quality or above (typ 48 kHz sampling rate). Opus will automatically use mono at very low bitrates, though a certain amount of stereo encoding can still be used - content dependent even when mono is specified as the typical stereo mode in the table below.<br />
<br />
{| class="wikitable" style="text-align:center"<br />
|-<br />
!Bitrate target<br />
!Stereo mode<br />
!Bandwidth<br />
!typ SILK/CELT use<br />
!Music quality notes<br />
!Use cases/notes/competitive codecs<br />
|-<br />
!6 kbps<br />
|mono<br />
|4 kHz<br />
|SILK<br />
|Poor, muffled sound but intelligible lyrics.<br />
| -<br />
|-<br />
!8 kbps<br />
|mono<br />
|4 kHz<br />
|SILK<br />
|Poor, muffled but OK for bitrate<br />
| -<br />
|-<br />
!14 to 16 kbps<br />
|mono<br />
|6 kHz<br />
|SILK<br />
|Fairly Poor but OK for bitrate<br />
|Perhaps acceptable for incidental music<br />
|-<br />
!22 to 24 kbps<br />
|mono<br />
|8 kHz<br />
|SILK<br />
|Fair but OK for bitrate<br />
|OK for incidental music<br />
|-<br />
!32 kbps<br />
|mono<br />
|12 kHz<br />
|hybrid<br />
|Moderately good mono, reasonably bright treble (c.f. mono cassette)<br />
|Good for podcasts, audiobooks, CELT-only poss for music. Competitor HE-AAC@32kbps is stereo full-band but with annoying artifacts.<br />
|-<br />
!39 to 40 kbps<br />
|stereo<br />
|12 kHz<br />
|hybrid/CELT<br />
|Moderately good stereo, reasonably bright treble (c.f. stereo cassette)<br />
|Stereo podcasts, audiobooks, very low bitrate music<br />
|-<br />
!48 kbps<br />
|stereo<br />
|20 kHz<br />
|CELT<br />
|Full bandwidth stereo music, some artifacts, rarely nasty<br />
|Stereo podcasts, audiobooks, low bitrate music<br />
|-<br />
!64 kbps<br />
|stereo<br />
|20 kHz<br />
|CELT<br />
|Full bandwidth stereo music, nice sound, detectable differences to original (mostly 'not annoying')<br />
|Music storage & streaming. Beat HE-AAC, Vorbis, MP3 in [http://people.xiph.org/~greg/opus/ha2011/ listening test]<br />
|-<br />
!96 kbps<br />
|stereo<br />
|20 kHz<br />
|CELT<br />
|Full bandwidth stereo music, good quality approaching transparency<br />
|Music storage & high quality streaming.<br />
|-<br />
!112 kbps<br />
|stereo<br />
|20 kHz<br />
|CELT<br />
|Fairly close to transparency (needs more testing)<br />
|Music storage & high quality streaming. Very low-latency stereo networked music performance/jam sessions at OK quality (see below table)<br />
|-<br />
!128 kbps<br />
|stereo<br />
|20 kHz<br />
|CELT<br />
|Very close to transparency (needs more testing). Most modern codecs competitive (AAC-LC, Vorbis, MP3)<br />
|Music storage & streaming. Future download music sales.<br />
|-<br />
!256 kbps<br />
|stereo<br />
|20 kHz<br />
|CELT<br />
|Transparent with very low chance of artifacts (a few killer samples still detectable). Most old & new lossy codecs competitive.<br />
|Music storage & streaming, dedicated limited-bandwidth audio links (e.g. wireless, [http://en.wikipedia.org/wiki/Bluetooth_profile#Advanced_Audio_Distribution_Profile_.28A2DP.29 A2DP-bluetooth] type links). <br />
|-<br />
!510 kbps<br />
|stereo<br />
|20 kHz<br />
|CELT<br />
|Maximum possible stereo bitrate target (actual rate often less than 510 for default frame size). Most old and new lossy codecs competitive, plus near-lossless [[lossyWAV]] and [[WavPack | WavPack lossy]]<br />
|Music storage, dedicated limited-bitrate audio links (e.g. wireless, minimum latency high quality audio. LossyWAV and WavPack lossy are very competitive for storage, and WavPack lossy --blocksize=256 may be competitive with minimum latency mode also.<br />
|-<br />
!>510 kbps<br />
| -<br />
| -<br />
| -<br />
|Above Opus bitrate range allowed for stereo sources<br />
|Settle for 510kbps or use [[lossless]], [[lossyWAV]], [[WavPack | WavPack lossy]] or lossy transform/subband codecs like [[Vorbis]], [[Musepack]] at very high settings.<br />
|-<br />
|}<br />
<br />
===Lower latency versus quality/bitrate trade-off===<br />
====Packet overhead in interactive applications====<br />
For interactive use on the Internet or other packet-based networks, total bandwidth used will be subject to packet overhead. The more packet headers that are transmitted every second, the greater will be the overhead that is required. For this reason, Opus, while defaulting to 20.0ms frames, supports 60.0ms frames to reduce overhead when transporting low-bitrate SILK frames at the expense of greater latency, which may still be acceptable for speech, and also supports 10.0ms SILK frames to reduce latency somewhat at the expense of packet overhead.<br />
<br />
In the CELT layer, which tends to operate at higher bitrates than SILK, 20.0ms frames are the default, but frames of 10.0ms, 5.0ms and 2.5ms are also possible, which directly increases the frame overhead by transmitting more packets per second to achieve lower latency. In addition, as we'll see below it also reduces the quality/bitrate tradeoff of the CELT layer itself.<br />
<br />
None of the bitrates mentioned in this article account for the packet overhead.<br />
<br />
====CELT layer latency versus quality/bitrate trade-off====<br />
Unlike the SILK layer, which works on fixed 10.0ms blocks, 1, 2 or 6 of which can be combined into an Opus frame, the CELT layer is able to modify the encoding block lengths available to enable its use with shorter frames.<br />
<br />
When the CELT layer uses 10.0ms, 5.0ms and 2.5ms frames instead of the default 20.0ms, it must use smaller transform block sizes to achieve this, thereby reducing frequency resolution in the MDCT compared to the default transform window, thus reducing encoding efficiency for tonal signals. To obtain the same frequency precision for a sound divided into shorter transform windows, improved amplitude precision is necessary, resulting in increased bitrate to obtain the same perceptual quality (or conversely lower quality at the same bitrate).<br />
<br />
These reduced-latency modes remain efficient for transient signals, which use short blocks anyway.<br />
<br />
In all modes, the algorithmic delay consists of the frame size plus an additional 2.5ms delay. The CELT layer requires 2.5ms for MDCT window overlap.<br />
<br />
Xiph.org used matched PEAQ scores (approximate perceptual quality assessment made in software) for the CELT0.10 codec that was used as the basis of the CELT layer in the Opus reference release, which indicate the following [http://people.xiph.org/~xiphmont/demo/celt/demo.html#demo approximate equivalent settings] for stereo music.<br />
<br />
{| class="wikitable" style="text-align:center"<br />
|-<br />
!Frame size<br />
!Algorithmic delay<br />
!Bitrate to match 64kbps@22.5ms delay<br />
!fractional bitrate increase<br />
|-<br />
!20.0 ms<br />
|22.5 ms<br />
|64.0 kbps<br />
|0.0 %<br />
|-<br />
!10.0 ms<br />
|12.5 ms<br />
|70.4 kbps<br />
|10.0 %<br />
|-<br />
!5.0 ms<br />
|7.5 ms<br />
|84.8 kbps<br />
|32.5 %<br />
|-<br />
!2.5 ms<br />
|5.0 ms<br />
|112.0 kbps<br />
|75.0 %<br />
|-<br />
|}<br />
<br />
N.B. This table is useful for interactive streaming only. For music storage & delayed playback or non-interactive streaming, latency reduction is not important and the default 20.0ms frame size is preferable.<br />
<br />
== Hardware & Software Support ==<br />
<br />
Much of this section is based heavily on the Jan 12th 2013 version of the '''Support''' section of the [http://en.wikipedia.org/wiki/Opus_%28audio_format%29 Wikipedia article], which is more likely to be kept updated and to provide links to further information about the supporting platforms.<br />
<br />
The format and algorithms are openly documented and the reference implementation is published as free software. The reference implementation (Opus Audio Tools, opus-tools), consisting of separate encoders and decoders, is published under the terms of a BSD-like license. It is written in C programming language and can be compiled for hardware architectures with or without floating point unit. The accompanying diagnostic tool opusinfo reports detailed technical information about Opus files, including information on the standard compliance of the bitstream format. It is based on ogginfo from the vorbis-tools and therefore, unlike the encoder and decoder, available under the terms of version 2 of the GPL.<br />
<br />
=== Commandline binaries & libopus versions ===<br />
The commandline tools of the reference version are available pre-compiled for the most popular operating systems at [http://opus-codec.org/downloads opus-codec.org] and [https://ftp.mozilla.org/pub/mozilla.org/opus/ Mozilla's ftp server]. No other implementations of opus are currently known. The libopus commandline tools include encoder ''opusenc'', decoder ''opusdec'', and with a different license, the ''opusinfo'' opus stream & metadata analyzer.<br />
<br />
The '''latest stable release''' is recommended for general use and as of early 2013 is considered competitive with or superior to the best alternative speech or general music encoders at most supported bitrates.<br />
<br />
==== libopus v1.0 (recommended latest stable release) ====<br />
Released 11 Sep 2012 when RFC6716 was standardized but mostly fully developed by late 2011.<br />
<br />
'''Stable''', '''well-tuned''' ''opusenc'' reference encoder as included in RFC documentation.<br />
<br />
CELT layer closely related to CELT 0.10 implements Constrained VBR mode by default (bitrate boost used mainly for transients), plus true CBR.<br />
<br />
==== libopus v1.1-alpha ====<br />
Source code released 21 Dec 2012 for testing & user feedback ([https://ftp.mozilla.org/pub/mozilla.org/opus/win32/opus-tools-0.1.6-opus-1.1-alpha-win32.zip win32 binaries]), but not yet considered stable and well tested enough for general release.<br />
<br />
CELT layer [http://jmspeex.livejournal.com/11737.html quality improvements] introduced to provide '''unconstrained VBR''' include a rate boost not just for transients but now for highly tonal signals too and rate reduction when stereo image is narrow. There's also a rewrite of its '''transient detection''' code and '''time-frequency analysis''' code, and rewritten '''dynamic allocation''' code (HF/LF tilt and Band Boost) to allow more aggressive changes from the typical static allocation when warranted.<br />
<br />
There are many minor improvements to '''speech quality''' in both SILK and CELT layers.<br />
<br />
'''DC-rejection''' below 3 Hz also aids quality if inaudible DC offset is present with no effect on deep bass notes.<br />
<br />
'''Automatic speech/music detection''' is introduced to optimize encoding mode choices, especially near the bitrate target range (presumably around 24~40kbps) where the encoder may perform best with SILK, hybrid or CELT depending on content type. Below that range SILK performs best for both music & speech, and above it CELT performs best for speech & music. The detection, without look-ahead, takes a second or two typically and will sometimes make incorrect decisions. The developers would be keen to know of examples of its failure.<br />
<br />
'''Automatic bandwidth detection''' is also introduced to save wasted bits allocated to absent frequencies, and while easier to implement, developers would also been keen to know of any failure of this feature (potentially caused by aliasing, quantization and dithering/noise-shaping in source material).<br />
<br />
=== VoIP software ===<br />
* The voice-chat software Mumble supports Opus as its main codec.<br />
* SIP softphones Phoner and PhonerLite support Opus<br />
* The SIP and IAX2 client SFLphone is being fitted with Opus support.<br />
* Integration of Opus into the Skype client is finished, although no version with Opus support has yet been published.<br />
* TrueConf video conferencing solutions support Opus.<br />
* Opus support is planned for Jitsi 2.0, together with VP8 video<br />
* Empathy may use any format supported in GStreamer, including Opus.<br />
* Line2 has replaced their current codec with Opus. Their iOS app will be the first to be released with the Opus. The Android app will follow later.<br />
* CSipSimple supports Opus, Codec2, G.726 and G.722.1 with an additional plug-in.<br />
* The voice-chat software TeamSpeak 3 supports Opus for voice and music in pre-release server 3.0.7-pre2 and beta client version 3.0.10<br />
<br />
=== Web frameworks and browsers ===<br />
* Opus support is mandatory for WebRTC implementations.<br />
* Mozilla supports Opus beginning with version 15 of Firefox and Thunderbird, plus Seamonkey, which is uses shared codebase.<br />
* Depending on the backend in use, Opera supports inline playback of embedded Opus files. Official support for Opus and WebRTC are on the development roadmap.<br />
* Chromium and Google Chrome will have audio support as of version 25.<br />
* Maxthon Cloud Browser<br />
<br />
=== Streaming audio ===<br />
* Icecast. (examples: [http://dir.xiph.org/ Stream directory], [http://smj.delfa.net/opus_64.m3u 64k]/[http://smj.delfa.net/opus_256.m3u 256k] [http://smj.delfa.net/ Smooth Jazz Opus Stream], [http://www.absoluteradio.co.uk/listen/labs.html Absolute Radio Opus Trial] 7 stations at 24,64,96 kbps, [http://icecast.ofdoom.com:8000/burst-opus.ogg Icecast Of Doom 96k]<br />
* Krad Radio<br />
* Liquidsoap<br />
<br />
=== Operating systems and desktop multimedia frameworks ===<br />
* In Debian GNU/Linux the Opus development tools and supporting libraries can be installed from the preconfigured repositories in the next stable version ("wheezy") that is expected to be released in early 2013.<br />
* For Microsoft Windows, there are DirectShow filters supporting Opus, including DC-Bass Source Mod and the LAV Filters.<br />
* In GStreamer the integration of Opus support is complete.<br />
* FFmpeg supports decoding and encoding Opus via the external library libopus.<br />
<br />
=== Hardware support ===<br />
* Support in [[Rockbox]] is available in the developer version. This means hardware support for a series of portable media players (including some products from the iPod series by Apple and Sansa, iriver and Archos devices) and with "Rockbox as an Application" (RaaA) also on Android devices.<br />
<br />
=== Player software ===<br />
* VLC media player supports Opus since version 2.0.4<br />
* AIMP supports Opus natively as of version 3.20 build 1125 beta 1.<br />
* [[foobar2000]] supports the format natively as of v1.1.14 beta 1.<br />
* Mpxplay supports Opus (using a decoder DLL) as of v1.60 alpha 2<br />
* Android has a number of player apps supporting Opus, including PowerAmp and others.<br />
* [[Winamp]] supports for Opus via [http://forums.winamp.com/showthread.php?p=2925154#post2925154 3rd party plug-in].<br />
<br />
=== Other software ===<br />
* CDBurnerXP<br />
* MediaCoder<br />
* Report-IT<br />
* [[MP3tag|MP3tag]]<br />
<br />
== References & Notes ==<br />
<br />
*{{note|homepage|a}}[http://opus-codec.org/ opus-codec.org homepage]<br />
*{{note|FAQ|b}}[http://wiki.xiph.org/OpusFAQ Opus FAQ]<br />
*{{note|RFC|c}}[http://tools.ietf.org/html/rfc6716 IETF RFC 6716]<br />
<br />
[[Category:Codecs]]<br />
[[Category:Lossy]]<br />
[[Category:Encoder/Decoder]]</div>Gottkaiserhttps://wiki.hydrogenaud.io/index.php?title=OpusOpus2013-06-19T15:32:18Z<p>Gottkaiser: added software which supports Opus</p>
<hr />
<div>{{Software Infobox<br />
| name = Opus<br />
| logo = [[Image:opus-logo.png|250px|Official Opus logo]]<br />
| screenshot =<br />
| caption = Opus Interactive Audio Codec<br />
| maintainer = [http://xiph.org/ Xiph.Org Foundation]<br />
| stable_release = 1.0.2<br />
| preview_release = exp_analysis7<br />
| operating_system = Windows, Mac OS/X, Linux/BSD<br />
| use = Encoder/Decoder<br />
| license = 3-clause BSD license<br />
| website = [http://www.opus-codec.org/ opus-codec.org]<br />
}}<br />
<br />
'''Opus''' is a [[lossy]] audio compression format developed by the Internet Engineering Task Force (IETF) designed to be suitable for interactive real-time applications over the Internet,{{ref|homepage|a}} including music as well as speech, yet it is also very competitive for use as a storage and playback format, being a [http://people.xiph.org/~greg/opus/ha2011/ class leader at around 64 kbps]. As an open format standardised through [http://tools.ietf.org/html/rfc6716 Request for Comments (RFC) 6716],{{ref|RFC|c}} a high quality reference implementation is provided under the 3-clause BSD license{{ref|homepage|a}} which compiles and runs on the vast majority of general purpose and embedded (fixed point) processors. Many Software patents which cover Opus are licensed under royalty-free terms.{{ref|FAQ|b}} Opus is also a Mandatory To Implement (MTI) codec for the upcoming WebRTC (Web Real Time Communication) specification of the World Wide Web Consortium (W3C).<br />
<br />
Opus incorporates technology from two codecs, the speech-oriented SILK codec developed by Skype and the multi-purpose low-latency CELT codec developed by Xiph.org with significant changes to each to ensure they can work together.{{ref|RFC|c}} Opus can seamlessly transition among high and low bitrates, using a linear prediction codec (the SILK layer) at lower bitrates and a lapped transform codec (the CELT layer) at higher bitrates, as well as a hybrid of the two for a short overlap in which SILK encodes the 0-8kHz spectrum and the CELT layer encodes only the frequencies above 8kHz.{{ref|RFC|c}} Opus has very low algorithmic delay (typ 22.5 ms) compared to popular music formats such as [[MP3]], [[Vorbis |Ogg Vorbis]], [[AAC | LC-AAC and HE-AAC]] (all over 100 ms), yet performs very competitively with them in terms of quality per bitrate, making it comparably viable as a storage & playback format. Also unlike Vorbis, Opus does not require the definition of large codebooks for each individual file, making it also preferable for short clips of audio, such as those often used by game developers, a field where patent-free Vorbis is commonly used.{{ref|RFC|c}}<br />
<br />
Considerably more details of the history and potential applications for Opus are included in the ''Wikipedia'' page for '''[http://en.wikipedia.org/wiki/Opus_%28audio_format%29 Opus (audio format)]'''<br />
<br />
==Characteristics==<br />
Opus supports bitrates from 6kbps to 510kbps for typical stereo audio sources (and a maximum of around 255 kbps per channel for multichannel audio), with the 'sweet spot' for music and general audio around 30kbps (mono) and 40-100 kbps (stereo). It is intrinsically [[VBR | variable bitrate]], though constrained VBR and [[CBR | constant bitrate]] modes are possible where required. In the case of the reference release, libopus, the target bitrate is calibrated against the internal constant quality targets so that over a typical music collection, something very close to the target bitrate will be achieved. This bitrate-calibrated approach differs from most VBR encoders (e.g. LAME, helix mp3, qaac, Nero aacenc, Ogg Vorbis, Musepack) where a setting on some 'constant quality' scale (which differs between encoders) is used and the bitrate will fall where it may. Improved future versions can be expected to offer improved quality at the same setting. Independent implementations may adopt a different approach.<br />
<br />
Opus is able to seamlessly adapt its mode of operation without glitches or sound interruption (an illustrative demonstration of [http://opus-codec.org/examples/#gauge bitrate scalability] is on the Opus Examples page), which can be particularly useful for mixed-content audio or varying network conditions, making the unified Opus codec superior to a suite of different codecs that might otherwise cover the same range of bitrate and quality settings and would require out-of-band signalling to instigate codec switching. The switching includes the choice of mono, stereo and other channel mappings, the use of the speech-oriented SILK layer, the general-purpose CELT layer or the hybrid of both, and the use of different audio bandwidths (4kHz, 6kHz, 8kHz, 12kHz, 20kHz) as well as the quality adjustments within the same operating mode that are available in most VBR-capable codecs.<br />
<br />
Of importance mainly to interactive uses, but potentially useful in time-delayed audio streaming also, Opus includes packet loss concealment (PLC) in all modes and, in the speech-oriented modes where the SILK layer is active it also supports Forward Error Correction (FEC) where the expected rate of packet loss can be indicated to the encoder by the user or by application software and critical frames (e.g. consonant sounds) can be retransmitted at low bitrate to preserve intelligibility.<br />
<br />
For music and general audio, the CELT layer of Opus builds on knowledge gained during xiph.org's Vorbis development and ensures as a primary goal that the total energy in each spectral band is preserved while requiring only a modest bitrate overhead to achieve this, thereby eliminating a lot of bitrate-starvation artifacts such as 'birdies' that are common in low-bitrate MP3, especially during transients, applause and cymbal sounds. This technique likewise increases coding efficiency at bitrates targetting transparent music reproduction. Short blocks (2.5 ms) are also possible for efficient transient handling. Short blocks can also be used exclusively, if very low algorithmic delay (5.0ms) is required to enable very low-latency interative audio (e.g. live networked music performances such as remote jam sessions), though greater bitrate is then required to maintain the same quality (illustrated in [http://people.xiph.org/~xiphmont/demo/celt/demo.html#demo Monty's CELT demo page] under Constant PEAQ value, varying latency). CELT uses a number of additional techniques and provides additional advanced tools to enable encoder tuning.<br />
<br />
Opus natively supports [[gapless playback]] (though [[Gapless_playback#Poorly_designed_playback_systems | poor player design]] might itself induce interruptions during playback). Playback gain is also required, making some form of [[ReplayGain]] or [[ReplayGain_2.0_specification | similar]] volume control possible in any compliant player.<br />
<br />
==Bitrate performance==<br />
For mono speech, Opus ranges from intelligible narrowband speech reproduction starting at 6 kbps to medium-band, wideband and superwideband speech, reaching full-band speech by around 32 kbps. Above about 32 kbps, the SILK layer is no longer used at all, as CELT alone gives superior quality.<br />
<br />
For music, the SILK modes are quite tolerable and better than CELT at very low bitrates. The hybrid mode is adopted as bitrate increases, extending bandwidth first to 12kHz (comparable with compact cassette) then to the full 20kHz and CELT then takes over. Assuming the source is stereo, the transition from mono to stereo typically happens between the transition from 12kHz to 20kHz.<br />
<br />
==Indicative bitrate and quality==<br />
The table below gives illustrative, indicative quality guidance based on typical modes used internally by Opus and a range of listening tests.<br />
<br />
In the experimental libopus version 1.1-alpha, automatic detection of speech/music and bandwidth detection have been introduced to improve mode decisions, and VBR is less constrained, all with the aim of maximizing the quality/bitrate tradeoff. Thus changes are likely, and this table is likely to require small updates as the encoder is improved.<br />
<br />
===Speech encoding quality===<br />
This table assumes a '''monophonic''' source sampled at CD quality or above (typ 48 kHz sampling rate) but mentions stereo compatibility for 40kbps+. The default 20ms frame size (22.5ms latency) is assumed.<br />
<br />
{| class="wikitable" style="text-align:center"<br />
|-<br />
!Bitrate target<br />
!Bandwidth<br />
!typ SILK/CELT use<br />
!Speech quality notes<br />
!Use cases/notes/competitive codecs<br />
|-<br />
!1 to 5 kbps<br />
| -<br />
| -<br />
| <6kbps bitrate not supported<br />
| Try [http://codec2.org/ codec2] for 1.2-2.4 kbps speech<br />
|-<br />
!6 kbps<br />
|4 kHz<br />
|SILK<br />
|Fair, intelligible<br />
|AMR-NB may be a little better, but higher latency & proprietary, Speex also competitive<br />
|-<br />
!8 kbps<br />
|4 kHz narrowband<br />
|SILK<br />
|Close to telephone quality<br />
|AMR-NB & AMR-WB similar quality, but higher latency & proprietary. Speex competitive.<br />
|-<br />
!12 kbps<br />
|6 kHz medium-band<br />
|SILK<br />
|Medium bandwidth, better than telephone quality<br />
|Similar quality to AMR-WB<br />
|-<br />
!16 kbps<br />
|8 kHz wideband<br />
|SILK<br />
|Wideband speech quality<br />
|Similar to/better than AMR-WB<br />
|-<br />
!24 kbps<br />
|12 kHz super-wideband<br />
|hybrid<br />
|Near transparent speech<br />
|Better than AMR-WB. Podcasts/audiobooks/talk-radio.<br />
|-<br />
!32 kbps<br />
|20 kHz<br />
|hybrid / possibly CELT<br />
|Essentially transparent speech plus moderately good mono music<br />
|Much better than AMR-WB. Podcasts/audiobooks/talk-radio.<br />
|-<br />
!40 kbps<br />
|20 kHz<br />
|CELT<br />
|Essentially transparent mono or stereo speech, fairly good stereo music<br />
|Stereo podcasts/audiobooks/talk radio with some music<br />
|-<br />
!48 kbps+<br />
|20 kHz<br />
|CELT<br />
|Essentially transparent mono or stereo speech, reasonable music<br />
|Flexible general purpose modes to suit mixed music and speech<br />
|-<br />
|}<br />
<br />
===Music encoding quality===<br />
This table assumes a '''stereophonic''' source sampled at CD quality or above (typ 48 kHz sampling rate). Opus will automatically use mono at very low bitrates, though a certain amount of stereo encoding can still be used - content dependent even when mono is specified as the typical stereo mode in the table below.<br />
<br />
{| class="wikitable" style="text-align:center"<br />
|-<br />
!Bitrate target<br />
!Stereo mode<br />
!Bandwidth<br />
!typ SILK/CELT use<br />
!Music quality notes<br />
!Use cases/notes/competitive codecs<br />
|-<br />
!6 kbps<br />
|mono<br />
|4 kHz<br />
|SILK<br />
|Poor, muffled sound but intelligible lyrics.<br />
| -<br />
|-<br />
!8 kbps<br />
|mono<br />
|4 kHz<br />
|SILK<br />
|Poor, muffled but OK for bitrate<br />
| -<br />
|-<br />
!14 to 16 kbps<br />
|mono<br />
|6 kHz<br />
|SILK<br />
|Fairly Poor but OK for bitrate<br />
|Perhaps acceptable for incidental music<br />
|-<br />
!22 to 24 kbps<br />
|mono<br />
|8 kHz<br />
|SILK<br />
|Fair but OK for bitrate<br />
|OK for incidental music<br />
|-<br />
!32 kbps<br />
|mono<br />
|12 kHz<br />
|hybrid<br />
|Moderately good mono, reasonably bright treble (c.f. mono cassette)<br />
|Good for podcasts, audiobooks, CELT-only poss for music. Competitor HE-AAC@32kbps is stereo full-band but with annoying artifacts.<br />
|-<br />
!39 to 40 kbps<br />
|stereo<br />
|12 kHz<br />
|hybrid/CELT<br />
|Moderately good stereo, reasonably bright treble (c.f. stereo cassette)<br />
|Stereo podcasts, audiobooks, very low bitrate music<br />
|-<br />
!48 kbps<br />
|stereo<br />
|20 kHz<br />
|CELT<br />
|Full bandwidth stereo music, some artifacts, rarely nasty<br />
|Stereo podcasts, audiobooks, low bitrate music<br />
|-<br />
!64 kbps<br />
|stereo<br />
|20 kHz<br />
|CELT<br />
|Full bandwidth stereo music, nice sound, detectable differences to original (mostly 'not annoying')<br />
|Music storage & streaming. Beat HE-AAC, Vorbis, MP3 in [http://people.xiph.org/~greg/opus/ha2011/ listening test]<br />
|-<br />
!96 kbps<br />
|stereo<br />
|20 kHz<br />
|CELT<br />
|Full bandwidth stereo music, good quality approaching transparency<br />
|Music storage & high quality streaming.<br />
|-<br />
!112 kbps<br />
|stereo<br />
|20 kHz<br />
|CELT<br />
|Fairly close to transparency (needs more testing)<br />
|Music storage & high quality streaming. Very low-latency stereo networked music performance/jam sessions at OK quality (see below table)<br />
|-<br />
!128 kbps<br />
|stereo<br />
|20 kHz<br />
|CELT<br />
|Very close to transparency (needs more testing). Most modern codecs competitive (AAC-LC, Vorbis, MP3)<br />
|Music storage & streaming. Future download music sales.<br />
|-<br />
!256 kbps<br />
|stereo<br />
|20 kHz<br />
|CELT<br />
|Transparent with very low chance of artifacts (a few killer samples still detectable). Most old & new lossy codecs competitive.<br />
|Music storage & streaming, dedicated limited-bandwidth audio links (e.g. wireless, [http://en.wikipedia.org/wiki/Bluetooth_profile#Advanced_Audio_Distribution_Profile_.28A2DP.29 A2DP-bluetooth] type links). <br />
|-<br />
!510 kbps<br />
|stereo<br />
|20 kHz<br />
|CELT<br />
|Maximum possible stereo bitrate target (actual rate often less than 510 for default frame size). Most old and new lossy codecs competitive, plus near-lossless [[lossyWAV]] and [[WavPack | WavPack lossy]]<br />
|Music storage, dedicated limited-bitrate audio links (e.g. wireless, minimum latency high quality audio. LossyWAV and WavPack lossy are very competitive for storage, and WavPack lossy --blocksize=256 may be competitive with minimum latency mode also.<br />
|-<br />
!>510 kbps<br />
| -<br />
| -<br />
| -<br />
|Above Opus bitrate range allowed for stereo sources<br />
|Settle for 510kbps or use [[lossless]], [[lossyWAV]], [[WavPack | WavPack lossy]] or lossy transform/subband codecs like [[Vorbis]], [[Musepack]] at very high settings.<br />
|-<br />
|}<br />
<br />
===Lower latency versus quality/bitrate trade-off===<br />
====Packet overhead in interactive applications====<br />
For interactive use on the Internet or other packet-based networks, total bandwidth used will be subject to packet overhead. The more packet headers that are transmitted every second, the greater will be the overhead that is required. For this reason, Opus, while defaulting to 20.0ms frames, supports 60.0ms frames to reduce overhead when transporting low-bitrate SILK frames at the expense of greater latency, which may still be acceptable for speech, and also supports 10.0ms SILK frames to reduce latency somewhat at the expense of packet overhead.<br />
<br />
In the CELT layer, which tends to operate at higher bitrates than SILK, 20.0ms frames are the default, but frames of 10.0ms, 5.0ms and 2.5ms are also possible, which directly increases the frame overhead by transmitting more packets per second to achieve lower latency. In addition, as we'll see below it also reduces the quality/bitrate tradeoff of the CELT layer itself.<br />
<br />
None of the bitrates mentioned in this article account for the packet overhead.<br />
<br />
====CELT layer latency versus quality/bitrate trade-off====<br />
Unlike the SILK layer, which works on fixed 10.0ms blocks, 1, 2 or 6 of which can be combined into an Opus frame, the CELT layer is able to modify the encoding block lengths available to enable its use with shorter frames.<br />
<br />
When the CELT layer uses 10.0ms, 5.0ms and 2.5ms frames instead of the default 20.0ms, it must use smaller transform block sizes to achieve this, thereby reducing frequency resolution in the MDCT compared to the default transform window, thus reducing encoding efficiency for tonal signals. To obtain the same frequency precision for a sound divided into shorter transform windows, improved amplitude precision is necessary, resulting in increased bitrate to obtain the same perceptual quality (or conversely lower quality at the same bitrate).<br />
<br />
These reduced-latency modes remain efficient for transient signals, which use short blocks anyway.<br />
<br />
In all modes, the algorithmic delay consists of the frame size plus an additional 2.5ms delay. The CELT layer requires 2.5ms for MDCT window overlap.<br />
<br />
Xiph.org used matched PEAQ scores (approximate perceptual quality assessment made in software) for the CELT0.10 codec that was used as the basis of the CELT layer in the Opus reference release, which indicate the following [http://people.xiph.org/~xiphmont/demo/celt/demo.html#demo approximate equivalent settings] for stereo music.<br />
<br />
{| class="wikitable" style="text-align:center"<br />
|-<br />
!Frame size<br />
!Algorithmic delay<br />
!Bitrate to match 64kbps@22.5ms delay<br />
!fractional bitrate increase<br />
|-<br />
!20.0 ms<br />
|22.5 ms<br />
|64.0 kbps<br />
|0.0 %<br />
|-<br />
!10.0 ms<br />
|12.5 ms<br />
|70.4 kbps<br />
|10.0 %<br />
|-<br />
!5.0 ms<br />
|7.5 ms<br />
|84.8 kbps<br />
|32.5 %<br />
|-<br />
!2.5 ms<br />
|5.0 ms<br />
|112.0 kbps<br />
|75.0 %<br />
|-<br />
|}<br />
<br />
N.B. This table is useful for interactive streaming only. For music storage & delayed playback or non-interactive streaming, latency reduction is not important and the default 20.0ms frame size is preferable.<br />
<br />
== Hardware & Software Support ==<br />
<br />
Much of this section is based heavily on the Jan 12th 2013 version of the '''Support''' section of the [http://en.wikipedia.org/wiki/Opus_%28audio_format%29 Wikipedia article], which is more likely to be kept updated and to provide links to further information about the supporting platforms.<br />
<br />
The format and algorithms are openly documented and the reference implementation is published as free software. The reference implementation (Opus Audio Tools, opus-tools), consisting of separate encoders and decoders, is published under the terms of a BSD-like license. It is written in C programming language and can be compiled for hardware architectures with or without floating point unit. The accompanying diagnostic tool opusinfo reports detailed technical information about Opus files, including information on the standard compliance of the bitstream format. It is based on ogginfo from the vorbis-tools and therefore, unlike the encoder and decoder, available under the terms of version 2 of the GPL.<br />
<br />
=== Commandline binaries & libopus versions ===<br />
The commandline tools of the reference version are available pre-compiled for the most popular operating systems at [http://opus-codec.org/downloads opus-codec.org] and [https://ftp.mozilla.org/pub/mozilla.org/opus/ Mozilla's ftp server]. No other implementations of opus are currently known. The libopus commandline tools include encoder ''opusenc'', decoder ''opusdec'', and with a different license, the ''opusinfo'' opus stream & metadata analyzer.<br />
<br />
The '''latest stable release''' is recommended for general use and as of early 2013 is considered competitive with or superior to the best alternative speech or general music encoders at most supported bitrates.<br />
<br />
==== libopus v1.0 (recommended latest stable release) ====<br />
Released 11 Sep 2012 when RFC6716 was standardized but mostly fully developed by late 2011.<br />
<br />
'''Stable''', '''well-tuned''' ''opusenc'' reference encoder as included in RFC documentation.<br />
<br />
CELT layer closely related to CELT 0.10 implements Constrained VBR mode by default (bitrate boost used mainly for transients), plus true CBR.<br />
<br />
==== libopus v1.1-alpha ====<br />
Source code released 21 Dec 2012 for testing & user feedback ([https://ftp.mozilla.org/pub/mozilla.org/opus/win32/opus-tools-0.1.6-opus-1.1-alpha-win32.zip win32 binaries]), but not yet considered stable and well tested enough for general release.<br />
<br />
CELT layer [http://jmspeex.livejournal.com/11737.html quality improvements] introduced to provide '''unconstrained VBR''' include a rate boost not just for transients but now for highly tonal signals too and rate reduction when stereo image is narrow. There's also a rewrite of its '''transient detection''' code and '''time-frequency analysis''' code, and rewritten '''dynamic allocation''' code (HF/LF tilt and Band Boost) to allow more aggressive changes from the typical static allocation when warranted.<br />
<br />
There are many minor improvements to '''speech quality''' in both SILK and CELT layers.<br />
<br />
'''DC-rejection''' below 3 Hz also aids quality if inaudible DC offset is present with no effect on deep bass notes.<br />
<br />
'''Automatic speech/music detection''' is introduced to optimize encoding mode choices, especially near the bitrate target range (presumably around 24~40kbps) where the encoder may perform best with SILK, hybrid or CELT depending on content type. Below that range SILK performs best for both music & speech, and above it CELT performs best for speech & music. The detection, without look-ahead, takes a second or two typically and will sometimes make incorrect decisions. The developers would be keen to know of examples of its failure.<br />
<br />
'''Automatic bandwidth detection''' is also introduced to save wasted bits allocated to absent frequencies, and while easier to implement, developers would also been keen to know of any failure of this feature (potentially caused by aliasing, quantization and dithering/noise-shaping in source material).<br />
<br />
=== VoIP software ===<br />
* The voice-chat software Mumble supports Opus as its main codec.<br />
* SIP softphones Phoner and PhonerLite support Opus<br />
* The SIP and IAX2 client SFLphone is being fitted with Opus support.<br />
* Integration of Opus into the Skype client is finished, although no version with Opus support has yet been published.<br />
* TrueConf video conferencing solutions support Opus.<br />
* Opus support is planned for Jitsi 2.0, together with VP8 video<br />
* Empathy may use any format supported in GStreamer, including Opus.<br />
* Line2 has replaced their current codec with Opus. Their iOS app will be the first to be released with the Opus. The Android app will follow later.<br />
* CSipSimple supports Opus, Codec2, G.726 and G.722.1 with an additional plug-in.<br />
* The voice-chat software TeamSpeak 3 supports Opus for voice and music in pre-release server 3.0.7-pre2 and beta client version 3.0.10<br />
<br />
=== Web frameworks and browsers ===<br />
* Opus support is mandatory for WebRTC implementations.<br />
* Mozilla supports Opus beginning with version 15 of Firefox and Thunderbird, plus Seamonkey, which is uses shared codebase.<br />
* Depending on the backend in use, Opera supports inline playback of embedded Opus files. Official support for Opus and WebRTC are on the development roadmap.<br />
* Chromium and Google Chrome will have audio support as of version 25.<br />
* Maxthon Cloud Browser<br />
<br />
=== Streaming audio ===<br />
* Icecast. (examples: [http://dir.xiph.org/ Stream directory], [http://smj.delfa.net/opus_64.m3u 64k]/[http://smj.delfa.net/opus_256.m3u 256k] [http://smj.delfa.net/ Smooth Jazz Opus Stream], [http://www.absoluteradio.co.uk/listen/labs.html Absolute Radio Opus Trial] 7 stations at 24,64,96 kbps, [http://icecast.ofdoom.com:8000/burst-opus.ogg Icecast Of Doom 96k]<br />
* Krad Radio<br />
* Liquidsoap<br />
<br />
=== Operating systems and desktop multimedia frameworks ===<br />
* In Debian GNU/Linux the Opus development tools and supporting libraries can be installed from the preconfigured repositories in the next stable version ("wheezy") that is expected to be released in early 2013.<br />
* For Microsoft Windows, there are DirectShow filters supporting Opus, including DC-Bass Source Mod and the LAV Filters.<br />
* In GStreamer the integration of Opus support is complete.<br />
* FFmpeg supports decoding and encoding Opus via the external library libopus.<br />
<br />
=== Hardware support ===<br />
* Support in [[Rockbox]] is available in the developer version. This means hardware support for a series of portable media players (including some products from the iPod series by Apple and Sansa, iriver and Archos devices) and with "Rockbox as an Application" (RaaA) also on Android devices.<br />
<br />
=== Player software ===<br />
* VLC media player supports Opus since version 2.0.4<br />
* AIMP supports Opus natively as of version 3.20 build 1125 beta 1.<br />
* [[foobar2000]] supports the format natively as of v1.1.14 beta 1.<br />
* Mpxplay supports Opus (using a decoder DLL) as of v1.60 alpha 2<br />
* Android has a number of player apps supporting Opus, including PowerAmp and others.<br />
* [[Winamp]] supports for Opus via a [http://forums.winamp.com/showthread.php?p=2925154#post2925154 3rd party plug-in].<br />
<br />
=== Other software ===<br />
* CDBurnerXP<br />
* MediaCoder<br />
* Report-IT<br />
* [[MP3tag|MP3tag]]<br />
<br />
== References & Notes ==<br />
<br />
*{{note|homepage|a}}[http://opus-codec.org/ opus-codec.org homepage]<br />
*{{note|FAQ|b}}[http://wiki.xiph.org/OpusFAQ Opus FAQ]<br />
*{{note|RFC|c}}[http://tools.ietf.org/html/rfc6716 IETF RFC 6716]<br />
<br />
[[Category:Codecs]]<br />
[[Category:Lossy]]<br />
[[Category:Encoder/Decoder]]</div>Gottkaiserhttps://wiki.hydrogenaud.io/index.php?title=Mp3tagMp3tag2013-04-01T20:13:48Z<p>Gottkaiser: </p>
<hr />
<div>{{Software Infobox<br />
| name = MP3tag<br />
| logo =<br />
| screenshot = [[Image:MP3tag-screenshot.png|250px|MP3tag screenshot]]<br />
| caption = powerful and easy-to-use tool to edit metadata <br />
| maintainer = Florian Heidenreich<br />
| stable_release = [http://www.mp3tag.de/en/download.html v2.54] (2012-12-22)<br />
| preview_release = [http://developer.mp3tag.de/ v2.54b] (2013-04-01)<br />
| operating_system = Windows<br />
| use = Metadata<br />
| license = Freeware <br />
| website = [http://www.mp3tag.de/en/ http://www.mp3tag.de]<br />
}}<br />
<br />
=Introduction=<br />
'''MP3tag''' is an powerful [[Metadata|metadata]] editor for lots of common audio formats. You can rename files based on the [[Tagging|tag]] information, replace characters or words from tags and filenames, import/export tag information, create playlists.<br />
<br />
The program supports online freedb database lookups for selected files, allowing you to automatically gather proper tag information for select files or CDs.<br />
<br />
==Features==<br />
<br />
* Write [[ID3v1.1]]-, [[ID3v2]]-, [[APEv2]]-Tags and [[Vorbis_Comment|Vorbis Comments]] to multiple files at once<br />
* Full Unicode support<br />
* Support for embedded cover art<br />
* Automatically create playlists<br />
* Recursive subfolders support<br />
* Remove parts or the entire tag of multiple files<br />
* Rename files based on the tag information<br />
* Import tags from filenames<br />
* Format tags and filenames<br />
* Replace characters or words from tags and filenames<br />
* Regular Expressions<br />
* Export tag information to user-defined formats (like html, rtf, csv, xml)<br />
* Import tag information from online databases like freedb or Amazon (also by text-search)<br />
* Import tag information from local freedb databases<br />
* Support for [[ID3v2|ID3v2.3]] (ISO-8859-1 and UTF-16) and [[ID3v2|ID3v2.4]] with [[UTF-8]]<br />
<br />
==Supported formats==<br />
<br />
* Advanced Audio Coding ([[AAC|aac]])<br />
* Free Lossless Audio Codec ([[FLAC|flac]])<br />
* Monkey's Audio ([[APE|ape]])<br />
* Mpeg Layer 3 ([[MP3|mp3]])<br />
* MPEG-4 ([[MP4|mp4]] / [[M4A|m4a]] / m4b / [[iTunes]] compatible)<br />
* Musepack ([[MPC|mpc]])<br />
* Ogg Vorbis ([[Ogg_Vorbis|ogg]])<br />
* OptimFROG ([[OptimFROG|ofr]])<br />
* OptimFROG DualStream (ofs)<br />
* Opus ([[Opus|opus]])<br />
* Speex ([[Speex|spx]])<br />
* TAK ([[TAK]])<br />
* True Audio (tta)<br />
* Windows Media Audio ([[WMA|wma]])<br />
* WavPack ([[WavPack|wv]])<br />
<br />
==Operating Systems==<br />
* Win XP<br />
* Win 2003<br />
* Win Vista<br />
* Win 2008<br />
* Win 7<br />
* Win 8<br />
<br />
Windows 2000 is no longer supported as of version 2.40. Version 2.39 is still available on the download page at the MP3tag website.<br />
<br />
==External links==<br />
* [http://www.mp3tag.de/en/ Mp3tag: Homepage]<br />
* [http://www.mp3tag.de/en/download.html Mp3tag: Download]<br />
* [http://www.anytag.de/forums/ Mp3tag: official forum]<br />
* [http://www.mp3tag.de/en/changelog.html Mp3tag: changelog]<br />
* [http://wiki.slimdevices.com/index.php/Mp3tagGuide Mp3tagGuide - Slim Devices]<br />
* [http://www.anytag.de/forums/index.php?showtopic=1794 Mp3tag: additional Web Sources]<br />
<br />
[[Category:Software]]<br />
[[Category:Tag editors]]</div>Gottkaiserhttps://wiki.hydrogenaud.io/index.php?title=Mp3tagMp3tag2013-04-01T20:13:29Z<p>Gottkaiser: </p>
<hr />
<div>{{Software Infobox<br />
| name = MP3tag<br />
| logo =<br />
| screenshot = [[Image:MP3tag-screenshot.png|250px|MP3tag screenshot]]<br />
| caption = powerful and easy-to-use tool to edit metadata <br />
| maintainer = Florian Heidenreich<br />
| stable_release = [http://www.mp3tag.de/en/download.html v2.54] (2012-12-22)<br />
| preview_release = [http://developer.mp3tag.de/ 2.54b] (2013-04-01)<br />
| operating_system = Windows<br />
| use = Metadata<br />
| license = Freeware <br />
| website = [http://www.mp3tag.de/en/ http://www.mp3tag.de]<br />
}}<br />
<br />
=Introduction=<br />
'''MP3tag''' is an powerful [[Metadata|metadata]] editor for lots of common audio formats. You can rename files based on the [[Tagging|tag]] information, replace characters or words from tags and filenames, import/export tag information, create playlists.<br />
<br />
The program supports online freedb database lookups for selected files, allowing you to automatically gather proper tag information for select files or CDs.<br />
<br />
==Features==<br />
<br />
* Write [[ID3v1.1]]-, [[ID3v2]]-, [[APEv2]]-Tags and [[Vorbis_Comment|Vorbis Comments]] to multiple files at once<br />
* Full Unicode support<br />
* Support for embedded cover art<br />
* Automatically create playlists<br />
* Recursive subfolders support<br />
* Remove parts or the entire tag of multiple files<br />
* Rename files based on the tag information<br />
* Import tags from filenames<br />
* Format tags and filenames<br />
* Replace characters or words from tags and filenames<br />
* Regular Expressions<br />
* Export tag information to user-defined formats (like html, rtf, csv, xml)<br />
* Import tag information from online databases like freedb or Amazon (also by text-search)<br />
* Import tag information from local freedb databases<br />
* Support for [[ID3v2|ID3v2.3]] (ISO-8859-1 and UTF-16) and [[ID3v2|ID3v2.4]] with [[UTF-8]]<br />
<br />
==Supported formats==<br />
<br />
* Advanced Audio Coding ([[AAC|aac]])<br />
* Free Lossless Audio Codec ([[FLAC|flac]])<br />
* Monkey's Audio ([[APE|ape]])<br />
* Mpeg Layer 3 ([[MP3|mp3]])<br />
* MPEG-4 ([[MP4|mp4]] / [[M4A|m4a]] / m4b / [[iTunes]] compatible)<br />
* Musepack ([[MPC|mpc]])<br />
* Ogg Vorbis ([[Ogg_Vorbis|ogg]])<br />
* OptimFROG ([[OptimFROG|ofr]])<br />
* OptimFROG DualStream (ofs)<br />
* Opus ([[Opus|opus]])<br />
* Speex ([[Speex|spx]])<br />
* TAK ([[TAK]])<br />
* True Audio (tta)<br />
* Windows Media Audio ([[WMA|wma]])<br />
* WavPack ([[WavPack|wv]])<br />
<br />
==Operating Systems==<br />
* Win XP<br />
* Win 2003<br />
* Win Vista<br />
* Win 2008<br />
* Win 7<br />
* Win 8<br />
<br />
Windows 2000 is no longer supported as of version 2.40. Version 2.39 is still available on the download page at the MP3tag website.<br />
<br />
==External links==<br />
* [http://www.mp3tag.de/en/ Mp3tag: Homepage]<br />
* [http://www.mp3tag.de/en/download.html Mp3tag: Download]<br />
* [http://www.anytag.de/forums/ Mp3tag: official forum]<br />
* [http://www.mp3tag.de/en/changelog.html Mp3tag: changelog]<br />
* [http://wiki.slimdevices.com/index.php/Mp3tagGuide Mp3tagGuide - Slim Devices]<br />
* [http://www.anytag.de/forums/index.php?showtopic=1794 Mp3tag: additional Web Sources]<br />
<br />
[[Category:Software]]<br />
[[Category:Tag editors]]</div>Gottkaiserhttps://wiki.hydrogenaud.io/index.php?title=Mp3tagMp3tag2013-04-01T16:23:13Z<p>Gottkaiser: </p>
<hr />
<div>{{Software Infobox<br />
| name = MP3tag<br />
| logo =<br />
| screenshot = [[Image:MP3tag-screenshot.png|250px|MP3tag screenshot]]<br />
| caption = powerful and easy-to-use tool to edit metadata <br />
| maintainer = Florian Heidenreich<br />
| stable_release = [http://www.mp3tag.de/en/download.html v2.54] (2012-12-22)<br />
| preview_release = [http://developer.mp3tag.de/ 2.54a] (2013-04-01)<br />
| operating_system = Windows<br />
| use = Metadata<br />
| license = Freeware <br />
| website = [http://www.mp3tag.de/en/ http://www.mp3tag.de]<br />
}}<br />
<br />
=Introduction=<br />
'''MP3tag''' is an powerful [[Metadata|metadata]] editor for lots of common audio formats. You can rename files based on the [[Tagging|tag]] information, replace characters or words from tags and filenames, import/export tag information, create playlists.<br />
<br />
The program supports online freedb database lookups for selected files, allowing you to automatically gather proper tag information for select files or CDs.<br />
<br />
==Features==<br />
<br />
* Write [[ID3v1.1]]-, [[ID3v2]]-, [[APEv2]]-Tags and [[Vorbis_Comment|Vorbis Comments]] to multiple files at once<br />
* Full Unicode support<br />
* Support for embedded cover art<br />
* Automatically create playlists<br />
* Recursive subfolders support<br />
* Remove parts or the entire tag of multiple files<br />
* Rename files based on the tag information<br />
* Import tags from filenames<br />
* Format tags and filenames<br />
* Replace characters or words from tags and filenames<br />
* Regular Expressions<br />
* Export tag information to user-defined formats (like html, rtf, csv, xml)<br />
* Import tag information from online databases like freedb or Amazon (also by text-search)<br />
* Import tag information from local freedb databases<br />
* Support for [[ID3v2|ID3v2.3]] (ISO-8859-1 and UTF-16) and [[ID3v2|ID3v2.4]] with [[UTF-8]]<br />
<br />
==Supported formats==<br />
<br />
* Advanced Audio Coding ([[AAC|aac]])<br />
* Free Lossless Audio Codec ([[FLAC|flac]])<br />
* Monkey's Audio ([[APE|ape]])<br />
* Mpeg Layer 3 ([[MP3|mp3]])<br />
* MPEG-4 ([[MP4|mp4]] / [[M4A|m4a]] / m4b / [[iTunes]] compatible)<br />
* Musepack ([[MPC|mpc]])<br />
* Ogg Vorbis ([[Ogg_Vorbis|ogg]])<br />
* OptimFROG ([[OptimFROG|ofr]])<br />
* OptimFROG DualStream (ofs)<br />
* Opus ([[Opus|opus]])<br />
* Speex ([[Speex|spx]])<br />
* TAK ([[TAK]])<br />
* True Audio (tta)<br />
* Windows Media Audio ([[WMA|wma]])<br />
* WavPack ([[WavPack|wv]])<br />
<br />
==Operating Systems==<br />
* Win XP<br />
* Win 2003<br />
* Win Vista<br />
* Win 2008<br />
* Win 7<br />
* Win 8<br />
<br />
Windows 2000 is no longer supported as of version 2.40. Version 2.39 is still available on the download page at the MP3tag website.<br />
<br />
==External links==<br />
* [http://www.mp3tag.de/en/ Mp3tag: Homepage]<br />
* [http://www.mp3tag.de/en/download.html Mp3tag: Download]<br />
* [http://www.anytag.de/forums/ Mp3tag: official forum]<br />
* [http://www.mp3tag.de/en/changelog.html Mp3tag: changelog]<br />
* [http://wiki.slimdevices.com/index.php/Mp3tagGuide Mp3tagGuide - Slim Devices]<br />
* [http://www.anytag.de/forums/index.php?showtopic=1794 Mp3tag: additional Web Sources]<br />
<br />
[[Category:Software]]<br />
[[Category:Tag editors]]</div>Gottkaiserhttps://wiki.hydrogenaud.io/index.php?title=BonkencBonkenc2010-11-16T18:09:25Z<p>Gottkaiser: redirection to fre:ac</p>
<hr />
<div>#REDIRECT [[Fre:ac]]</div>Gottkaiserhttps://wiki.hydrogenaud.io/index.php?title=BonkEncBonkEnc2010-11-16T18:08:20Z<p>Gottkaiser: redirect to fre:ac</p>
<hr />
<div>#REDIRECT [[Fre:ac]]</div>Gottkaiserhttps://wiki.hydrogenaud.io/index.php?title=FreacFreac2010-11-16T18:07:25Z<p>Gottkaiser: redirection to fre:ac</p>
<hr />
<div>#REDIRECT [[Fre:ac]]</div>Gottkaiserhttps://wiki.hydrogenaud.io/index.php?title=Fre:acFre:ac2010-11-16T18:07:15Z<p>Gottkaiser: creation of a new site for fre:ac (former BonkEnc)</p>
<hr />
<div>{{title|fre:ac}}<br />
<br />
{{Software Infobox<br />
| name = fre:ac<br />
| logo =<br />
| screenshot = [[Image:BonkEnc-screenshot.png|250px|fre:ac screenshot]]<br />
| caption = Slim CD ripper, audio encoder and converter <br />
| maintainer = Robert Kausch<br />
| stable_release = 1.0.17 (14 November 2010)<br />
| preview_release = <br />
| operating_system = Windows<br />
| use = Digital Audio Extraction, transcoding<br />
| license = GPL <br />
| website = [http://www.freac.org http://www.freac.org]<br />
}}<br />
<br />
=Introduction=<br />
'''fre:ac''' ('''former''' BonkEnc ) is a CD ripper, audio encoder and converter for various formats. It's a very slim program and can be run from a USB stick. BonkEnc is available under GPL liscense and has native support for numerous languages. <br />
<br />
== Features ==<br />
* CD ripping<br />
** [[Cdparanoia]] mode<br />
** jitter correction<br />
** CD Text support<br />
* [[Transcoding]] from on to another format<br />
* using Compact Disc Database (CDDB)<br />
* support [[ID3v1]], [[ID3v2]], MP4-Metadata and [[Vorbis_comment|Vorbis comment]] [[Tags]]<br />
* keeps image tags when converting from FLAC to MP3<br />
* creating cue sheets and playlists<br />
* full UTF-8 Unicode support<br />
* additional command line interface (CD Ripping/Encoding)<br />
<br />
== Supported Formats ==<br />
* [[MP3]] ([[LAME]])<br />
* [[AAC]] ([[FAAC]])<br />
* [[MP4]]/[[M4A]]<br />
* [[Ogg_Vorbis|Ogg Vorbis]]<br />
* [[FLAC]]<br />
* [http://www.logarithmic.net/pfh/bonk Bonk] v0.11<br />
<br />
== Recommended Settings ==<br />
{| border="0" valign="top"<br />
|<br />
* [[LAME]] MP3 Encoder v3.97:<br />
* [[FAAC]] MP4/AAC Encoder v1.26:<br />
* Ogg Vorbis Encoder ([[aoTuV|aoTuV beta 5]]):<br />
* FLAC Audio Encoder v1.2.1:<br />
||<br />
[[LAME#Recommended_encoder_settings|here]]<br/><br />
to add<br/><br />
[[Recommended_Ogg_Vorbis#Recommended_Encoder_Settings|here]]<br/><br />
Use preset 5<br />
|}<br />
<br />
== Supported languages ==<br />
{| border="0" -valign="top"<br />
|-valign="top"<br />
||<br />
* Catalan<br />
* Chinese<br />
* Czech<br />
* Danish<br />
* Dutch<br />
* English<br />
* Esperanto<br />
* Finnish<br />
||<br />
* French<br />
* German<br />
* Greek<br />
* Hungarian<br />
* Italian<br />
* Japanese<br />
* Korean<br />
* Lithuanian<br />
||<br />
* Polish<br />
* Portuguese<br />
* Russian<br />
* Serbian<br />
* Slovak<br />
* Spanish<br />
* Swedish<br />
* Turkish<br />
||<br />
* Ukrainian<br />
* Romanian<br />
|}<br />
<br />
[[Category:CD Rippers]]<br />
[[Category:Software]]</div>Gottkaiserhttps://wiki.hydrogenaud.io/index.php?title=File:BonkEnc-screenshot.pngFile:BonkEnc-screenshot.png2010-11-16T17:45:58Z<p>Gottkaiser: uploaded a new version of "File:BonkEnc-screenshot.png":&#32;Screenshot of BonkEnc v1.0.17</p>
<hr />
<div>Screenshot of BonkEnc v1.0</div>Gottkaiserhttps://wiki.hydrogenaud.io/index.php?title=BonkEncBonkEnc2010-02-02T10:28:17Z<p>Gottkaiser: version update</p>
<hr />
<div>{{Software Infobox<br />
| name = BonkEnc<br />
| screenshot = [[Image:BonkEnc-screenshot.png|250px|BonkEnc screenshot]]<br />
| caption = Slim CD ripper, audio encoder and converter <br />
| maintainer = Robert Kausch<br />
| stable_release = 1.0.14 (21 October 2009)<br />
| preview_release = <br />
| operating_system = Windows<br />
| use = Digital Audio Extraction<br />
| license = GPL <br />
| website = [http://www.bonkenc.org http://www.bonkenc.org]<br />
}}<br />
<br />
=Introduction=<br />
'''BonkEnc''' is a CD ripper, audio encoder and converter for various formats. It's a very slim program and can be run from a USB stick. BonkEnc is available under GPL liscense and has native support for numerous languages. <br />
<br />
== Features ==<br />
* CD ripping<br />
** [[Cdparanoia]] mode<br />
** jitter correction<br />
** CD Text support<br />
* [[Transcoding]] from on to another format<br />
* using Compact Disc Database (CDDB)<br />
* support [[ID3v1]], [[ID3v2]], MP4-Metadata and [[Vorbis_comment|Vorbis comment]] [[Tags]]<br />
* keeps image tags when converting from FLAC to MP3<br />
* creating cue sheets and playlists<br />
* full UTF-8 Unicode support<br />
* additional command line interface (CD Ripping/Encoding)<br />
<br />
== Supported Formats ==<br />
* [[MP3]] ([[LAME]])<br />
* [[AAC]] ([[FAAC]])<br />
* [[MP4]]/[[M4A]]<br />
* [[Ogg_Vorbis|Ogg Vorbis]]<br />
* [[FLAC]]<br />
* [http://www.logarithmic.net/pfh/bonk Bonk] v0.11<br />
<br />
== Recommended Settings ==<br />
{| border="0" valign="top"<br />
|<br />
* [[LAME]] MP3 Encoder v3.97:<br />
* [[FAAC]] MP4/AAC Encoder v1.26:<br />
* Ogg Vorbis Encoder ([[aoTuV|aoTuV beta 5]]):<br />
* FLAC Audio Encoder v1.2.1:<br />
||<br />
[[LAME#Recommended_encoder_settings|here]]<br/><br />
to add<br/><br />
[[Recommended_Ogg_Vorbis#Recommended_Encoder_Settings|here]]<br/><br />
Use preset 5<br />
|}<br />
<br />
== Supported languages ==<br />
{| border="0" -valign="top"<br />
|-valign="top"<br />
||<br />
* Catalan<br />
* Chinese<br />
* Czech<br />
* Danish<br />
* Dutch<br />
* English<br />
* Esperanto<br />
* Finnish<br />
||<br />
* French<br />
* German<br />
* Greek<br />
* Hungarian<br />
* Italian<br />
* Japanese<br />
* Korean<br />
* Lithuanian<br />
||<br />
* Polish<br />
* Portuguese<br />
* Russian<br />
* Serbian<br />
* Slovak<br />
* Spanish<br />
* Swedish<br />
* Turkish<br />
||<br />
* Ukrainian<br />
* Romanian<br />
|}<br />
<br />
<br />
<br />
== External links==<br />
* [http://www.bonkenc.org/ BonkEnc: Homepage]<br />
* [http://www.bonkenc.org/index.php?option=com_content&task=blogcategory&id=3&Itemid=33 BonkEnc: Download]<br />
* [http://sourceforge.net/forum/?group_id=27149 BonkEnc: official forum] official support forum <br />
<br />
[[Category:Software]]</div>Gottkaiserhttps://wiki.hydrogenaud.io/index.php?title=Download_pageDownload page2010-02-02T10:22:23Z<p>Gottkaiser: /* Transcoders */ adding links</p>
<hr />
<div>All programs mentioned anywhere in the wiki can be downloaded here.<br />
See also the [[:Category:Software|Software Category]] article for more software not listed here. <br />
<br />
==CD Rippers==<br />
===Windows===<br />
{| border="0" cellpadding="0" cellspacing="1" style="text-align:center; border:2px solid #ccccff; margin-bottom: 20px;"<br />
|- style="background:#ccccff"<br />
! style="width:150px;" | Name<br />
! style="width:90px;" | License<br />
! style="width:100px;" | Website<br />
! style="width:300px;" | Description<br />
|-<br />
! align="left" | [[CDex]]<br />
| GPL<br />
| [http://cdexos.sourceforge.net/ here]<br />
| align="left" | An open-source ripper for Windows that uses cdparanoia functionality<br />
|<br />
|- style="background-color: #eeeeee;"<br />
! align="left" | [[DBpowerAMP with AccurateRip|DBpowerAMP]]<br />
| Free, Shareware <br />
| [http://www.dbpoweramp.com/ here]<br />
| align="left" | A secure ripper for Windows that includes Accurate Stream functionality<br />
|-<br />
! align="left" | Deep Ripper<br />
| GPL<br />
| [http://www.deepburner.com/ here]<br />
|<br />
|- style="background-color: #eeeeee;"<br />
! align="left" | [[EAC]]<br />
| Free<br />
| [http://www.exactaudiocopy.de/ here]<br />
| align="left" | A secure ripper for Windows, C2 error pointers, Accurate Stream, etc.<br />
|-<br />
! align="left" | [[BonkEnc]]<br />
| GPL<br />
| [http://www.bonkenc.org/ here]<br />
| align="left" | Ripper with [[Cdparanoia]] support. It's an open-source project.<br />
|}<br />
<br />
===Mac OS X===<br />
{| border="0" cellpadding="0" cellspacing="1" style="text-align:center; border:2px solid #bbffbb; margin-bottom: 20px;"<br />
|- style="background:#bbffbb;"<br />
! style="width:150px;" | Name<br />
! style="width:90px;" | License<br />
! style="width:100px;" | Website<br />
! style="width:350px;" | Description<br />
|-<br />
! align="left" | [[Max]]<br />
| GPL<br />
| [http://sbooth.org/Max/ here]<br />
| align="left" | A secure ripper for OS X that uses additional cdparanoia functionality<br />
|-<br />
|- style="background-color:#eeeeee;"<br />
! align="left" | [[XLD]]<br />
| GPL<br />
| [http://tmkk.hp.infoseek.co.jp/xld/index_e.html here]<br />
| align="left" | X Lossless Decoder(XLD) is a tool for Mac OS X that is able to decode/convert/play various 'lossless' audio files. The supported audio files can be split into some tracks with cue sheet when decoding. Can convert between many lossless and lossy formats. Plugin oriented design, for easy exchange for new encoders.<br />
|}<br />
===Linux===<br />
{| border="0" cellpadding="0" cellspacing="1" style="text-align:center; border:2px solid #ffcccc; margin-bottom: 20px;"<br />
|- style="background:#ffcccc;"<br />
! style="width:150px;" | Name<br />
! style="width:90px;" | License<br />
! style="width:100px;" | Website<br />
! style="width:320px;" | Description<br />
|- style="background-color:#eeeeee;"<br />
! align="left" | abcde <br />
| GPL <br />
|[http://www.hispalinux.es/~data/abcde.php here] <br />
| align="left" | A command-line based ripper with cdparanoia functionality<br />
|- <br />
! align="left" | [[cdparanoia]]<br />
| BSD, GPL<br />
| [http://www.xiph.org/paranoia/ here]<br />
| align="left" | One of the first secure standalone rippers for the Linux platform<br />
|- style="background-color:#eeeeee;"<br />
! align="left" | [[Grip]] <br />
| GPL <br />
| [http://www.nostatic.org/grip here] <br />
| align="left" | An open-source Gnome interface ripper that uses cdparanoia functionality <br />
|- <br />
! align="left" | [[Rubyripper]] <br />
| GPL <br />
| [http://www.rubyforge.org/ here] <br />
| align="left" | A secure ripper for the Linux that uses additional cdparanoia functionality<br />
|}<br />
<br />
==CD/DVD Writers==<br />
===Windows===<br />
{| border="0" cellpadding="0" cellspacing="1" style="text-align:center; border:2px solid #ccccff; margin-bottom: 20px;"<br />
|- style="background:#ccccff"<br />
! style="width:185px;" | Name<br />
! style="width:80px;" | Unicode<br />
! style="width:90px;" | License<br />
! style="width:100px;" | Website<br />
! style="width:270px;" | Description<br />
|-<br />
! align="left" | BurnAtOnce<br />
| N<br />
| Free<br />
| [http://www.burnatonce.com/ here]<br />
| align="left" | CD writing application based upon CDRDAO <br />
|<br />
|- style="background-color: #eeeeee;"<br />
! align="left" | [[Burrrn]] (CDA only)<br />
| N<br />
| Free<br />
| [http://www.burrrn.net/ here]<br />
|<br />
|-<br />
! align="left" | CDBurnerXP<br />
| <br />
| Free<br />
| [http://www.cdburnerxp.se/ here]<br />
|<br />
|- style="background-color: #eeeeee;"<br />
! align="left" | DeepBurner Free<br />
| N<br />
| GPL<br />
| [http://www.deepburner.com/ here]<br />
|<br />
|-<br />
! align="left" | DeepBurner Pro<br />
| <br />
| Shareware<br />
| [http://www.deepburner.com/ here]<br />
|<br />
|- style="background-color: #eeeeee;"<br />
! align="left" | Express Burn<br />
| N<br />
| Free<br />
| [http://nch.com.au/burn/index.html here]<br />
|<br />
|-<br />
! align="left" | Express Burn Plus<br />
| <br />
| Shareware<br />
| [http://nch.com.au/burn/index.html here]<br />
|<br />
|-style="background-color: #eeeeee;"<br />
! align="left" | Infra Recorder<br />
| N<br />
| GPL <br />
| [http://infrarecorder.sourceforge.net/ here]<br />
| <br />
|-<br />
! align="left" | [[Nero]]<br />
| N<br />
| Shareware<br />
| [http://www.nero.com/ here]<br />
| align="left" |<br />
|-style="background-color: #eeeeee;"<br />
! align="left" | SilentNight Micro-CD Burner<br />
| N<br />
| Free<br />
| [http://www.silentnight2004.com/Download.html here]<br />
|<br />
|}<br />
<br />
===Mac OS X===<br />
{| border="0" cellpadding="0" cellspacing="1" style="text-align:center; border:2px solid #bbffbb; margin-bottom: 20px;"<br />
|- style="background:#bbffbb;"<br />
! style="width:130px;" | Name<br />
! style="width:80px;" | Unicode<br />
! style="width:90px;" | License<br />
! style="width:100px;" | Website<br />
! style="width:310px;" | Description<br />
|- <br />
! align="left" | [[DVD-Audio Tools]]<br />
| Y <br />
| GPL <br />
| [http://dvd-audio.sourceforge.net/ here] <br />
| align="left" | Open-source DVD-Audio authoring application <br />
|- style="background-color: #eeeeee;"<br />
! align="left" | [[FireStarter FX]] <br />
| N <br />
| Free<br />
| [http://www.projectomega.org/subcat.php?lg=en&php=products_firestarter here] <br />
| align="left" | Free OS X Cocoa CD writing application<br />
|- <br />
! align="left" | [[X-CD-Roast]] <br />
| N <br />
| Free <br />
| [http://www.xcdroast.org/xcdr098/xcdrosX.html here] <br />
| align="left" | New OS X port of this Linux CD writing application<br />
|- style="background-color: #eeeeee;"<br />
! align="left" | Burn<br />
| N <br />
| Free<br />
| [http://burn-osx.sourceforge.net/Pages/English/home.html/ here] <br />
| align="left" | Versatile CD/DVD authoring application<br />
|}<br />
<br />
===Linux===<br />
{| border="0" cellpadding="0" cellspacing="1" style="text-align:center; border:2px solid #ffcccc; margin-bottom: 20px;"<br />
|- style="background:#ffcccc;"<br />
! style="width:130px;" | Name<br />
! style="width:80px;" | Unicode<br />
! style="width:90px;" | License<br />
! style="width:100px;" | Website<br />
! align="center" style="width:260px;" | Description<br />
|-<br />
! align="left" | CDRDAO <br />
| N <br />
| GPL <br />
| [http://www.cdrdao.org/ here] <br />
| align="left" | Cdrdao records audio or data CD-Rs in disk-at-once (DAO) mode<br />
|- style="background-color:#eeeeee;"<br />
! align="left" | DVD-Audio Tools <br />
| Y <br />
| GPL <br />
| [http://dvd-audio.sourceforge.net/ here] <br />
| align="left" | Open-source DVD-Audio authoring application<br />
|-<br />
! align="left" | [[Gnome Baker]] <br />
| N<br />
| GPL <br />
| [http://www.gnomefiles.org/app.php?soft_id=291 here] <br />
| align="left" | Popular open-source Gnome interface CD/DVD writing application<br />
|- style="background-color:#eeeeee;"<br />
! align="left" | [[K3b]]<br />
| N<br />
| GPL<br />
| [http://www.k3b.org/ here]<br />
| align="left" | Popular open-source KDE CD writing application for Linux platform <br />
|- <br />
! align="left" | [[X-CD-Roast]] <br />
| Y <br />
| GPL <br />
| [http://www.xcdroast.org here] <br />
| align="left" | New open-source Gnome interface CD/DVD writing application<br />
|- style="background-color:#eeeeee;"<br />
! align="left" | [[Brasero]]<br />
| N<br />
| GPL<br />
| [http://projects.gnome.org/brasero/ here]<br />
| align="left" | Brasero is a application to burn CD/DVD for the Gnome Desktop.(Gnome Default) <br />
|}<br />
<br />
==Multimedia Players==<br />
===Windows===<br />
{| border="0" cellpadding="0" cellspacing="1" style="text-align:center; border:2px solid #ccccff; margin-bottom: 20px;"<br />
|- style="background:#ccccff"<br />
! style="width:120px;" | Name<br />
! style="width:100px;" | License<br />
! style="width:100px;" | Website<br />
! align="center" style="width:220px;" | Description<br />
|-<br />
! align="left" | [[foobar2000]]<br />
| Free, BSD<br />
| [http://www.foobar2000.org/ here]<br />
| align="left" | Advanced tagging, plugin capabilities, and kernel streaming support<br />
|- style="background-color: #eeeeee;"<br />
! align="left" | [[MediaMonkey]]<br />
| Free, Shareware<br />
| [http://www.mediamonkey.com/ here]<br />
| align="left" | Supports many Winamp plugins<br />
|-<br />
! align="left" | MusikCube<br />
| BSD<br />
| [http://www.musikcube.com/ here]<br />
| align="left" | Supports dynamic playlists and advanced SQL capabilities <br />
|- style="background-color: #eeeeee;"<br />
! align="left" | VUplayer<br />
| Free<br />
| [http://www.vuplayer.com/ here]<br />
| align="left" | Supports many popular digital audio codecs and MOD tracker formats <br />
|-<br />
! align="left" | [[Winamp]]<br />
| Free, Shareware<br />
| [http://www.winamp.com/ here]<br />
| align="left" | Popular audio player for Windows<br />
| align="left" |<br />
|- style="background-color: #eeeeee;"<br />
! align="left" | [[wxMusik]]<br />
| GPL<br />
| [http://musik.berlios.de/ here]<br />
| align="left" |A cross-platform open-source audio player<br />
|- <br />
! align="left" | [[VLC]]<br />
| Free<br />
| [http://www.videolan.org/vlc// here]<br />
| align="left" | VLC media player is a highly portable multimedia player and multimedia framework capable of reading most audio and video formats as well as DVDs, Audio CDs VCDs, and various streaming protocols. <br />
|}<br />
<br />
===Mac OS X===<br />
{| border="0" cellpadding="0" cellspacing="1" style="text-align:center; border:2px solid #bbffbb; margin-bottom: 20px;"<br />
|- style="background:#bbffbb;"<br />
! style="width:120px;" | Name<br />
! style="width:100px;" | License<br />
! style="width:100px;" | Website<br />
! style="width:220px;" | Description<br />
|-<br />
! align="left" | Cog<br />
| GPL<br />
| [http://cogosx.sourceforge.net/ here]<br />
| align="left" | An open-source digital audio player for OS X.<br />
|- style="background-color: #eeeeee;"<br />
! align="left" | [[wxMusik]]<br />
| GPL<br />
| [http://musik.berlios.de/ here]<br />
| align="left" |A cross-platform open-source audio player<br />
|- <br />
! align="left" | Play<br />
| GPL<br />
| [http://sbooth.org/Play/ here]<br />
| align="left" |Play is an application for playing and managing audio files.<br />
|- style="background-color: #eeeeee;"<br />
! align="left" | [[VLC]]<br />
| Free<br />
| [http://www.videolan.org/vlc// here]<br />
| align="left" | VLC media player is a highly portable multimedia player and multimedia framework capable of reading most audio and video formats as well as DVDs, Audio CDs VCDs, and various streaming protocols. <br />
|}<br />
<br />
===Linux===<br />
{| border="0" cellpadding="0" cellspacing="1" style="text-align:center; border:2px solid #ffcccc; margin-bottom: 20px;"<br />
|- style="background:#ffcccc;"<br />
! style="width:120px;" | Name<br />
! style="width:100px;" | License<br />
! style="width:100px;" | Website<br />
! style="width:220px;" | Description<br />
|- style="background-color: #eeeeee;"<br />
! align="left" | [[Amarok]] <br />
| GPL <br />
| [http://amarok.kde.org/ here] <br />
| align="left" | Popular open-source KDE audio player similiar to Foobar2000<br />
|- <br />
! align="left" | [[wxMusik]]<br />
| GPL<br />
| [http://musik.berlios.de/ here]<br />
| align="left" |A cross-platform open-source audio player<br />
|- style="background-color: #eeeeee;"<br />
! align="left" | [[XMMS]] <br />
| GPL <br />
| [http://www.xmms.org/ here] <br />
| align="left" | Popular open-source audio player similiar to Winamp <br />
|- <br />
! align="left" | [[VLC]]<br />
| Free<br />
| [http://www.videolan.org/vlc// here]<br />
| align="left" | VLC media player is a highly portable multimedia player and multimedia framework capable of reading most audio and video formats as well as DVDs, Audio CDs VCDs, and various streaming protocols. <br />
|}<br />
<br />
===PocketPC===<br />
''These players may not play all your media files. Check their websites for the format support.''<br />
* GSPlayer: http://hp.vector.co.jp/authors/VA032810/<br />
* MortPlayer: http://www.sto-helit.de/<br />
* TCPMP: http://tcpmp.corecodec.org/about<br />
<br />
==Tagging Utilities==<br />
===Windows===<br />
{| border="0" cellpadding="0" cellspacing="1" style="text-align:center; border:2px solid #ccccff; margin-bottom: 20px;"<br />
|- style="background:#ccccff"<br />
! style="width:150px;" | Name<br />
! style="width:100px;" | License<br />
! style="width:100px;" | Website<br />
! style="width:270px;" | Description<br />
|-<br />
! align="left" | Abander TagControl<br />
| Shareware<br />
| [http://www.softartstudio.com/tagcontrol/ here]<br />
| <br />
|- style="background-color: #eeeeee;"<br />
! align="left" | AudioShell<br />
| Free<br />
| [http://www.softpointer.com/AudioShell.htm here]<br />
| align="left" | Integrates with Windows Explorer<br />
|- <br />
! align="left" | Frontah<br />
| Free<br />
| [http://home.vxu.se/mdati00/frontah/ here]<br />
| align="left" | Transcode and tag editor for ID3v1.x, ID3v2.x, Lyrics3, Vorbis Comment, APEv1 & APEv2 tags. Supports ANSI, UTF8 and UTF16 text encoding depends on tag type.<br />
|- style="background-color: #eeeeee;"<br />
! align="left" | Magic MP3 Tagger<br />
| Shareware<br />
| [http://www.magic-tagger.com here]<br />
| align="left" | Optimized for automatic music identification<br />
|- <br />
! align="left" | [[MediaMonkey]]<br />
| Free, Shareware<br />
| [http://www.mediamonkey.com/ here]<br />
| align="left" | Also a Media Player & Library<br />
|- style="background-color: #eeeeee;"<br />
! align="left" | MetatOGGer<br />
| Free<br />
| [http://fireblast.free.fr/ here]<br />
| align="left" | Tags MP3 ([[ID3]]) and Ogg files (Vorbis comment, including Ogg FLAC and Speex)<br />
|-<br />
! align="left" | MP3 Book Helper<br />
| Free<br />
| [http://mp3bookhelper.sourceforge.net/ here]<br />
| align="left" | Tags [[ID3v1]], ID3v2.3, and Vorbis comments. Features: FreeDB, unicode, guessing and matching, and supporting PAR, SFV, SV, and NFO generation.<br />
|- style="background-color: #eeeeee;"<br />
! align="left" | [[MP3tag]]<br />
| Free<br />
| [http://www.mp3tag.de/ here]<br />
| align="left" | Tags all files supporting [[ID3]], [[APEv2]], and [[Vorbis_Comment|Vorbis Comments]], not only MP3s<br />
|-<br />
! align="left" | [http://www.mp3-tag.com/ MP3 Tag Editor]<br />
| Shareware<br />
| [http://www.mp3-tag.com/ here]<br />
| align="left" | Software to edit tags in audio files of [[MP3]], [[WMA]], [[OGG]], [[ASF]], and other music format.<br />
|- style="background-color: #eeeeee;"<br />
! align="left" | Mp3/Tag Studio<br />
| Shareware<br />
| [http://www.magnusbrading.com/mp3ts/ here]<br />
| align="left" | Supports ID3v1 & v2 '''only'''. Powerful matching and fancy filters<br />
|-<br />
! align="left" | [[Tag.exe]]<br />
| GPL<br />
| [http://www.synthetic-soul.co.uk/tag/ here]<br />
| align="left" | Command-line universal tagger for Windows<br />
|- style="background-color: #eeeeee;"<br />
! align="left" | Tag &amp; Rename<br />
| Shareware<br />
| [http://www.softpointer.com/tr.htm here]<br />
|<br />
|-<br />
! align="left" | TagScanner<br />
| Free/Donate<br />
| [http://xdev.narod.ru/tagscan_e.htm here]<br />
|<br />
|- style="background-color: #eeeeee;"<br />
! align="left" | The GodFather<br />
| Card/Donate<br />
| [http://users.otenet.gr/~jtcliper/tgf/ here]<br />
|<br />
|-<br />
! align="left" | [http://wmptagext.sourceforge.net/download.html WMPTSE]<br />
| Free/Donate<br />
| [http://wmptagext.sourceforge.net here]<br />
| align="left" | Software to integrate other tag format than [[ID3]] into Microsoft Windows Media Player.<br />
|}<br />
<br />
===Mac OS X===<br />
{| border="0" cellpadding="0" cellspacing="1" style="text-align:center; border:2px solid #bbffbb; margin-bottom: 20px;"<br />
|- style="background:#bbffbb;"<br />
! style="width:150px;" | Name<br />
! style="width:100px;" | License<br />
! style="width:100px;" | Website<br />
! style="width:270px;" | Description<br />
|-<br />
! align="left" | Tag<br />
| GPL<br />
| [http://sbooth.org/Tag/ here]<br />
| align="left" | An open-source tagging application for OS X<br />
|}<br />
<br />
===Linux===<br />
{| border="0" cellpadding="0" cellspacing="1" style="text-align:center; border:2px solid #ffcccc; margin-bottom: 20px;"<br />
|- style="background:#ffcccc;"<br />
! style="width:150px;" | Name<br />
! style="width:100px;" | License<br />
! style="width:100px;" | Website<br />
! style="width:270px;" | Description<br />
|-<br />
! align="left" | EasyTAG<br />
| GPL<br />
| [http://easytag.sourceforge.net/ here]<br />
| align="left" | Gnome tagging utility<br />
|- style="background-color: #eeeeee;"<br />
! align="left" | <br />
|}<br />
<br />
==Encoders, Decoders, Etc.==<br />
All basic tools needed to make use of the audio formats supported here.<br />
<br />
===[[MP3]]===<br />
* [[LAME]] encoder/decoder: [http://www.rarewares.org/mp3.html download pre-compiled binaries here]. Also check the [[Lame Compiles|Latest recommended version]] page.<br />
* [[MP3Gain]], a Replay Gain-like utility: [http://mp3gain.sourceforge.net/download.php download here]<br />
<br />
===Ogg [[Vorbis]]===<br />
Currently, all recommended Ogg Vorbis utilities are available at the [http://www.rarewares.org/ogg.html Rarewares Ogg Vorbis page]. The following tools are important:<br />
<br />
* '''OggEnc2''': A command-line Ogg Vorbis encoder that can be used with most CD rippers.<br />
<br />
* '''OggDec''': Command-line decoder.<br />
<br />
* '''[[OggDropXPd]]''': An easy to use, drag'n'drop encoder/decoder with support for automatic tagging, renaming and playlist creation on encoding.<br />
<br />
* ''Encoding DLLs'': For encoding within CDex or WinLame.<br />
<br />
* '''VorbisGain''': The [[Replay Gain]] utility for Ogg Vorbis.<br />
<br />
In addition, the [[Lancer]] suite — a highly SSE-optimized suite of utilities and libraries — are available at [http://homepage3.nifty.com/blacksword/ this page] ''(in Japanese)''. See [[Lancer#Platform-specific Builds|this section]] for information about the different builds.<br />
<br />
===[[Musepack]] (MPC)===<br />
* [http://www.musepack.net/index.php?pg=win Download MPC for Windows]<br />
* [http://www.musepack.net/index.php?pg=lin Download MPC for Linux]<br />
* [http://www.musepack.net/index.php?pg=osx Download MPC for Mac OS X]<br />
* [http://www.musepack.net/index.php?pg=src Download MPC source code]<br />
<br />
* [http://forum.musepack.net/showthread.php?t=395 Forum announcement of SV8 release]<br />
<br />
===[[FLAC]]===<br />
* CoolEdit / Adobe Audition Filter supporting FLAC: [http://www.vuplayer.com/other.php download here]<br />
* Various FLAC-related utilities (incl. Replay Gain utility): [http://flac.sourceforge.net/download.html FLAC's SourceForge Download page]<br />
<br />
==Transcoders==<br />
''Note: Although these tools may convert from one encoding to another, please remember that [[transcoding]] to any [[lossy]] encoding <u>will</u> result in a degraded quality.''<br />
* BeSweet: http://besweet.notrace.dk/<br />
* [[BonkEnc]]<br />
* dBpowerAMP Music Converter (dMC): http://www.dbpoweramp.com/dmc.htm<br />
* [[foobar2000]] (needs 3rd-party encoders)<br />
* MediaCoder: http://www.rarewares.org/mediacoder/<br />
* Omni Encoder: http://omniencoder.autobotcity.net/<br />
* [[Winamp]]<br />
* WinLAME: http://winlame.sourceforge.net/<br />
<br />
==Processing utilities==<br />
===Windows===<br />
{| border="0" cellpadding="0" cellspacing="1" style="text-align:center; border:2px solid #ccccff; margin-bottom: 20px;"<br />
|- style="background:#ccccff;"<br />
! style="width:120px;" | Name<br />
! style="width:100px;" | License<br />
! style="width:100px;" | Website<br />
! style="width:400px;" | Description<br />
|- style="background-color: #eeeeee;"<br />
! align="left" | [[lossyWAV]]<br />
| GPL<br />
| [http://www.hydrogenaudio.org/forums/index.php?s=&showtopic=56129&view=findpost&p=504087 here]<br />
| align="left" | lossyWAV is a lossy pre-processor for [[Wikipedia:Pulse-code modulation|PCM]] (uncompressed) WAV files. It reduces [[Wikipedia:Audio bit depth|bit depth]] of the input signal, which, when used in conjunction with certain lossless codecs, reduces the bitrate of the encoded file significantly.<br />
|}<br />
<br />
==Drivers==<br />
===ASPI===<br />
* Ahead Nero ASPI Driver: [ftp://ftp6.nero.com/wnaspi32.dll download]<br />
* Adaptec Windows ASPI Package: [http://www.adaptec.com/worldwide/support/suppdetail.jsp?sess=no&prodkey=ASPI-4.70 official website]<br />
* ForceASPI [http://radified.com/ASPI/forceaspi.htm radified.com]<br />
* ASPI4all [http://www.cdr-zone.com/software/aspi_layers/aspi4all.html CDR-Zone.COM]<br />
* FrogAspi [http://www.frogaspi.org/ official website]<br />
* VOB ASAPI Driver 1.3: [http://www.rarewares.org/files/ASAPI.exe download]<br />
<br />
===Sound===<br />
* ALSA Project [http://www.alsa-project.org/ official website]<br />
* kX Project [http://kxproject.lugosoft.com/ official website]<br />
* ZonaISIS [http://www.hispasonic.com/zonaisis/index.htm unofficial]<br />
* I have a dream ... [http://members.aol.com/cridi/ unofficial]<br />
<br />
==Links==<br />
* [http://www.reactos.org/wiki/index.php/Untested_%28open_source%29_software_list Open source softwares @ ReactOS wiki]<br />
* [http://www.rarewares.org/ RareWares]<br />
<br />
<br />
[[Category:Software]]</div>Gottkaiserhttps://wiki.hydrogenaud.io/index.php?title=LossyWAVLossyWAV2010-02-02T10:18:18Z<p>Gottkaiser: adding category software</p>
<hr />
<div>{{Software Infobox<br />
| name = lossyWAV<br />
| screenshot = <br />
| caption = <br />
| maintainer = [http://www.hydrogenaudio.org/forums/index.php?showuser=42400 Nick.C]<br />
| stable_release = 1.2.0<br />
| preview_release = <none><br />
| operating_system = [[Wikipedia:Microsoft Windows|Windows]]<br />
| use = [[Wikipedia:Digital signal processing|Digital signal processing]]<br />
| license = [[Wikipedia:GNU General Public License|GNU GPL]]<br />
| website = [http://www.hydrogenaudio.org/forums/index.php?s=&showtopic=64666&view=findpost&p=577042 Hydrogenaudio]<br />
}}<br />
lossyWAV is a [[Wikipedia:Free software|free]], [[lossy]] pre-processor for [[PCM]] audio contained in the [[RIFF_WAVE|WAV]] file format. Proposed by [http://www.hydrogenaudio.org/forums/index.php?showuser=409 David Robinson], it reduces [[Wikipedia:Audio bit depth|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.<br />
lossyWAV's primary goal is to maintain [[transparency]] with a high degree of confidence when processing any audio data.<br />
<br />
==History==<br />
lossyWAV is based on the lossyFLAC idea proposed by [http://www.hydrogenaudio.org/forums/index.php?showuser=409 David Robinson] at Hydrogenaudio, which is a method of carefully reducing the bitdepth of samples, therefore utilising the wasted bits feature of the FLAC lossless codec. The aim is to transparently reduce audio bit depth (by making some lower significant bits ([[Wikipedia:Least_significant_bit|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.[http://www.hydrogenaudio.org/forums/index.php?s=&showtopic=55522&view=findpost&p=498179] 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.<br />
<br />
[http://www.hydrogenaudio.org/forums/index.php?showuser=42400 Nick Currie] ported the original [[Wikipedia:MATLAB|MATLAB]] implementation to [[Wikipedia:Borland Delphi|Delphi]] (Many thanks [[Wikipedia:CodeGear|CodeGear]] for Turbo Explorer!!) with a liberal sprinkling of [[Wikipedia:IA-32|IA-32]] and [[Wikipedia:x87|x87]] Assembly Language for speed.<br />
<br />
Subsequently, lossyFLAC proved itself to work with other lossless codecs, so the application name was changed to lossyWAV. <br />
<br />
Since then, Nick has heavily developed and built upon lossyWAV, with valuable tuning performed by [http://www.hydrogenaudio.org/forums/index.php?showuser=25015 Horst Albrecht] at Hydrogenaudio. Although the current lossyWAV implementation has built on David's original method, the method itself still very much belongs to its author.<br />
<br />
==Indicative bitrate reduction==<br />
It must be stressed that lossyWAV is a pure variable bit-depth pre-processor in that the overall sample size remains the same after processing but the number of significant bits used for the samples in a codec-block can change on a block-by-block basis. 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 [[Wikipedia:fast Fourier transform|fast Fourier Transform]] (FFT) analyses of at least two lengths (default quality preset (-q 5) = 32, 64 & 1024 [[Wikipedia:Sampling %28signal processing%29|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 [[Wikipedia:white noise|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.<br />
<br />
{| class="wikitable" style="text-align:center"<br />
|-<br />
!lossyWAV Test Set (16 bit / 44.1kHz)<br />
!Codec<br />
!lossless<br />
!--insane<br />
!--extreme<br />
!--standard<br />
!--portable<br />
!--zero<br />
|-<br />
!10 Album Test Set<br />
| TAK<br />
| 820 kbit/s<br />
| ??? kbit/s<br />
| ??? kbit/s<br />
| ??? kbit/s<br />
| ??? kbit/s<br />
| ??? kbit/s<br />
|-<br />
!10 Album Test Set<br />
| FLAC<br />
| 854 kbit/s<br />
| 627 kbit/s<br />
| 544 kbit/s<br />
| 460 kbit/s<br />
| 376 kbit/s<br />
| 288 kbit/s<br />
|-<br />
!10 Album Test Set<br />
| Wavpack<br />
| 852 kbit/s<br />
| ??? kbit/s<br />
| ??? kbit/s<br />
| ??? kbit/s<br />
| ??? kbit/s<br />
| ??? kbit/s<br />
|}<br />
<br />
==File identification==<br />
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.[http://www.hydrogenaudio.org/forums/index.php?s=&showtopic=55522&view=findpost&p=498559]<br />
<br />
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.<br />
<br />
Combinations of lossyWAV with each specific encoder are referred to as lossy'''X''', where '''X''' is an abbreviation of the lossless codec name. Combination names are listed in the "[[LossyWAV#Known supported codecs|known supported codecs]]" section below.<br />
<br />
lossyWAV inserts a variable-length 'fact' chunk into the WAV file immediately after the 'fmt ' chunk. This takes the form:<pre>fact/<size>/lossyWAV x.y.z @ dd/mm/yyyy hh:mm:ss, -q 5</pre>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.<br />
<br />
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.<br />
<br />
==Quality presets==<br />
*--insane: (-q 10) Highest quality preset, generally considered to be excessive;<br />
*--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;<br />
*--standard: (-q 5) Default preset, generally accepted to be transparent;<br />
*--portable: (-q 2.5) DAP quality preset for use on a compatible [[Wikipedia:Digital audio player|DAP]].[http://www.hydrogenaudio.org/forums/index.php?s=&showtopic=56129&view=findpost&p=531316]<br />
<br />
All tuning has been performed on quality preset --standard with higher presets being more conservative. Quality preset --standard is generally accepted to be (and from testing so far is) transparent. If you find a track which --standard fails to achieve transparency after processing, please post a sample (no more than 30 seconds) in the development thread.<br />
<br />
The --altpreset parameter was introduced at 1.2.0 which creates a second quality range using modified internal presets and extends the quality range from -4 to 10 (--quality -4 --altpreset is equivalent to --quality 0 --limit 15159 in the default quality range).<br />
<br />
==Supported input formats==<br />
*[[WAV]]: 9-bit to 32-bit integer; 1 to 8 channels; sample rate &ge; 32kHz [[Pulse Code Modulation|PCM]]. Very high sample rates (&gt;48kHz) have not been extensively tested. Tunings have been focussed on 16-bit, 44.1kHz samples (i.e. [[Wikipedia:Red Book (audio CD standard)|CD]] PCM).<br />
<br />
==Codec compatibility==<br />
{| class="wikitable" style="text-align:center"<br />
|-<br />
!Codec<br />
!Supported<br />
!Encoder parameters<br />
!Combination name<br />
|-<br />
! [[Free Lossless Audio Codec|FLAC]]<br />
| '''Yes'''<br />
| -'''5''' -'''b''' 512 --'''keep-foreign-metadata'''<br />
| lossy'''FLAC'''<br />
|-<br />
! [[Lossless Predictive Audio Compression|LPAC]]<br />
| '''Yes'''<br />
| -'''b'''512<br />
| lossy'''LPAC'''<br />
|-<br />
! [[Wikipedia:Audio Lossless Coding|MPEG-4 ALS]]<br />
| '''Yes'''<br />
| -'''l''' -'''n'''512<br />
| lossy'''ALS'''<br />
|-<br />
! [[TAK]]<br />
| '''Yes'''<br />
| -'''fsl'''512<br />
| lossy'''TAK'''<br />
|-<br />
! [[WavPack]]<br />
| '''Yes'''<br />
| --'''blocksize'''=512<br />
| lossy'''WV'''<br />
|-<br />
! [[Windows Media Audio#Windows Media Audio Lossless|WMA Lossless]]<br />
| '''Yes'''<br />
| &mdash;<br />
| lossy'''WMALSL'''<br />
|-<br />
! [[Apple Lossless]]<br />
| No<br />
| &mdash;<br />
| &mdash;<br />
|-<br />
! [[Lossless Audio|LA]]<br />
| No<br />
| &mdash;<br />
| &mdash;<br />
|-<br />
! [[Monkey's Audio]]<br />
| No<br />
| &mdash;<br />
| &mdash;<br />
|-<br />
! [[OptimFROG]]<br />
| No<br />
| &mdash;<br />
| &mdash;<br />
|-<br />
! [[Wikipedia:TTA (codec)|TTA]]<br />
| No<br />
| &mdash;<br />
| &mdash;<br />
|}<br />
<br />
* Combinations of lossyWAV with each specific encoder are referred to as lossy'''X''', where '''X''' is an abbreviation of the lossless codec name.<br />
<br />
<br />
There is also [http://www.hometheaterhifi.com/volume_8_4/dvd-benchmark-part-6-dvd-audio-11-2001.html#Meridian%20Lossless%20Packing%20(MLP)%20in%20a%20Nutshell evidence] &mdash; so-called "Bit Shifting" &mdash; to suggest that lossyWAV may work with [[Wikipedia:Meridian Lossless Packing|MLP]], but this remains untested due to prohibitive prices of encoders.<br />
<br />
A comparison of portable media players is [[Wikipedia:Comparison of portable media players#Audio Formats|here]], which shows FLAC and WMA Lossless compatibility among listed players.<br />
Any player supported by [http://www.rockbox.org Rockbox] can use FLAC or WavPack files after installing Rockbox.<br />
===Important note===<br />
'''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.'''<br />
===Bonus feature===<br />
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.<br />
<br />
==Using lossyWAV==<br />
===Application settings===<br />
<pre><br />
lossyWAV 1.2.0, Copyright (C) 2007,2008,2009 Nick Currie. Copyleft.<br />
<br />
This program is free software: you can redistribute it and/or modify it under<br />
the terms of the GNU General Public License as published by the Free Software<br />
Foundation, either version 3 of the License, or (at your option) any later<br />
version.<br />
<br />
This program is distributed in the hope that it will be useful,but WITHOUT ANY<br />
WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR A<br />
PARTICULAR PURPOSE. See the GNU General Public License for more details.<br />
<br />
You should have received a copy of the GNU General Public License along with<br />
this program. If not, see <http://www.gnu.org/licenses/>.<br />
<br />
Usage : lossyWAV <input wav file> <options><br />
<br />
Example : lossyWAV musicfile.wav<br />
<br />
Quality Options:<br />
<br />
-I, --insane highest quality output, suitable for transcoding;<br />
-E, --extreme high quality output, also suitable for transcoding;<br />
-S, --standard default quality output, considered to be transparent;<br />
-P, --portable good quality output for DAP use, not fully transparent.<br />
-Z, --zero lowest quality preset, probably contains artifacts.<br />
<br />
Standard Options:<br />
<br />
-C, --correction write correction file for processed WAV file; default=off.<br />
-f, --force forcibly over-write output file if it exists; default=off.<br />
-h, --help display help.<br />
-L, --longhelp display extended help.<br />
-M, --merge merge existing lossy.wav and lwcdf.wav files.<br />
-o, --outdir <t> destination directory for the output file(s).<br />
-v, --version display the lossyWAV version number.<br />
-w, --writetolog create (or add to) lossyWAV.log in the output directory.<br />
<br />
Special thanks go to:<br />
<br />
David Robinson for the publication of his lossyFLAC method, guidance, and<br />
the motivation to implement his method as lossyWAV.<br />
<br />
Horst Albrecht for ABX testing, valuable support in tuning the internal<br />
presets, constructive criticism and all the feedback.<br />
<br />
Sebastian Gesemann for the noise shaping coefficients and help in using them<br />
in the lossyWAV noise shaping implementation.<br />
<br />
Matteo Frigo and for the excellent libfftw3-3.dll contained in the FFTW<br />
Steven G Johnson distribution (v3.2.1 or v3.2.2).<br />
<br />
Mark G Beckett for the Delphi unit that provides an interface to the<br />
(Univ. of Edinburgh) relevant fftw routines in libfftw3-3.dll.<br />
<br />
Don Cross for the Complex-FFT algorithm originally used.</pre><br />
<br />
===Example drag 'n' drop batch file===<br />
Simply drag the FLAC files onto this batch file and it will process, recode in FLAC and copy ALL of the tags from the input FLAC file, placing the output lossyFLAC file in the same directory as the input FLAC file. Requires flac.exe and [http://www.synthetic-soul.co.uk/tag/ tag.exe] to be somewhere on the path. <br />
<pre>@echo off<br />
:repeat<br />
if %1.==. goto end<br />
if exist %1 flac -d %1 --stdout --silent|lossywav - --stdout --standard --stdinname %1|flac - -b 512 -o "%~dpn1.lossy.flac" --silent && tag --fromfile %1 "%~dpn1.lossy.flac"<br />
shift<br />
goto repeat<br />
:end</pre><br />
<br />
===lossyWAV and FFTW===<br />
Since version 1.2.0, lossyWAV has been compatible with [[Wikipedia:FFTW|FFTW]] although not dependent on it. Should the user wish to take advantage of the increased processing speed available when using FFTW (from superior FFT implementations), libfftw3-3.dll should be placed in a directory on the host computer which features on the path.<br />
<br />
===lossyWAV with WINE===<br />
The cause of lossyWAV's WINE incompatibility was found and removed during the development of 1.2.0 and retrospectively amended for 1.1.0b in a maintenance release (1.1.0c).<br />
<br />
===Example [[foobar2000]] converter settings===<br />
lossyFLAC settings:<pre>Encoder: C:\Windows\System32\cmd.exe<br />
Extension : lossy.flac<br />
Parameters: /d /c C:\"Program Files"\bin\lossywav - --standard --silent --stdout|<br />
C:\"Program Files"\bin\flac - -b 512 -5 -f -o%d<br />
Format is : lossless or hybrid<br />
Highest BPS mode supported: 24 </pre><br />
<br />
lossyTAK settings:<pre>Encoder: C:\Windows\System32\cmd.exe<br />
Extension : lossy.tak<br />
Parameters : /d /c C:\"Program Files"\bin\lossywav - --standard --silent --stdout|<br />
C:\"Program Files"\bin\takc -e -p2m -fsl512 -ihs - %d<br />
Format is: lossless or hybrid<br />
Highest BPS mode supported: 24</pre><br />
<br />
lossyWV settings:<pre>Encoder: C:\Windows\System32\cmd.exe<br />
Extension : lossy.wv<br />
Parameters: /d /c C:\"Program Files"\bin\lossywav - --standard --silent --stdout|<br />
C:\"Program Files"\bin\wavpack -hm --blocksize=512 --merge-blocks -i - %d<br />
Format is : lossless or hybrid<br />
Highest BPS mode supported: 24</pre><br />
<br />
There is a known problem within foobar2000 (although more likely to do with cmd.exe itself) when running an executable within the cmd.exe command line from a path which includes spaces. The suggested fix for this is to enclose the element of the path which contains spaces within double quotation marks ("), e.g. C:\"Program Files"\directory_where_executable_is\executable_name<br />
<br />
===Example EAC settings===<br />
:''See [[EAC and LossyWAV]].''<br />
<br />
==Frequently asked questions==<br />
*'''Question:''' Why is the ".wav" file extension used?<br />
*'''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.<br />
<br />
*'''Question:''' Why create a processor which means that I cannot be sure that a lossless file is truly lossless?<br />
*'''Answer:''' Unless one creates the lossless file personally, one can '''never''' be completely sure that the file is indeed lossless. E.g. a lossless file you receive could be transcoded from [[MP3]] without your knowledge. To distinguish a lossyWAV file from lossless files it is recommended to use the extension .lossy.EXT where EXT is the original extension e.g. .lossy.flac<br />
<br />
*'''Question:''' Is it [[Variable Bitrate|VBR]]?<br />
*'''Short answer:''' Yes.<br />
<br />
*'''Question:''' Do I need to re-process to change lossless codecs?<br />
*'''Short answer:''' No.<br />
<br />
*'''Question:''' Is it [[transparency|transparent]]?<br />
*'''Short answer:''' At preset --standard, almost certainly.<br />
<br />
*'''Question:''' Is it [[lossless]]?<br />
*'''Short answer:''' No.<br />
<br />
*'''Question:''' Will it ever have a [[Constant Bitrate|CBR]] mode?<br />
*'''Short answer:''' No.<br />
<br />
*'''Question:''' Why should I use this?<br />
*'''Answer:'''<br />
:*high quality<br />
:*extremely low chance of audible [[artifact]]s<br />
:*reasonable [[bitrate]]s<br />
:*usable with unmodified, established lossless formats.<br />
<br />
==External links==<br />
*[http://www.hydrogenaudio.org/forums/index.php?showtopic=55522 Original lossyFLAC thread] - Introduction of the concept by David Robinson (Replay Gain developer) and initial development<br />
----<br />
*[http://www.hydrogenaudio.org/forums/index.php?showtopic=65499 lossyWAV 1.2.0 development thread]<br />
*[http://www.hydrogenaudio.org/forums/index.php?s=&showtopic=77042 lossyWAV 1.2.0 release thread] - Release of version 1.2.0 on 16 December 2009<br />
----<br />
*[http://www.hydrogenaudio.org/forums/index.php?showtopic=63254 lossyWAV 1.1.0 development thread]<br />
*[http://www.hydrogenaudio.org/forums/index.php?showtopic=64617 lossyWAV 1.1.0 release thread] - Release of version 1.1.0 on 12 July 2008<br />
----<br />
*[http://www.hydrogenaudio.org/forums/index.php?showtopic=56129 lossyWAV Development thread] - Conversion of the original MATLAB script to Delphi and evolution of the method<br />
*[http://www.hydrogenaudio.org/forums/index.php?showtopic=63225 lossyWAV 1.0.0 release thread] - Release of version 1.0.0b on 12 May 2008<br />
<br />
[[Category:Software]]</div>Gottkaiserhttps://wiki.hydrogenaud.io/index.php?title=TAKTAK2009-01-06T16:46:35Z<p>Gottkaiser: /* Windows */</p>
<hr />
<div>{{Codec Infobox<br />
| name = Tom's lossless Audio Kompressor<br />
| logo =<br />
| type = lossless<br />
| purpose = lossless audio compression.<br />
| maintainer = Thomas Becker<br />
| recommended_encoder = TAK encoder<br />
| recommended_text = TAK v1.1.0<br />
| website = [http://thbeck.de/Tak/Tak.html ThBeck.de/Tak/Tak.html] ''(german)''<br />
}}<br />
<br />
== Description ==<br />
'''Tom's lossless Audio Kompressor''' ('''TAK''') is a lossless audio compressor which promises compression performance similar to [[Monkey's Audio]] “High” and decompression speed similar to [[Free Lossless Audio Codec|FLAC]].<br />
<br />
=== Features ===<br />
* High compression<br />
* Fast compression and decompression speed<br />
* Streaming support (necessary headers for decompressing the audio are written to the stream every 2 seconds)<br />
* Piping support for encoding<br />
* Error tolerance (single bit error will never affect more than 250 ms)<br />
* Error detection (each frame protected by a 24-bit checksum (CRC))<br />
* High-resolution (up to 24-bit/channel) audio support<br />
* Support for up to 192 Khz Audio<br />
* Seeking without seek table<br />
* APEv2 tags supported at end of file<br />
<br />
=== Pros ===<br />
* Fast encoding speed (while providing better compression TAK encodes as fast as [[Free Lossless Audio Codec|FLAC]] -8 in TAK's “Insane” and several times faster in “Turbo” mode)<br />
* Fast decompression speed (on par with FLAC / [[WavPack]])<br />
* Good compression levels (on par with [[Monkey's Audio]] High)<br />
* Error Robustness<br />
* Fast Seeking<br />
<br />
=== Cons ===<br />
* Closed Source (at the moment)<br />
* No hardware support<br />
* Very limited software support (playback: [[Winamp]] & [[foobar2000]] plugins, tagging: Mp3Tag)<br />
<br />
<br />
== Hardware and Software That Support TAK ==<br />
=== Hardware ===<br />
* None<br />
<br />
=== Software ===<br />
==== Windows ====<br />
* offical TAK Applications v1.1.0 (Applications, Winamp plugin, SDK, Decoding library) [http://www.hydrogenaudio.org/forums/index.php?showtopic=68456&st=0 here]<br />
* foo_input_tak, TAK decoder for [[foobar2000]] [http://foosion.foobar2000.org/components/ here] (supports tagging and [[Replay Gain]])<br />
* [[Mp3tag]] – universal tag editor with support for TAK<br />
* [http://etree.org/shnutils/shntool/ shntool] (since version 3.0.6)<br />
<br />
==== Linux ====<br />
* The TAK reference applications (GUI as well as commandline) are known to run on Linux via Wine.<br />
<br />
<br />
== Recommended Settings ==<br />
* Default compression: “-p2” (formerly ''Normal'') is the most attractive setting, providing an excellent compromise between compression and encoding speed. (At compression levels close to [[Monkey's Audio]] High (<0.4% difference), it is able to encode more quickly.)<br />
takc -e [input file]<br />
* Highest compression: “-pMax” (same as -p5m) (This will create files which are comparable in size to file created using [[Monkey's Audio]] High. Decompression speed is comparable to [[WavPack]] Normal.)<br />
takc -e -pMax [input file]<br />
* Fastest compression: “-p0” (This will create files which are comparable in size to [[Monkey's Audio]] Fast or [[WavPack]] High. Decompression speed is comparable to [[Free Lossless Audio Codec|FLAC]] 0.)<br />
takc -e -p0 [input file]<br />
<br />
=== TAK Performance Graph ===<br />
[[Image:TAK_performance_graph_1-0-4.png|frame|center|Graph showing encoding and decoding rate against compression, using data from Synthetic Soul's test on TAK 1.0.4<br />(see [[TAK#External Links|External Links]])]]<br />
<br />
<br />
== Using TAK ==<br />
=== TAK with [[foobar2000]] ===<br />
* Copy the takc.exe to your [[foobar2000]] directory<br />
* Go to File → Preferences → Tools → Converter<br />
* Set it up as shown:<br />
[[Image:Tak_foobar_converter.png|frame|center|Screenshot of foobar 0.9.5 Converter settings for TAK 1.0.3]]<br />
'''Note:''' replace -p2 with the desired compression level.<br />
<br />
* TAK introduced encoding from STDIN in version 1.0.3, eliminating the need for a temporary file and greatly improving overall compression time. If you are using an earlier version of TAK use the following command line instead:<br />
-e -p2 %s %d<br />
* Use [[APEv2 specification|APEv2]] tagging (will be used as internal tagging)<br />
<br />
<br />
=== TAK with EAC ===<br />
Please read the [[EAC and TAK|wiki guide]], which details how to create TAK files with [[Exact Audio Copy|EAC]].<br />
<br />
<br />
== Future Features ==<br />
* Unicode support<br />
* MD5 audio checksums for verification and identification<br />
* A German version<br />
* Embedded cue sheets<br />
* Embedded cover art<br />
* Multichannel audio<br />
<br />
<br />
== Frequently Asked Questions ==<br />
; Is the codec safe for use?<br />
: Yes. To check, convert a WAVE to TAK and back and compare the two (or use foobar's bitcompare tool).<br />
; Why should I use TAK?<br />
: TAK offers high compression ratios with great decoding rates.<br />
; What can I compress with TAK?<br />
: TAK 1.0 can compress any integer-format (up to 24 bits per channel) PCM RIFF WAVE file (.wav). Piping support as of v1.0.3 is implemented, so converting lossless files to WAV first is not necessary.<br />
; What about hardware support?<br />
: None at the moment. Although, ''-p0'', ''-p1'' and ''-p2'' are the candidates for hardware playback.<br />
; When will the source be opened?<br />
: Yes, TAK will be open-source, as soon as the code is ported to C or C++ and documented. However, Thomas has mentioned that he would like to improve the codec before opening the source.<br />
<br />
<br />
== External Links ==<br />
* [http://thbeck.de/Tak/Tak.html thbeck.de/Tak/Tak.html] – Official Website ''(german)''<br />
* [http://www.hydrogenaudio.org/forums/index.php?showtopic=68454 TAK 1.1.0 Release Announcement / Discussion Thread on HA] ''(english)''<br />
* [http://www.hydrogenaudio.org/forums/index.php?showtopic=68456&st=0 TAK 1.1.0 Downloads]<br />
* [http://synthetic-soul.co.uk/comparison/lossless/ synthetic-soul.co.uk/comparison/lossless] – Comparison with Other Codecs (by Synthetic Soul)<br />
* [http://flac.sourceforge.net/comparison.html flac.sourceforge.net/comparison.html] – An Updated Comparison (from FLAC Homepage)<br />
<br />
<br />
[[Category:Lossless]]<br />
[[Category:Encoder/Decoder]]</div>Gottkaiserhttps://wiki.hydrogenaud.io/index.php?title=TAKTAK2009-01-06T16:44:30Z<p>Gottkaiser: /* Windows */</p>
<hr />
<div>{{Codec Infobox<br />
| name = Tom's lossless Audio Kompressor<br />
| logo =<br />
| type = lossless<br />
| purpose = lossless audio compression.<br />
| maintainer = Thomas Becker<br />
| recommended_encoder = TAK encoder<br />
| recommended_text = TAK v1.1.0<br />
| website = [http://thbeck.de/Tak/Tak.html ThBeck.de/Tak/Tak.html] ''(german)''<br />
}}<br />
<br />
== Description ==<br />
'''Tom's lossless Audio Kompressor''' ('''TAK''') is a lossless audio compressor which promises compression performance similar to [[Monkey's Audio]] “High” and decompression speed similar to [[Free Lossless Audio Codec|FLAC]].<br />
<br />
=== Features ===<br />
* High compression<br />
* Fast compression and decompression speed<br />
* Streaming support (necessary headers for decompressing the audio are written to the stream every 2 seconds)<br />
* Piping support for encoding<br />
* Error tolerance (single bit error will never affect more than 250 ms)<br />
* Error detection (each frame protected by a 24-bit checksum (CRC))<br />
* High-resolution (up to 24-bit/channel) audio support<br />
* Support for up to 192 Khz Audio<br />
* Seeking without seek table<br />
* APEv2 tags supported at end of file<br />
<br />
=== Pros ===<br />
* Fast encoding speed (while providing better compression TAK encodes as fast as [[Free Lossless Audio Codec|FLAC]] -8 in TAK's “Insane” and several times faster in “Turbo” mode)<br />
* Fast decompression speed (on par with FLAC / [[WavPack]])<br />
* Good compression levels (on par with [[Monkey's Audio]] High)<br />
* Error Robustness<br />
* Fast Seeking<br />
<br />
=== Cons ===<br />
* Closed Source (at the moment)<br />
* No hardware support<br />
* Very limited software support (playback: [[Winamp]] & [[foobar2000]] plugins, tagging: Mp3Tag)<br />
<br />
<br />
== Hardware and Software That Support TAK ==<br />
=== Hardware ===<br />
* None<br />
<br />
=== Software ===<br />
==== Windows ====<br />
* offical TAK Applications v1.1.0 (Applications, Winamp, SDK, Decoding library) [http://www.hydrogenaudio.org/forums/index.php?showtopic=68456&st=0 here]<br />
* foo_input_tak, TAK decoder for [[foobar2000]] [http://foosion.foobar2000.org/components/ here] (supports tagging and [[Replay Gain]])<br />
* [[Mp3tag]] – universal tag editor with support for TAK<br />
* [http://etree.org/shnutils/shntool/ shntool] (since version 3.0.6)<br />
<br />
==== Linux ====<br />
* The TAK reference applications (GUI as well as commandline) are known to run on Linux via Wine.<br />
<br />
<br />
== Recommended Settings ==<br />
* Default compression: “-p2” (formerly ''Normal'') is the most attractive setting, providing an excellent compromise between compression and encoding speed. (At compression levels close to [[Monkey's Audio]] High (<0.4% difference), it is able to encode more quickly.)<br />
takc -e [input file]<br />
* Highest compression: “-pMax” (same as -p5m) (This will create files which are comparable in size to file created using [[Monkey's Audio]] High. Decompression speed is comparable to [[WavPack]] Normal.)<br />
takc -e -pMax [input file]<br />
* Fastest compression: “-p0” (This will create files which are comparable in size to [[Monkey's Audio]] Fast or [[WavPack]] High. Decompression speed is comparable to [[Free Lossless Audio Codec|FLAC]] 0.)<br />
takc -e -p0 [input file]<br />
<br />
=== TAK Performance Graph ===<br />
[[Image:TAK_performance_graph_1-0-4.png|frame|center|Graph showing encoding and decoding rate against compression, using data from Synthetic Soul's test on TAK 1.0.4<br />(see [[TAK#External Links|External Links]])]]<br />
<br />
<br />
== Using TAK ==<br />
=== TAK with [[foobar2000]] ===<br />
* Copy the takc.exe to your [[foobar2000]] directory<br />
* Go to File → Preferences → Tools → Converter<br />
* Set it up as shown:<br />
[[Image:Tak_foobar_converter.png|frame|center|Screenshot of foobar 0.9.5 Converter settings for TAK 1.0.3]]<br />
'''Note:''' replace -p2 with the desired compression level.<br />
<br />
* TAK introduced encoding from STDIN in version 1.0.3, eliminating the need for a temporary file and greatly improving overall compression time. If you are using an earlier version of TAK use the following command line instead:<br />
-e -p2 %s %d<br />
* Use [[APEv2 specification|APEv2]] tagging (will be used as internal tagging)<br />
<br />
<br />
=== TAK with EAC ===<br />
Please read the [[EAC and TAK|wiki guide]], which details how to create TAK files with [[Exact Audio Copy|EAC]].<br />
<br />
<br />
== Future Features ==<br />
* Unicode support<br />
* MD5 audio checksums for verification and identification<br />
* A German version<br />
* Embedded cue sheets<br />
* Embedded cover art<br />
* Multichannel audio<br />
<br />
<br />
== Frequently Asked Questions ==<br />
; Is the codec safe for use?<br />
: Yes. To check, convert a WAVE to TAK and back and compare the two (or use foobar's bitcompare tool).<br />
; Why should I use TAK?<br />
: TAK offers high compression ratios with great decoding rates.<br />
; What can I compress with TAK?<br />
: TAK 1.0 can compress any integer-format (up to 24 bits per channel) PCM RIFF WAVE file (.wav). Piping support as of v1.0.3 is implemented, so converting lossless files to WAV first is not necessary.<br />
; What about hardware support?<br />
: None at the moment. Although, ''-p0'', ''-p1'' and ''-p2'' are the candidates for hardware playback.<br />
; When will the source be opened?<br />
: Yes, TAK will be open-source, as soon as the code is ported to C or C++ and documented. However, Thomas has mentioned that he would like to improve the codec before opening the source.<br />
<br />
<br />
== External Links ==<br />
* [http://thbeck.de/Tak/Tak.html thbeck.de/Tak/Tak.html] – Official Website ''(german)''<br />
* [http://www.hydrogenaudio.org/forums/index.php?showtopic=68454 TAK 1.1.0 Release Announcement / Discussion Thread on HA] ''(english)''<br />
* [http://www.hydrogenaudio.org/forums/index.php?showtopic=68456&st=0 TAK 1.1.0 Downloads]<br />
* [http://synthetic-soul.co.uk/comparison/lossless/ synthetic-soul.co.uk/comparison/lossless] – Comparison with Other Codecs (by Synthetic Soul)<br />
* [http://flac.sourceforge.net/comparison.html flac.sourceforge.net/comparison.html] – An Updated Comparison (from FLAC Homepage)<br />
<br />
<br />
[[Category:Lossless]]<br />
[[Category:Encoder/Decoder]]</div>Gottkaiserhttps://wiki.hydrogenaud.io/index.php?title=TAKTAK2009-01-06T16:43:53Z<p>Gottkaiser: /* Windows */</p>
<hr />
<div>{{Codec Infobox<br />
| name = Tom's lossless Audio Kompressor<br />
| logo =<br />
| type = lossless<br />
| purpose = lossless audio compression.<br />
| maintainer = Thomas Becker<br />
| recommended_encoder = TAK encoder<br />
| recommended_text = TAK v1.1.0<br />
| website = [http://thbeck.de/Tak/Tak.html ThBeck.de/Tak/Tak.html] ''(german)''<br />
}}<br />
<br />
== Description ==<br />
'''Tom's lossless Audio Kompressor''' ('''TAK''') is a lossless audio compressor which promises compression performance similar to [[Monkey's Audio]] “High” and decompression speed similar to [[Free Lossless Audio Codec|FLAC]].<br />
<br />
=== Features ===<br />
* High compression<br />
* Fast compression and decompression speed<br />
* Streaming support (necessary headers for decompressing the audio are written to the stream every 2 seconds)<br />
* Piping support for encoding<br />
* Error tolerance (single bit error will never affect more than 250 ms)<br />
* Error detection (each frame protected by a 24-bit checksum (CRC))<br />
* High-resolution (up to 24-bit/channel) audio support<br />
* Support for up to 192 Khz Audio<br />
* Seeking without seek table<br />
* APEv2 tags supported at end of file<br />
<br />
=== Pros ===<br />
* Fast encoding speed (while providing better compression TAK encodes as fast as [[Free Lossless Audio Codec|FLAC]] -8 in TAK's “Insane” and several times faster in “Turbo” mode)<br />
* Fast decompression speed (on par with FLAC / [[WavPack]])<br />
* Good compression levels (on par with [[Monkey's Audio]] High)<br />
* Error Robustness<br />
* Fast Seeking<br />
<br />
=== Cons ===<br />
* Closed Source (at the moment)<br />
* No hardware support<br />
* Very limited software support (playback: [[Winamp]] & [[foobar2000]] plugins, tagging: Mp3Tag)<br />
<br />
<br />
== Hardware and Software That Support TAK ==<br />
=== Hardware ===<br />
* None<br />
<br />
=== Software ===<br />
==== Windows ====<br />
* offical TAK Applications v1.1.0 (Applications, Winamp, SDK, Decoding library [http://www.hydrogenaudio.org/forums/index.php?showtopic=68456&st=0 here]<br />
* foo_input_tak, TAK decoder for [[foobar2000]] [http://foosion.foobar2000.org/components/ here] (supports tagging and [[Replay Gain]])<br />
* [[Mp3tag]] – universal tag editor with support for TAK<br />
* [http://etree.org/shnutils/shntool/ shntool] (since version 3.0.6)<br />
<br />
==== Linux ====<br />
* The TAK reference applications (GUI as well as commandline) are known to run on Linux via Wine.<br />
<br />
<br />
== Recommended Settings ==<br />
* Default compression: “-p2” (formerly ''Normal'') is the most attractive setting, providing an excellent compromise between compression and encoding speed. (At compression levels close to [[Monkey's Audio]] High (<0.4% difference), it is able to encode more quickly.)<br />
takc -e [input file]<br />
* Highest compression: “-pMax” (same as -p5m) (This will create files which are comparable in size to file created using [[Monkey's Audio]] High. Decompression speed is comparable to [[WavPack]] Normal.)<br />
takc -e -pMax [input file]<br />
* Fastest compression: “-p0” (This will create files which are comparable in size to [[Monkey's Audio]] Fast or [[WavPack]] High. Decompression speed is comparable to [[Free Lossless Audio Codec|FLAC]] 0.)<br />
takc -e -p0 [input file]<br />
<br />
=== TAK Performance Graph ===<br />
[[Image:TAK_performance_graph_1-0-4.png|frame|center|Graph showing encoding and decoding rate against compression, using data from Synthetic Soul's test on TAK 1.0.4<br />(see [[TAK#External Links|External Links]])]]<br />
<br />
<br />
== Using TAK ==<br />
=== TAK with [[foobar2000]] ===<br />
* Copy the takc.exe to your [[foobar2000]] directory<br />
* Go to File → Preferences → Tools → Converter<br />
* Set it up as shown:<br />
[[Image:Tak_foobar_converter.png|frame|center|Screenshot of foobar 0.9.5 Converter settings for TAK 1.0.3]]<br />
'''Note:''' replace -p2 with the desired compression level.<br />
<br />
* TAK introduced encoding from STDIN in version 1.0.3, eliminating the need for a temporary file and greatly improving overall compression time. If you are using an earlier version of TAK use the following command line instead:<br />
-e -p2 %s %d<br />
* Use [[APEv2 specification|APEv2]] tagging (will be used as internal tagging)<br />
<br />
<br />
=== TAK with EAC ===<br />
Please read the [[EAC and TAK|wiki guide]], which details how to create TAK files with [[Exact Audio Copy|EAC]].<br />
<br />
<br />
== Future Features ==<br />
* Unicode support<br />
* MD5 audio checksums for verification and identification<br />
* A German version<br />
* Embedded cue sheets<br />
* Embedded cover art<br />
* Multichannel audio<br />
<br />
<br />
== Frequently Asked Questions ==<br />
; Is the codec safe for use?<br />
: Yes. To check, convert a WAVE to TAK and back and compare the two (or use foobar's bitcompare tool).<br />
; Why should I use TAK?<br />
: TAK offers high compression ratios with great decoding rates.<br />
; What can I compress with TAK?<br />
: TAK 1.0 can compress any integer-format (up to 24 bits per channel) PCM RIFF WAVE file (.wav). Piping support as of v1.0.3 is implemented, so converting lossless files to WAV first is not necessary.<br />
; What about hardware support?<br />
: None at the moment. Although, ''-p0'', ''-p1'' and ''-p2'' are the candidates for hardware playback.<br />
; When will the source be opened?<br />
: Yes, TAK will be open-source, as soon as the code is ported to C or C++ and documented. However, Thomas has mentioned that he would like to improve the codec before opening the source.<br />
<br />
<br />
== External Links ==<br />
* [http://thbeck.de/Tak/Tak.html thbeck.de/Tak/Tak.html] – Official Website ''(german)''<br />
* [http://www.hydrogenaudio.org/forums/index.php?showtopic=68454 TAK 1.1.0 Release Announcement / Discussion Thread on HA] ''(english)''<br />
* [http://www.hydrogenaudio.org/forums/index.php?showtopic=68456&st=0 TAK 1.1.0 Downloads]<br />
* [http://synthetic-soul.co.uk/comparison/lossless/ synthetic-soul.co.uk/comparison/lossless] – Comparison with Other Codecs (by Synthetic Soul)<br />
* [http://flac.sourceforge.net/comparison.html flac.sourceforge.net/comparison.html] – An Updated Comparison (from FLAC Homepage)<br />
<br />
<br />
[[Category:Lossless]]<br />
[[Category:Encoder/Decoder]]</div>Gottkaiserhttps://wiki.hydrogenaud.io/index.php?title=TAKTAK2009-01-06T16:42:39Z<p>Gottkaiser: /* Windows */</p>
<hr />
<div>{{Codec Infobox<br />
| name = Tom's lossless Audio Kompressor<br />
| logo =<br />
| type = lossless<br />
| purpose = lossless audio compression.<br />
| maintainer = Thomas Becker<br />
| recommended_encoder = TAK encoder<br />
| recommended_text = TAK v1.1.0<br />
| website = [http://thbeck.de/Tak/Tak.html ThBeck.de/Tak/Tak.html] ''(german)''<br />
}}<br />
<br />
== Description ==<br />
'''Tom's lossless Audio Kompressor''' ('''TAK''') is a lossless audio compressor which promises compression performance similar to [[Monkey's Audio]] “High” and decompression speed similar to [[Free Lossless Audio Codec|FLAC]].<br />
<br />
=== Features ===<br />
* High compression<br />
* Fast compression and decompression speed<br />
* Streaming support (necessary headers for decompressing the audio are written to the stream every 2 seconds)<br />
* Piping support for encoding<br />
* Error tolerance (single bit error will never affect more than 250 ms)<br />
* Error detection (each frame protected by a 24-bit checksum (CRC))<br />
* High-resolution (up to 24-bit/channel) audio support<br />
* Support for up to 192 Khz Audio<br />
* Seeking without seek table<br />
* APEv2 tags supported at end of file<br />
<br />
=== Pros ===<br />
* Fast encoding speed (while providing better compression TAK encodes as fast as [[Free Lossless Audio Codec|FLAC]] -8 in TAK's “Insane” and several times faster in “Turbo” mode)<br />
* Fast decompression speed (on par with FLAC / [[WavPack]])<br />
* Good compression levels (on par with [[Monkey's Audio]] High)<br />
* Error Robustness<br />
* Fast Seeking<br />
<br />
=== Cons ===<br />
* Closed Source (at the moment)<br />
* No hardware support<br />
* Very limited software support (playback: [[Winamp]] & [[foobar2000]] plugins, tagging: Mp3Tag)<br />
<br />
<br />
== Hardware and Software That Support TAK ==<br />
=== Hardware ===<br />
* None<br />
<br />
=== Software ===<br />
==== Windows ====<br />
* offical TAK Applications v1.1.0 (Applications, Winamp, SDK, Plugin, Decoding library [http://www.hydrogenaudio.org/forums/index.php?showtopic=68456&st=0 here]<br />
* foo_input_tak, TAK decoder for [[foobar2000]] [http://foosion.foobar2000.org/components/ here] (supports tagging and [[Replay Gain]])<br />
* [[Mp3tag]] – universal tag editor with support for TAK<br />
* [http://etree.org/shnutils/shntool/ shntool] (since version 3.0.6)<br />
<br />
==== Linux ====<br />
* The TAK reference applications (GUI as well as commandline) are known to run on Linux via Wine.<br />
<br />
<br />
== Recommended Settings ==<br />
* Default compression: “-p2” (formerly ''Normal'') is the most attractive setting, providing an excellent compromise between compression and encoding speed. (At compression levels close to [[Monkey's Audio]] High (<0.4% difference), it is able to encode more quickly.)<br />
takc -e [input file]<br />
* Highest compression: “-pMax” (same as -p5m) (This will create files which are comparable in size to file created using [[Monkey's Audio]] High. Decompression speed is comparable to [[WavPack]] Normal.)<br />
takc -e -pMax [input file]<br />
* Fastest compression: “-p0” (This will create files which are comparable in size to [[Monkey's Audio]] Fast or [[WavPack]] High. Decompression speed is comparable to [[Free Lossless Audio Codec|FLAC]] 0.)<br />
takc -e -p0 [input file]<br />
<br />
=== TAK Performance Graph ===<br />
[[Image:TAK_performance_graph_1-0-4.png|frame|center|Graph showing encoding and decoding rate against compression, using data from Synthetic Soul's test on TAK 1.0.4<br />(see [[TAK#External Links|External Links]])]]<br />
<br />
<br />
== Using TAK ==<br />
=== TAK with [[foobar2000]] ===<br />
* Copy the takc.exe to your [[foobar2000]] directory<br />
* Go to File → Preferences → Tools → Converter<br />
* Set it up as shown:<br />
[[Image:Tak_foobar_converter.png|frame|center|Screenshot of foobar 0.9.5 Converter settings for TAK 1.0.3]]<br />
'''Note:''' replace -p2 with the desired compression level.<br />
<br />
* TAK introduced encoding from STDIN in version 1.0.3, eliminating the need for a temporary file and greatly improving overall compression time. If you are using an earlier version of TAK use the following command line instead:<br />
-e -p2 %s %d<br />
* Use [[APEv2 specification|APEv2]] tagging (will be used as internal tagging)<br />
<br />
<br />
=== TAK with EAC ===<br />
Please read the [[EAC and TAK|wiki guide]], which details how to create TAK files with [[Exact Audio Copy|EAC]].<br />
<br />
<br />
== Future Features ==<br />
* Unicode support<br />
* MD5 audio checksums for verification and identification<br />
* A German version<br />
* Embedded cue sheets<br />
* Embedded cover art<br />
* Multichannel audio<br />
<br />
<br />
== Frequently Asked Questions ==<br />
; Is the codec safe for use?<br />
: Yes. To check, convert a WAVE to TAK and back and compare the two (or use foobar's bitcompare tool).<br />
; Why should I use TAK?<br />
: TAK offers high compression ratios with great decoding rates.<br />
; What can I compress with TAK?<br />
: TAK 1.0 can compress any integer-format (up to 24 bits per channel) PCM RIFF WAVE file (.wav). Piping support as of v1.0.3 is implemented, so converting lossless files to WAV first is not necessary.<br />
; What about hardware support?<br />
: None at the moment. Although, ''-p0'', ''-p1'' and ''-p2'' are the candidates for hardware playback.<br />
; When will the source be opened?<br />
: Yes, TAK will be open-source, as soon as the code is ported to C or C++ and documented. However, Thomas has mentioned that he would like to improve the codec before opening the source.<br />
<br />
<br />
== External Links ==<br />
* [http://thbeck.de/Tak/Tak.html thbeck.de/Tak/Tak.html] – Official Website ''(german)''<br />
* [http://www.hydrogenaudio.org/forums/index.php?showtopic=68454 TAK 1.1.0 Release Announcement / Discussion Thread on HA] ''(english)''<br />
* [http://www.hydrogenaudio.org/forums/index.php?showtopic=68456&st=0 TAK 1.1.0 Downloads]<br />
* [http://synthetic-soul.co.uk/comparison/lossless/ synthetic-soul.co.uk/comparison/lossless] – Comparison with Other Codecs (by Synthetic Soul)<br />
* [http://flac.sourceforge.net/comparison.html flac.sourceforge.net/comparison.html] – An Updated Comparison (from FLAC Homepage)<br />
<br />
<br />
[[Category:Lossless]]<br />
[[Category:Encoder/Decoder]]</div>Gottkaiserhttps://wiki.hydrogenaud.io/index.php?title=TAKTAK2009-01-06T16:40:21Z<p>Gottkaiser: </p>
<hr />
<div>{{Codec Infobox<br />
| name = Tom's lossless Audio Kompressor<br />
| logo =<br />
| type = lossless<br />
| purpose = lossless audio compression.<br />
| maintainer = Thomas Becker<br />
| recommended_encoder = TAK encoder<br />
| recommended_text = TAK v1.1.0<br />
| website = [http://thbeck.de/Tak/Tak.html ThBeck.de/Tak/Tak.html] ''(german)''<br />
}}<br />
<br />
== Description ==<br />
'''Tom's lossless Audio Kompressor''' ('''TAK''') is a lossless audio compressor which promises compression performance similar to [[Monkey's Audio]] “High” and decompression speed similar to [[Free Lossless Audio Codec|FLAC]].<br />
<br />
=== Features ===<br />
* High compression<br />
* Fast compression and decompression speed<br />
* Streaming support (necessary headers for decompressing the audio are written to the stream every 2 seconds)<br />
* Piping support for encoding<br />
* Error tolerance (single bit error will never affect more than 250 ms)<br />
* Error detection (each frame protected by a 24-bit checksum (CRC))<br />
* High-resolution (up to 24-bit/channel) audio support<br />
* Support for up to 192 Khz Audio<br />
* Seeking without seek table<br />
* APEv2 tags supported at end of file<br />
<br />
=== Pros ===<br />
* Fast encoding speed (while providing better compression TAK encodes as fast as [[Free Lossless Audio Codec|FLAC]] -8 in TAK's “Insane” and several times faster in “Turbo” mode)<br />
* Fast decompression speed (on par with FLAC / [[WavPack]])<br />
* Good compression levels (on par with [[Monkey's Audio]] High)<br />
* Error Robustness<br />
* Fast Seeking<br />
<br />
=== Cons ===<br />
* Closed Source (at the moment)<br />
* No hardware support<br />
* Very limited software support (playback: [[Winamp]] & [[foobar2000]] plugins, tagging: Mp3Tag)<br />
<br />
<br />
== Hardware and Software That Support TAK ==<br />
=== Hardware ===<br />
* None<br />
<br />
=== Software ===<br />
==== Windows ====<br />
* offical TAK Applications v1.1.0 (Applications, Winamp, SDK, Plugin, Decoding library [http://www.hydrogenaudio.org/forums/index.php?showtopic=68456&st=0 here]<br />
* foo_input_tak, TAK decoder for [[foobar2000]] [http://foosion.foobar2000.org/components/ here] (supports tagging and [[Replay Gain]])<br />
* [[Mp3tag]] – universal tag editor with support for TAK<br />
* shntool (since version 3.0.6)<br />
<br />
==== Linux ====<br />
* The TAK reference applications (GUI as well as commandline) are known to run on Linux via Wine.<br />
<br />
<br />
== Recommended Settings ==<br />
* Default compression: “-p2” (formerly ''Normal'') is the most attractive setting, providing an excellent compromise between compression and encoding speed. (At compression levels close to [[Monkey's Audio]] High (<0.4% difference), it is able to encode more quickly.)<br />
takc -e [input file]<br />
* Highest compression: “-pMax” (same as -p5m) (This will create files which are comparable in size to file created using [[Monkey's Audio]] High. Decompression speed is comparable to [[WavPack]] Normal.)<br />
takc -e -pMax [input file]<br />
* Fastest compression: “-p0” (This will create files which are comparable in size to [[Monkey's Audio]] Fast or [[WavPack]] High. Decompression speed is comparable to [[Free Lossless Audio Codec|FLAC]] 0.)<br />
takc -e -p0 [input file]<br />
<br />
=== TAK Performance Graph ===<br />
[[Image:TAK_performance_graph_1-0-4.png|frame|center|Graph showing encoding and decoding rate against compression, using data from Synthetic Soul's test on TAK 1.0.4<br />(see [[TAK#External Links|External Links]])]]<br />
<br />
<br />
== Using TAK ==<br />
=== TAK with [[foobar2000]] ===<br />
* Copy the takc.exe to your [[foobar2000]] directory<br />
* Go to File → Preferences → Tools → Converter<br />
* Set it up as shown:<br />
[[Image:Tak_foobar_converter.png|frame|center|Screenshot of foobar 0.9.5 Converter settings for TAK 1.0.3]]<br />
'''Note:''' replace -p2 with the desired compression level.<br />
<br />
* TAK introduced encoding from STDIN in version 1.0.3, eliminating the need for a temporary file and greatly improving overall compression time. If you are using an earlier version of TAK use the following command line instead:<br />
-e -p2 %s %d<br />
* Use [[APEv2 specification|APEv2]] tagging (will be used as internal tagging)<br />
<br />
<br />
=== TAK with EAC ===<br />
Please read the [[EAC and TAK|wiki guide]], which details how to create TAK files with [[Exact Audio Copy|EAC]].<br />
<br />
<br />
== Future Features ==<br />
* Unicode support<br />
* MD5 audio checksums for verification and identification<br />
* A German version<br />
* Embedded cue sheets<br />
* Embedded cover art<br />
* Multichannel audio<br />
<br />
<br />
== Frequently Asked Questions ==<br />
; Is the codec safe for use?<br />
: Yes. To check, convert a WAVE to TAK and back and compare the two (or use foobar's bitcompare tool).<br />
; Why should I use TAK?<br />
: TAK offers high compression ratios with great decoding rates.<br />
; What can I compress with TAK?<br />
: TAK 1.0 can compress any integer-format (up to 24 bits per channel) PCM RIFF WAVE file (.wav). Piping support as of v1.0.3 is implemented, so converting lossless files to WAV first is not necessary.<br />
; What about hardware support?<br />
: None at the moment. Although, ''-p0'', ''-p1'' and ''-p2'' are the candidates for hardware playback.<br />
; When will the source be opened?<br />
: Yes, TAK will be open-source, as soon as the code is ported to C or C++ and documented. However, Thomas has mentioned that he would like to improve the codec before opening the source.<br />
<br />
<br />
== External Links ==<br />
* [http://thbeck.de/Tak/Tak.html thbeck.de/Tak/Tak.html] – Official Website ''(german)''<br />
* [http://www.hydrogenaudio.org/forums/index.php?showtopic=68454 TAK 1.1.0 Release Announcement / Discussion Thread on HA] ''(english)''<br />
* [http://www.hydrogenaudio.org/forums/index.php?showtopic=68456&st=0 TAK 1.1.0 Downloads]<br />
* [http://synthetic-soul.co.uk/comparison/lossless/ synthetic-soul.co.uk/comparison/lossless] – Comparison with Other Codecs (by Synthetic Soul)<br />
* [http://flac.sourceforge.net/comparison.html flac.sourceforge.net/comparison.html] – An Updated Comparison (from FLAC Homepage)<br />
<br />
<br />
[[Category:Lossless]]<br />
[[Category:Encoder/Decoder]]</div>Gottkaiserhttps://wiki.hydrogenaud.io/index.php?title=BonkEncBonkEnc2007-12-29T17:36:46Z<p>Gottkaiser: </p>
<hr />
<div>{{Software Infobox<br />
| name = BonkEnc<br />
| screenshot = [[Image:BonkEnc-screenshot.png|250px|BonkEnc screenshot]]<br />
| caption = Slim CD ripper, audio encoder and converter <br />
| maintainer = Robert Kausch<br />
| stable_release = 1.0.5 (04 of November 2007)<br />
| preview_release = <br />
| operating_system = Windows<br />
| use = Digital Audio Extraction<br />
| license = GPL <br />
| website = [http://www.bonkenc.org http://www.bonkenc.org]<br />
}}<br />
<br />
=Introduction=<br />
'''BonkEnc''' is a CD ripper, audio encoder and converter for various formats. It's a very slim program and can be run from a USB stick. BonkEnc is available under GPL liscense and has native support for numerous languages. <br />
<br />
== Features ==<br />
* CD ripping<br />
** [[Cdparanoia]] mode<br />
** jitter correction<br />
** CD Text support<br />
* [[Transcoding]] from on to another format<br />
* using Compact Disc Database (CDDB)<br />
* support [[ID3v1]], [[ID3v2]], MP4-Metadata and [[Vorbis_comment|Vorbis comment]] [[Tags]]<br />
* keeps image tags when converting from FLAC to MP3<br />
* creating cue sheets and playlists<br />
* full UTF-8 Unicode support<br />
* additional command line interface (CD Ripping/Encoding)<br />
<br />
== Supported Formats ==<br />
* [[MP3]] ([[LAME]])<br />
* [[AAC]] ([[FAAC]])<br />
* [[MP4]]/[[M4A]]<br />
* [[Ogg_Vorbis|Ogg Vorbis]]<br />
* [[FLAC]]<br />
* [http://www.logarithmic.net/pfh/bonk Bonk] v0.11<br />
<br />
== Recommended Settings ==<br />
{| border="0" valign="top"<br />
|<br />
* [[LAME]] MP3 Encoder v3.97:<br />
* [[FAAC]] MP4/AAC Encoder v1.26:<br />
* Ogg Vorbis Encoder ([[aoTuV|aoTuV beta 5]]):<br />
* FLAC Audio Encoder v1.2.1:<br />
||<br />
[[LAME#Recommended_encoder_settings|here]]<br/><br />
to add<br/><br />
[[Recommended_Ogg_Vorbis#Recommended_Encoder_Settings|here]]<br/><br />
Use preset 5<br />
|}<br />
<br />
== Supported languages ==<br />
{| border="0" -valign="top"<br />
|-valign="top"<br />
||<br />
* Catalan<br />
* Chinese<br />
* Czech<br />
* Danish<br />
* Dutch<br />
* English<br />
* Esperanto<br />
* Finnish<br />
||<br />
* French<br />
* German<br />
* Greek<br />
* Hungarian<br />
* Italian<br />
* Japanese<br />
* Korean<br />
* Lithuanian<br />
||<br />
* Polish<br />
* Portuguese<br />
* Russian<br />
* Serbian<br />
* Slovak<br />
* Spanish<br />
* Swedish<br />
* Turkish<br />
||<br />
* Ukrainian<br />
* Romanian<br />
|}<br />
<br />
<br />
<br />
== External links==<br />
* [http://www.bonkenc.org/ BonkEnc: Homepage]<br />
* [http://www.bonkenc.org/index.php?option=com_content&task=blogcategory&id=3&Itemid=33 BonkEnc: Download]<br />
* [http://sourceforge.net/forum/?group_id=27149 BonkEnc: official forum] official support forum <br />
<br />
[[Category:Software]]</div>Gottkaiserhttps://wiki.hydrogenaud.io/index.php?title=Mp3tagMp3tag2007-08-21T19:11:52Z<p>Gottkaiser: version update</p>
<hr />
<div>{{Software Infobox<br />
| name = MP3tag<br />
| screenshot = [[Image:MP3tag-screenshot.png|250px|MP3tag screenshot]]<br />
| caption = powerful and easy-to-use tool to edit metadata <br />
| maintainer = Florian Heidenreich<br />
| stable_release = [http://www.mp3tag.de/en/download.html 2.39] (28.07.2007)<br />
| preview_release = [http://developer.mp3tag.de/ 2.39a] (19.08.2007)<br />
| operating_system = Windows<br />
| use = Metadata<br />
| license = Freeware <br />
| website = [http://www.mp3tag.de/en/ http://www.mp3tag.de]<br />
}}<br />
<br />
=Introduction=<br />
'''MP3tag''' is an powerful [[Metadata|metadata]] editor for lots of common audio formats. You can rename files based on the [[Tagging|tag]] information, replace characters or words from tags and filenames, import/export tag information, create playlists.<br />
<br />
The program supports online freedb database lookups for selected files, allowing you to automatically gather proper tag information for select files or CDs.<br />
<br />
==Features==<br />
<br />
* Write [[ID3v1.1]]-, [[ID3v2]]-, [[APEv2]]-Tags and [[Vorbis_Comment|Vorbis Comments]] to multiple files at once<br />
* Full Unicode support<br />
* Support for embedded cover art<br />
* Automatically create playlists<br />
* Recursive subfolders support<br />
* Remove parts or the entire tag of multiple files<br />
* Rename files based on the tag information<br />
* Import tags from filenames<br />
* Format tags and filenames<br />
* Replace characters or words from tags and filenames<br />
* Regular Expressions<br />
* Export tag information to user-defined formats (like html, rtf, csv, xml)<br />
* Import tag information from online databases like freedb or Amazon (also by text-search)<br />
* Import tag information from local freedb databases<br />
* Support for [[ID3v2|ID3v2.3]] (ISO-8859-1 and UTF-16) and [[ID3v2|ID3v2.4]] with [[UTF-8]]<br />
<br />
==Supported formats==<br />
<br />
* Advanced Audio Coding ([[AAC|aac]])<br />
* Free Lossless Audio Codec ([[FLAC|flac]])<br />
* Monkey's Audio ([[APE|ape]])<br />
* Mpeg Layer 3 ([[MP3|mp3]])<br />
* MPEG-4 ([[MP4|mp4]] / [[M4A|m4a]] / m4b / [[iTunes]] compatible)<br />
* Musepack ([[MPC|mpc]])<br />
* Ogg Vorbis ([[Ogg_Vorbis|ogg]])<br />
* OptimFROG ([[OptimFROG|ofr]])<br />
* OptimFROG DualStream (ofs)<br />
* Speex ([[Speex|spx]])<br />
* TAK ([[TAK]])<br />
* True Audio (tta)<br />
* Windows Media Audio ([[WMA|wma]])<br />
* WavPack ([[WavPack|wv]])<br />
<br />
==Operating Systems==<br />
* Win 2000<br />
* Win XP<br />
* Win 2003<br />
* Win Vista<br />
<br />
==External links==<br />
* [http://www.mp3tag.de/en/ Mp3tag: Homepage]<br />
* [http://www.mp3tag.de/en/download.html Mp3tag: Download]<br />
* [http://www.anytag.de/forums/ Mp3tag: official forum]<br />
* [http://www.mp3tag.de/download/mp3tagversion.en.html Mp3tag: changelog]<br />
* [http://wiki.slimdevices.com/index.cgi?Mp3tagGuide Mp3tagGuide - Slim Devices]<br />
* [http://www.anytag.de/forums/index.php?showtopic=1794 Mp3tag: additional Web Sources]<br />
<br />
[[Category:Software]]</div>Gottkaiserhttps://wiki.hydrogenaud.io/index.php?title=Replay_GainReplay Gain2007-08-20T11:38:16Z<p>Gottkaiser: </p>
<hr />
<div>'''Replay Gain''' is the name of a technique invented to achieve the same playback volume of audio files. It specifies the reference level of 89dB and an algorithm to measure the '''perceived''' loudness of audio data.<br />
<br />
Replay Gain is different from [[Normalization|peak normalization]]. In peak normalization, you merely ensure that the peak amplitude reaches a certain level. This does not ensure equal loudness. The Replay Gain technique measures the ''effective power'' (i.e. taking RMS after an ''Equal Loudness contour'') of the waveform, and amplifies the waveform accordingly. The result is that Replay Gained waveforms are usually more uniformly amplified that peak-normalized waveforms.<br />
<br />
== Implementations ==<br />
There are different Replay Gain implementations, each with its own uses and strength. Most of them use [[metadata]] to indicate the level of the volume change (Volume is adjusted on playback; needs player/decoder support) others modify the audio data itself. Generally it is recommended to use an implementation which uses the metadata and does not touch the audio itself.<br />
<br />
In a metadata based solution, information on both types of Replay Gain can be stored, and the desired playback effect can be switched back and forth in the appropriate player. However, if the audio data is permanently modified, only one type of Replay Gain can be chosen. Furthermore, if the audio data is modified, you may not be able to restore the original data, especially if the Replay Gain technique produces a negative amplification.<br />
<br />
Replay Gain has significant advantages over normalizing. It allows the loudness of a song to be consistant over an entire collection of audio, similar to (but more accurate than) RMS normalizing (this is called 'Track Gain', or 'Radio Gain' in earlier parlance). However, it also allows the loudness of an album to be consistant over a entire music collection, allowing the dynamics of album to remain (This is called 'Album Gain', or 'Audiophile Gain' in earlier parlance). This is usually very important in classical CDs, as there may be quiet segments and loud segments written into different tracks.<br />
<br />
=== MP3Gain ===<br />
[[MP3Gain]] is an implementation of Replay Gain. It either modifies target data reversibly, or attaches metadata.<br />
<br />
* Format: [[MP3]]<br />
* Method: Audio / Meta<br />
* Limitations: Limited to 1.5dB steps mode, may become irreversible<br />
* http://mp3gain.sourceforge.net/<br />
<br />
=== [[LAME]] ===<br />
* Method: Header ([http://gabriel.mp3-tech.org/mp3infotag.html mp3infotag])<br />
* Notes:<br />
** Tags added during encoding; not supported by any player yet; Track Gain only<br />
** Replay Gaining MP3's are usually done using MP3Gain (see [[Replay Gain#MP3Gain|above]]) or [[Replay Gain#foobar2000 Replay Gain scanner|foobar2000]]<br />
* http://lame.sourceforge.net/<br />
<br />
=== [[Musepack]] Replay Gain ===<br />
* Method: Header (similar to Meta data method)<br />
* Notes: Replay Gain values are stored in the header and Replay Gain is part of the Musepack specifications; therefore any Musepack decoder that does not support Replay Gain can be considered broken.<br />
* http://rarewares.org/mpc.html<br />
<br />
=== VorbisGain ===<br />
* Format: (Ogg) [[Vorbis]]<br />
* Method: Meta (in [[Vorbis comment]])<br />
* http://www.sjeng.org/vorbisgain.html<br />
** new compiles of VorbisGain at [http://www.rarewares.org/ogg.html www.rarewares.org]<br />
:'''''Note:''' Andavari has provided a very useful script to integrate VorbisGain, which is a CLI tool, into Windows Explorer. Please (Ogg) [[Vorbis#Replay Gain|check this section]].<br />
<br />
=== FLAC / METAFLAC ===<br />
* Format: [[Free Lossless Audio Codec|FLAC]]<br />
* Method: Meta (in [[Vorbis comment]])<br />
* http://flac.sf.net<br />
<br />
=== WavPack / WVGAIN ===<br />
* Format: [[WavPack]]<br />
* Method: Meta (in [[APEv2]] tag)<br />
* http://www.wavpack.com<br />
<br />
=== Wavegain ===<br />
* Format: waveform<br />
* Method: Audio<br />
* Limitations: Irreversible<br />
* http://www.rarewares.org/files/others/wavegain.zip<br />
<br />
=== [[foobar2000]] Replay Gain scanner ===<br />
* Format:<br />
** [[MP3]]: Values written to [[APEv2]] or [[ID3v2]] tags.<br />
** [[Musepack]]: Values written to header.<br />
** (Ogg) [[Vorbis]]: Values written to [[Vorbis comment]].<br />
** [[WavPack]]: Values written to [[APEv2]] tags.<br />
** [[AAC]]: Values written to [[APEv2]] tags.<br />
** [[MP4]]: Uses its own iTunes-compatible tagging system (though iTunes does not support Replay Gain).<br />
** [[Free Lossless Audio Codec|FLAC]]: Values written to [[Vorbis comment]].<br />
** [[APE]]: Values written to [[APEv2]] tags.<br />
** Modules ([[MOD]] etc.): Optionally saved into [[APEv2]] tags; otherwise the foobar2000 database is used.<br />
** All other formats are supported but the Replay Gain values are saved to the foobar2000 database.<br />
<br />
* You can also choose to only have the Replay Gain values saved in the foobar2000 database and leave the files untouched.<br />
<br />
* http://foobar2000.org<br />
<br />
=== [[MediaMonkey]] ===<br />
* Format:<br />
** [[MP3]]: Values written to [[APEv2]] or [[ID3v2]] tags.<br />
** (Ogg) [[Vorbis]]: Values written to [[Vorbis comment]].<br />
** [[WMA]]: Values stored in MediaMonkey's MDB database.<br />
** [[Free Lossless Audio Codec|FLAC]]: Values written to [[Vorbis comment]].<br />
** [[APE]]: Values written to [[APEv2]] tags.<br />
** [[WAV]]: Values stored in MediaMonkey's MDB database.<br />
* In addition to tags, all Replay Gain values are also stored in MediaMonkey's MDB database<br />
* Does not support Album/Audiophile Replay Gain<br />
* Also capable of (irreversibly) changing the volume of MP3 tracks, similar to [[MP3Gain]]<br />
* http://www.mediamonkey.com<br />
<br />
=== [[Winamp]] Replay Gain scanner===<br />
* Format:<br />
** [[MP3]]: Values written to [[ID3v2]] tags.<br />
** (Ogg) [[Vorbis]]: Values written to [[Vorbis comment]].<br />
** [[WMA]]: Values stored in Windows Media Audio tags.<br />
** [[Free Lossless Audio Codec|FLAC]]: Values written to [[Vorbis comment]].<br />
** [[APE]]: Values written to [[APEv2]] tags.<br />
** [[AAC]]: Values written to [[APEv2]] tags.<br />
** [[MP4]]<br />
** [[TAK]]: Values written to [[APEv2]] tags.<br />
* Support Album/Track Gain<br />
<br />
== Players support ==<br />
Replay Gain being present in the specs of FLAC, Musepack, and APE formats, any player that support those formats usually support Replay Gain.<br />
<br />
The situation with MP3 is rather different, as it was not part of the MP3 specs. The APEv2 tags metadata implementation is somewhat becoming the de-facto standard.<br />
<br />
=== Windows ===<br />
* [[Foobar2000]] supports Replay Gain in all possible aspects.<br />
* [[Winamp]] supports Replay Gain in album or track mode.<br />
* [[MediaMonkey]] supports track Replay Gain only<br />
* [[XMPlay]] recently implemented Replay Gain<br />
<br />
''...and probably others.''<br />
<br />
=== Linux ===<br />
* [[XMMS]]. Reads Replay Gain from [[Free Lossless Audio Codec|FLAC]], [[Musepack]], (Ogg) [[Vorbis]] ..<br />
:For [[MP3]], use the CVS version of the [http://xmms-mad.sourceforge.net/ xmms-mad] mp3 plugin (it's not yet released as binary, furthermore not available in distribs' versions for now. Meanwhile binaries are available here: [http://perso.crans.org/~krempp/xmms-mad/ custom binaries])<br />
* [[amarok]]. By using the amarok-script [http://kde-apps.org/content/show.php?content=26073 Replay Gain]<br />
:And possibly others, since [http://developer.kde.org/~wheeler/taglib.html TagLib] added support for [[APEv2]] tags in [[MP3]] files, players using this library (like [[amaroK]] and [[JuK]]) might support that kind of Replay Gain tags in the near future.<br />
* [http://www.sacredchao.net/quodlibet Quod Libet] reads Replay Gain from (Ogg) [[Vorbis]], [[MP3]], [[Free Lossless Audio Codec|FLAC]], and [[Musepack]].<br />
:Requires support to be enabled (via the appropriate python bindings and libraries) for the above formats. Does not support Replay Gain values stored in [[APEv2]] tags in [[MP3]]s. Replay Gain values are stored in RVA2 id3v2.4 frames. See the [http://www.sacredchao.net/quodlibet/wiki/Development/ID3Notes Quod Libet RVA2 / Replay Gain notes].<br />
* [http://www.musicpd.org/ Music Player Daemon] (MPD) reads Replay Gain from (Ogg) [[Vorbis]], [[Free Lossless Audio Codec|FLAC]], and [[Musepack]].<br />
:Foobar2000 style TXXX frames in [[MP3]]s are also supported in the latest development releases.<br />
<br />
=== Portable devices ===<br />
Current development builds of [http://www.rockbox.org/ Rockbox] support Replay Gain for some encoder formats. This is a rapidly evolving feature. Rockbox runs on a variety of portable players, including iRiver H100, H300 and H10 series; iPod 3rd gen, 4th gen (grayscale and color), 5th/5.5th gen video, 1st gen Nano and Mini 1st/2nd gen (Nano 2nd gen is not supported); Cowon iAudio X5 (including X5V and X5L)and M5 (including M5L); Toshiba Gigabeat X and F series (the S model is not supported); and SanDisk: Sansa E200 series (the R models are not supported).<br />
<br />
There are no other portable players known to support Replay Gain.<br />
<br />
The iPod features ''Soundcheck'', which seems to produce roughly the same normalization gains as Replay Gain, but doesn't provide an Album Gain.<br />
<br />
=== Hi-Fi ===<br />
Slim Devices a company owened by Logitech Inc, supports Replay Gain on both of their hi-end audiophile players, known as the [[Slim Devices Transporter|Transporter]] and the [[Slim Devices Squeezebox|Squeezebox]].<br />
<br />
== External links ==<br />
* [http://replaygain.hydrogenaudio.org Original Replay Gain website]<br />
* [http://en.wikipedia.org/wiki/Replay_Gain Replay Gain] at Wikipedia<br />
* [http://www.bobulous.org.uk/misc/Replay-Gain.html Replay Gain using Foobar 2000] (how-to).<br />
<br />
<br />
[[Category:Technical]]<br />
[[Category:Metadata]]</div>Gottkaiserhttps://wiki.hydrogenaud.io/index.php?title=Mp3tagMp3tag2007-07-29T12:23:55Z<p>Gottkaiser: version update</p>
<hr />
<div>{{Software Infobox<br />
| name = MP3tag<br />
| screenshot = [[Image:MP3tag-screenshot.png|250px|MP3tag screenshot]]<br />
| caption = powerful and easy-to-use tool to edit metadata <br />
| maintainer = Florian Heidenreich<br />
| stable_release = [http://www.mp3tag.de/en/download.html 2.39] (28.07.2007)<br />
| preview_release =<br />
| operating_system = Windows<br />
| use = Metadata<br />
| license = Freeware <br />
| website = [http://www.mp3tag.de/en/ http://www.mp3tag.de]<br />
}}<br />
<br />
=Introduction=<br />
'''MP3tag''' is an powerful [[Metadata|metadata]] editor for lots of common audio formats. You can rename files based on the [[Tagging|tag]] information, replace characters or words from tags and filenames, import/export tag information, create playlists.<br />
<br />
The program supports online freedb database lookups for selected files, allowing you to automatically gather proper tag information for select files or CDs.<br />
<br />
==Features==<br />
<br />
* Write [[ID3v1.1]]-, [[ID3v2]]-, [[APEv2]]-Tags and [[Vorbis_Comment|Vorbis Comments]] to multiple files at once<br />
* Full Unicode support<br />
* Support for embedded cover art<br />
* Automatically create playlists<br />
* Recursive subfolders support<br />
* Remove parts or the entire tag of multiple files<br />
* Rename files based on the tag information<br />
* Import tags from filenames<br />
* Format tags and filenames<br />
* Replace characters or words from tags and filenames<br />
* Regular Expressions<br />
* Export tag information to user-defined formats (like html, rtf, csv, xml)<br />
* Import tag information from online databases like freedb or Amazon (also by text-search)<br />
* Import tag information from local freedb databases<br />
* Support for [[ID3v2|ID3v2.3]] (ISO-8859-1 and UTF-16) and [[ID3v2|ID3v2.4]] with [[UTF-8]]<br />
<br />
==Supported formats==<br />
<br />
* Advanced Audio Coding ([[AAC|aac]])<br />
* Free Lossless Audio Codec ([[FLAC|flac]])<br />
* Monkey's Audio ([[APE|ape]])<br />
* Mpeg Layer 3 ([[MP3|mp3]])<br />
* MPEG-4 ([[MP4|mp4]] / [[M4A|m4a]] / m4b / [[iTunes]] compatible)<br />
* Musepack ([[MPC|mpc]])<br />
* Ogg Vorbis ([[Ogg_Vorbis|ogg]])<br />
* OptimFROG ([[OptimFROG|ofr]])<br />
* OptimFROG DualStream (ofs)<br />
* Speex ([[Speex|spx]])<br />
* TAK ([[TAK]])<br />
* True Audio (tta)<br />
* Windows Media Audio ([[WMA|wma]])<br />
* WavPack ([[WavPack|wv]])<br />
<br />
==Operating Systems==<br />
* Win 2000<br />
* Win XP<br />
* Win 2003<br />
* Win Vista<br />
<br />
==External links==<br />
* [http://www.mp3tag.de/en/ Mp3tag: Homepage]<br />
* [http://www.mp3tag.de/en/download.html Mp3tag: Download]<br />
* [http://www.anytag.de/forums/ Mp3tag: official forum]<br />
* [http://www.mp3tag.de/download/mp3tagversion.en.html Mp3tag: changelog]<br />
* [http://wiki.slimdevices.com/index.cgi?Mp3tagGuide Mp3tagGuide - Slim Devices]<br />
* [http://www.anytag.de/forums/index.php?showtopic=1794 Mp3tag: additional Web Sources]<br />
<br />
[[Category:Software]]</div>Gottkaiserhttps://wiki.hydrogenaud.io/index.php?title=Mp3tagMp3tag2007-07-19T14:18:45Z<p>Gottkaiser: version update</p>
<hr />
<div>{{Software Infobox<br />
| name = MP3tag<br />
| screenshot = [[Image:MP3tag-screenshot.png|250px|MP3tag screenshot]]<br />
| caption = powerful and easy-to-use tool to edit metadata <br />
| maintainer = Florian Heidenreich<br />
| stable_release = [http://www.mp3tag.de/en/download.html 2.38] (29.04.2007)<br />
| preview_release = [http://developer.mp3tag.de/ 2.38c] (18.07.2007)<br />
| operating_system = Windows<br />
| use = Metadata<br />
| license = Freeware <br />
| website = [http://www.mp3tag.de/en/ http://www.mp3tag.de]<br />
}}<br />
<br />
=Introduction=<br />
'''MP3tag''' is an powerful [[Metadata|metadata]] editor for lots of common audio formats. You can rename files based on the [[Tagging|tag]] information, replace characters or words from tags and filenames, import/export tag information, create playlists.<br />
<br />
The program supports online freedb database lookups for selected files, allowing you to automatically gather proper tag information for select files or CDs.<br />
<br />
==Features==<br />
<br />
* Write [[ID3v1.1]]-, [[ID3v2]]-, [[APEv2]]-Tags and [[Vorbis_Comment|Vorbis Comments]] to multiple files at once<br />
* Full Unicode support<br />
* Support for embedded cover art<br />
* Automatically create playlists<br />
* Recursive subfolders support<br />
* Remove parts or the entire tag of multiple files<br />
* Rename files based on the tag information<br />
* Import tags from filenames<br />
* Format tags and filenames<br />
* Replace characters or words from tags and filenames<br />
* Regular Expressions<br />
* Export tag information to user-defined formats (like html, rtf, csv, xml)<br />
* Import tag information from online databases like freedb or Amazon (also by text-search)<br />
* Import tag information from local freedb databases<br />
* Support for [[ID3v2|ID3v2.3]] (ISO-8859-1 and UTF-16) and [[ID3v2|ID3v2.4]] with [[UTF-8]]<br />
<br />
==Supported formats==<br />
<br />
* Advanced Audio Coding ([[AAC|aac]])<br />
* Free Lossless Audio Codec ([[FLAC|flac]])<br />
* Monkey's Audio ([[APE|ape]])<br />
* Mpeg Layer 3 ([[MP3|mp3]])<br />
* MPEG-4 ([[MP4|mp4]] / [[M4A|m4a]] / m4b / [[iTunes]] compatible)<br />
* Musepack ([[MPC|mpc]])<br />
* Ogg Vorbis ([[Ogg_Vorbis|ogg]])<br />
* OptimFROG ([[OptimFROG|ofr]])<br />
* OptimFROG DualStream (ofs)<br />
* Speex ([[Speex|spx]])<br />
* TAK ([[TAK]])<br />
* True Audio (tta)<br />
* Windows Media Audio ([[WMA|wma]])<br />
* WavPack ([[WavPack|wv]])<br />
<br />
==Operating Systems==<br />
* Win 2000<br />
* Win XP<br />
* Win 2003<br />
* Win Vista<br />
<br />
==External links==<br />
* [http://www.mp3tag.de/en/ Mp3tag: Homepage]<br />
* [http://www.mp3tag.de/en/download.html Mp3tag: Download]<br />
* [http://www.anytag.de/forums/ Mp3tag: official forum]<br />
* [http://www.mp3tag.de/download/mp3tagversion.en.html Mp3tag: changelog]<br />
* [http://wiki.slimdevices.com/index.cgi?Mp3tagGuide Mp3tagGuide - Slim Devices]<br />
* [http://www.anytag.de/forums/index.php?showtopic=1794 Mp3tag: additional Web Sources]<br />
<br />
[[Category:Software]]</div>Gottkaiserhttps://wiki.hydrogenaud.io/index.php?title=Mp3tagMp3tag2007-07-15T21:56:25Z<p>Gottkaiser: version update</p>
<hr />
<div>{{Software Infobox<br />
| name = MP3tag<br />
| screenshot = [[Image:MP3tag-screenshot.png|250px|MP3tag screenshot]]<br />
| caption = powerful and easy-to-use tool to edit metadata <br />
| maintainer = Florian Heidenreich<br />
| stable_release = [http://www.mp3tag.de/en/download.html 2.38] (29.04.2007)<br />
| preview_release = [http://developer.mp3tag.de/ 2.38b] (15.07.2007)<br />
| operating_system = Windows<br />
| use = Metadata<br />
| license = Freeware <br />
| website = [http://www.mp3tag.de/en/ http://www.mp3tag.de]<br />
}}<br />
<br />
=Introduction=<br />
'''MP3tag''' is an powerful [[Metadata|metadata]] editor for lot's of common audio formats. You can rename files based on the [[Tagging|tag]] information, replace characters or words from tags and filenames, import/export tag information, create playlists.<br />
<br />
The program supports online freedb database lookups for selected files, allowing you to automatically gather proper tag information for select files or CDs.<br />
<br />
==Features==<br />
<br />
* Write [[ID3v1.1]]-, [[ID3v2]]-, [[APEv2]]-Tags and [[Vorbis_Comment|Vorbis Comments]] to multiple files at once<br />
* Full Unicode support<br />
* Support for embedded cover art<br />
* Automatically create playlists<br />
* Recursive subfolders support<br />
* Remove parts or the entire tag of multiple files<br />
* Rename files based on the tag information<br />
* Import tags from filenames<br />
* Format tags and filenames<br />
* Replace characters or words from tags and filenames<br />
* Regular Expressions<br />
* Export tag information to user-defined formats (like html, rtf, csv, xml)<br />
* Import tag information from online databases like freedb or Amazon (also by text-search)<br />
* Import tag information from local freedb databases<br />
* Support for [[ID3v2|ID3v2.3]] (ISO-8859-1 and UTF-16) and [[ID3v2|ID3v2.4]] with [[UTF-8]]<br />
<br />
==Supported formats==<br />
<br />
* Advanced Audio Coding ([[AAC|aac]])<br />
* Free Lossless Audio Codec ([[FLAC|flac]])<br />
* Monkey's Audio ([[APE|ape]])<br />
* Mpeg Layer 3 ([[MP3|mp3]])<br />
* MPEG-4 ([[MP4|mp4]] / [[M4A|m4a]] / m4b / [[iTunes]] compatible)<br />
* Musepack ([[MPC|mpc]])<br />
* Ogg Vorbis ([[Ogg_Vorbis|ogg]])<br />
* OptimFROG ([[OptimFROG|ofr]])<br />
* OptimFROG DualStream (ofs)<br />
* Speex ([[Speex|spx]])<br />
* TAK ([[TAK]])<br />
* True Audio (tta)<br />
* Windows Media Audio ([[WMA|wma]])<br />
* WavPack ([[WavPack|wv]])<br />
<br />
==Operating Systems==<br />
* Win 2000<br />
* Win XP<br />
* Win 2003<br />
* Win Vista<br />
<br />
==External links==<br />
* [http://www.mp3tag.de/en/ Mp3tag: Homepage]<br />
* [http://www.mp3tag.de/en/download.html Mp3tag: Download]<br />
* [http://www.anytag.de/forums/ Mp3tag: official forum]<br />
* [http://www.mp3tag.de/download/mp3tagversion.en.html Mp3tag: changelog]<br />
* [http://wiki.slimdevices.com/index.cgi?Mp3tagGuide Mp3tagGuide - Slim Devices]<br />
* [http://www.anytag.de/forums/index.php?showtopic=1794 Mp3tag: additional Web Sources]<br />
<br />
[[Category:Software]]</div>Gottkaiserhttps://wiki.hydrogenaud.io/index.php?title=BonkEncBonkEnc2007-07-08T19:01:30Z<p>Gottkaiser: version update</p>
<hr />
<div>{{Software Infobox<br />
| name = BonkEnc<br />
| screenshot = [[Image:BonkEnc-screenshot.png|250px|BonkEnc screenshot]]<br />
| caption = Slim CD ripper, audio encoder and converter <br />
| maintainer = Robert Kausch<br />
| stable_release = 1.0.4 (08 of July 2007)<br />
| preview_release = <br />
| operating_system = Windows<br />
| use = Digital Audio Extraction<br />
| license = GPL <br />
| website = [http://www.bonkenc.org http://www.bonkenc.org]<br />
}}<br />
<br />
=Introduction=<br />
'''BonkEnc''' is a CD ripper, audio encoder and converter for various formats. It's a very slim program and can be run from a USB stick. BonkEnc is available under GPL liscense and has native support for numerous languages. <br />
<br />
== Features ==<br />
* CD ripping<br />
** [[Cdparanoia]] mode<br />
** jitter correction<br />
** CD Text support<br />
* [[Transcoding]] from on to another format<br />
* using Compact Disc Database (CDDB)<br />
* support [[ID3v1]], [[ID3v2]], MP4-Metadata and [[Vorbis_comment|Vorbis comment]] [[Tags]]<br />
* keeps image tags when converting from FLAC to MP3<br />
* creating cue sheets and playlists<br />
* full UTF-8 Unicode support<br />
* additional command line interface (CD Ripping/Encoding)<br />
<br />
== Supported Formats ==<br />
* [[MP3]] ([[LAME]])<br />
* [[AAC]] ([[FAAC]])<br />
* [[MP4]]/[[M4A]]<br />
* [[Ogg_Vorbis|Ogg Vorbis]]<br />
* [[FLAC]]<br />
* [http://www.logarithmic.net/pfh/bonk Bonk] v0.11<br />
<br />
== Recommended Settings ==<br />
{| border="0" valign="top"<br />
|<br />
* [[LAME]] MP3 Encoder v3.97:<br />
* [[FAAC]] MP4/AAC Encoder v1.25:<br />
* Ogg Vorbis Encoder ([[aoTuV|aoTuV beta 5]]):<br />
* FLAC Audio Encoder v1.1.4:<br />
||<br />
[[LAME#Recommended_encoder_settings|here]]<br/><br />
to add<br/><br />
[[Recommended_Ogg_Vorbis#Recommended_Encoder_Settings|here]]<br/><br />
Use preset 5<br />
|}<br />
<br />
== Supported languages ==<br />
{| border="0" -valign="top"<br />
|-valign="top"<br />
||<br />
* Catalan<br />
* Chinese<br />
* Czech<br />
* Danish<br />
* Dutch<br />
* English<br />
* Esperanto<br />
* Finnish<br />
||<br />
* French<br />
* German<br />
* Greek<br />
* Hungarian<br />
* Italian<br />
* Japanese<br />
* Korean<br />
* Lithuanian<br />
||<br />
* Polish<br />
* Portuguese<br />
* Russian<br />
* Serbian<br />
* Slovak<br />
* Spanish<br />
* Swedish<br />
* Turkish<br />
||<br />
* Ukrainian<br />
* Romanian<br />
|}<br />
<br />
<br />
<br />
== External links==<br />
* [http://www.bonkenc.org/ BonkEnc: Homepage]<br />
* [http://www.bonkenc.org/index.php?option=com_content&task=blogcategory&id=3&Itemid=33 BonkEnc: Download]<br />
* [http://sourceforge.net/forum/?group_id=27149 BonkEnc: official forum] official support forum <br />
<br />
[[Category:Software]]</div>Gottkaiserhttps://wiki.hydrogenaud.io/index.php?title=Mp3tagMp3tag2007-06-22T09:22:09Z<p>Gottkaiser: version update</p>
<hr />
<div>{{Software Infobox<br />
| name = MP3tag<br />
| screenshot = [[Image:MP3tag-screenshot.png|250px|MP3tag screenshot]]<br />
| caption = powerful and easy-to-use tool to edit metadata <br />
| maintainer = Florian Heidenreich<br />
| stable_release = [http://www.mp3tag.de/en/download.html 2.38] (29.04.2007)<br />
| preview_release = [http://developer.mp3tag.de/ 2.38a] (16.06.2007)<br />
| operating_system = Windows<br />
| use = Metadata<br />
| license = Freeware <br />
| website = [http://www.mp3tag.de/en/ http://www.mp3tag.de]<br />
}}<br />
<br />
=Introduction=<br />
'''MP3tag''' is an powerful [[Metadata|metadata]] editor for lot's of common audio formats. You can rename files based on the [[Tagging|tag]] information, replace characters or words from tags and filenames, import/export tag information, create playlists.<br />
<br />
The program supports online freedb database lookups for selected files, allowing you to automatically gather proper tag information for select files or CDs.<br />
<br />
==Features==<br />
<br />
* Write [[ID3v1.1]]-, [[ID3v2]]-, [[APEv2]]-Tags and [[Vorbis_Comment|Vorbis Comments]] to multiple files at once<br />
* Full Unicode support<br />
* Support for embedded cover art<br />
* Automatically create playlists<br />
* Recursive subfolders support<br />
* Remove parts or the entire tag of multiple files<br />
* Rename files based on the tag information<br />
* Import tags from filenames<br />
* Format tags and filenames<br />
* Replace characters or words from tags and filenames<br />
* Regular Expressions<br />
* Export tag information to user-defined formats (like html, rtf, csv, xml)<br />
* Import tag information from online databases like freedb or Amazon (also by text-search)<br />
* Import tag information from local freedb databases<br />
* Support for [[ID3v2|ID3v2.3]] (ISO-8859-1 and UTF-16) and [[ID3v2|ID3v2.4]] with [[UTF-8]]<br />
<br />
==Supported formats==<br />
<br />
* Advanced Audio Coding ([[AAC|aac]])<br />
* Free Lossless Audio Codec ([[FLAC|flac]])<br />
* Monkey's Audio ([[APE|ape]])<br />
* Mpeg Layer 3 ([[MP3|mp3]])<br />
* MPEG-4 ([[MP4|mp4]] / [[M4A|m4a]] / m4b / [[iTunes]] compatible)<br />
* Musepack ([[MPC|mpc]])<br />
* Ogg Vorbis ([[Ogg_Vorbis|ogg]])<br />
* OptimFROG ([[OptimFROG|ofr]])<br />
* OptimFROG DualStream (ofs)<br />
* Speex ([[Speex|spx]])<br />
* TAK ([[TAK]])<br />
* True Audio (tta)<br />
* Windows Media Audio ([[WMA|wma]])<br />
* WavPack ([[WavPack|wv]])<br />
<br />
==Operating Systems==<br />
* Win 2000<br />
* Win XP<br />
* Win 2003<br />
* Win Vista<br />
<br />
==External links==<br />
* [http://www.mp3tag.de/en/ Mp3tag: Homepage]<br />
* [http://www.mp3tag.de/en/download.html Mp3tag: Download]<br />
* [http://www.anytag.de/forums/ Mp3tag: official forum]<br />
* [http://www.mp3tag.de/download/mp3tagversion.en.html Mp3tag: changelog]<br />
* [http://wiki.slimdevices.com/index.cgi?Mp3tagGuide Mp3tagGuide - Slim Devices]<br />
* [http://www.anytag.de/forums/index.php?showtopic=1794 Mp3tag: additional Web Sources]<br />
<br />
[[Category:Software]]</div>Gottkaiserhttps://wiki.hydrogenaud.io/index.php?title=BonkEncBonkEnc2007-06-04T16:26:25Z<p>Gottkaiser: version update</p>
<hr />
<div>{{Software Infobox<br />
| name = BonkEnc<br />
| screenshot = [[Image:BonkEnc-screenshot.png|250px|BonkEnc screenshot]]<br />
| caption = Slim CD ripper, audio encoder and converter <br />
| maintainer = Robert Kausch<br />
| stable_release = 1.0.3 (04. June 2007)<br />
| preview_release = <br />
| operating_system = Windows<br />
| use = Digital Audio Extraction<br />
| license = GPL <br />
| website = [http://www.bonkenc.org http://www.bonkenc.org]<br />
}}<br />
<br />
=Introduction=<br />
'''BonkEnc''' is a CD ripper, audio encoder and converter for various formats. It's a very slim program and can be run from a USB stick. BonkEnc is available under GPL liscense and has native support for numerous languages. <br />
<br />
== Features ==<br />
* CD ripping<br />
** [[Cdparanoia]] mode<br />
** jitter correction<br />
** CD Text support<br />
* [[Transcoding]] from on to another format<br />
* using Compact Disc Database (CDDB)<br />
* support [[ID3v1]], [[ID3v2]], MP4-Metadata and [[Vorbis_comment|Vorbis comment]] [[Tags]]<br />
* keeps image tags when converting from FLAC to MP3<br />
* creating cue sheets and playlists<br />
* full UTF-8 Unicode support<br />
* additional command line interface (CD Ripping/Encoding)<br />
<br />
== Supported Formats ==<br />
* [[MP3]] ([[LAME]])<br />
* [[AAC]] ([[FAAC]])<br />
* [[MP4]]/[[M4A]]<br />
* [[Ogg_Vorbis|Ogg Vorbis]]<br />
* [[FLAC]]<br />
* [http://www.logarithmic.net/pfh/bonk Bonk] v0.11<br />
<br />
== Recommended Settings ==<br />
{| border="0" valign="top"<br />
|<br />
* [[LAME]] MP3 Encoder v3.97:<br />
* [[FAAC]] MP4/AAC Encoder v1.25:<br />
* Ogg Vorbis Encoder ([[aoTuV|aoTuV beta 5]]):<br />
* FLAC Audio Encoder v1.1.4:<br />
||<br />
[[LAME#Recommended_encoder_settings|here]]<br/><br />
to add<br/><br />
[[Recommended_Ogg_Vorbis#Recommended_Encoder_Settings|here]]<br/><br />
Use preset 5<br />
|}<br />
<br />
== Supported languages ==<br />
{| border="0" -valign="top"<br />
|-valign="top"<br />
||<br />
* Catalan<br />
* Chinese<br />
* Czech<br />
* Danish<br />
* Dutch<br />
* English<br />
* Esperanto<br />
* Finnish<br />
||<br />
* French<br />
* German<br />
* Greek<br />
* Hungarian<br />
* Italian<br />
* Japanese<br />
* Korean<br />
* Lithuanian<br />
||<br />
* Polish<br />
* Portuguese<br />
* Russian<br />
* Serbian<br />
* Slovak<br />
* Spanish<br />
* Swedish<br />
* Turkish<br />
||<br />
* Ukrainian<br />
* Romanian<br />
|}<br />
<br />
<br />
<br />
== External links==<br />
* [http://www.bonkenc.org/ BonkEnc: Homepage]<br />
* [http://www.bonkenc.org/index.php?option=com_content&task=blogcategory&id=3&Itemid=33 BonkEnc: Download]<br />
* [http://sourceforge.net/forum/?group_id=27149 BonkEnc: official forum] official support forum <br />
<br />
[[Category:Software]]</div>Gottkaiserhttps://wiki.hydrogenaud.io/index.php?title=HarddiskoggHarddiskogg2007-05-26T11:25:41Z<p>Gottkaiser: </p>
<hr />
<div>#REDIRECT [[HarddiskOgg]]</div>Gottkaiserhttps://wiki.hydrogenaud.io/index.php?title=HarddiskOggHarddiskOgg2007-05-26T11:24:04Z<p>Gottkaiser: </p>
<hr />
<div>{{Software Infobox<br />
| name = HarddiskOgg<br />
| screenshot = [[Image:HarddiskOgg-screenshot.png|250px|HarddiskOgg screenshot]]<br />
| caption = recording streams in lot's of formats <br />
| maintainer = Jan Lellmann<br />
| stable_release = [http://www.fridgesoft.de/downloads.php 2.52] (07.02.2007)<br />
| preview_release = <br />
| operating_system = Windows<br />
| use = recording<br />
| license = GPL<br />
| website = [http://www.fridgesoft.de/harddiskogg.php http://www.fridgesoft.de/harddiskogg.php]<br />
}}<br />
<br />
=Introduction=<br />
'''HarddiskOgg''' takes a wave input stream from a Windows compatible sampling device (including microphone input and line in) and converts it to an [[Ogg_Vorbis|Ogg Vorbis]] / Wave / [[Monkey%27s_Audio|Monkey's Audio]] / [[MP3]] stream. This happens in realtime.<br />
<br />
On decent PCs, you can even play the [[Ogg_Vorbis|Ogg Vorbis]] or [[MP3]] file with your favorite player while the recording goes on. This means you can listen for example to the radio in near realtime, but you can take a break whenever you want.<br />
With some TV card/sound card combinations it is possible that sound recorded from TV is very low on volume. HarddiskOgg will automatically amplify the input signal before the encoding stage.<br />
<br />
==Features==<br />
* Real-time encoding with bitrates from 32kbit/sec. up to 320kbit/sec.<br />
* Stereo or mono recording from 8kHz to 48kHz<br />
* Automatic numbering of output files<br />
* Can be placed in the systray and activated by a single click<br />
* Smart on-the-fly normalization for low-volume sources<br />
* Uses the free Ogg Vorbis encoding engine<br />
* LAME MP3 encoder compatible. (You have to include [[lame]] encoding DLL, due to patent issues)<br />
* Command line mode for easy integration or scheduled recordings<br />
<br />
==Supported formats==<br />
* [[Ogg_Vorbis|Ogg Vorbis]]<br />
* [[Monkey%27s_Audio|Monkey's Audio]]<br />
* Wave<br />
* [[MP3]] ([[LAME]])<br />
<br />
==Operating Systems==<br />
* Win 95<br />
* Win 98<br />
* Win 2000<br />
* Win XP<br />
<br />
==External links==<br />
* [http://www.fridgesoft.de/harddiskogg.php HarddiskOgg: Homepage]<br />
* [http://www.fridgesoft.de/downloads.php HarddiskOgg: Download]<br />
* [http://www.fridgesoft.de/bb/ HarddiskOgg: official forum]<br />
* [http://www.hydrogenaudio.org/forums/index.php?showtopic=55117 HarddiskOgg: Hydrogenaudio forum]<br />
<br />
[[Category:Software]]</div>Gottkaiserhttps://wiki.hydrogenaud.io/index.php?title=File:HarddiskOgg-screenshot.pngFile:HarddiskOgg-screenshot.png2007-05-26T11:22:37Z<p>Gottkaiser: HarddiskOgg screenshot</p>
<hr />
<div>HarddiskOgg screenshot</div>Gottkaiserhttps://wiki.hydrogenaud.io/index.php?title=File:HarddiskOgg-screenshot.gifFile:HarddiskOgg-screenshot.gif2007-05-26T11:03:15Z<p>Gottkaiser: HarddiskOgg screenshot</p>
<hr />
<div>HarddiskOgg screenshot</div>Gottkaiserhttps://wiki.hydrogenaud.io/index.php?title=LameLame2007-05-26T10:59:56Z<p>Gottkaiser: </p>
<hr />
<div>#REDIRECT [[LAME]]</div>Gottkaiserhttps://wiki.hydrogenaud.io/index.php?title=MP3packerMP3packer2007-05-25T18:42:44Z<p>Gottkaiser: </p>
<hr />
<div>{{Software Infobox<br />
| name = MP3packer<br />
| screenshot = [[Image:MP3packer-screenshot.png|250px|MP3packer screenshot]]<br />
| caption = rearrange and compress MP3 data<br />
| maintainer = Reed Wilson<br />
| stable_release = [http://omion.dyndns.org/mp3packer/mp3packer-1.17.rar 1.17] (23-05-2007)<br />
| preview_release = <br />
| operating_system = Windows and Linux<br />
| use = rearrange, compress<br />
| license = GPL<br />
| website = [http://omion.dyndns.org/mp3packer/mp3packer.html/ Homepage]<br />
}}<br />
<br />
=Introduction=<br />
'''MP3packer''' is a program which can rearrange the data within an [[MP3]] to fulfill specific goals. By default, the program generates the smallest [[MP3]] possible (with the least padding). However, many people also use it to turn [[VBR]] files into [[CBR]] for use with players which don't support [[VBR]].<br />
<br />
It can save space by storing frame data in the smallest possible frame. Usually MP3s are already stored in the most efficient way possible. However, for high-bitrate [[CBR]] files (like --preset insane) there can be a lot of wasted space<br />
<br />
==Features==<br />
<br />
* Can make --preset insane files up to 10% smaller [[lossless|losslessly]] (depending on the [[LAME]] version used)<br />
* Squeezes out all the padding it can from any [[MP3]] (Will not produce a larger file, unless you use the -b switch or something goes wrong)<br />
* Writes valid [[LAME]] / [http://de.wikipedia.org/wiki/Xing-Header/ XING header] for proper [[VBR]] seeking<br />
* Many people also use this backwards, to losslessly turn [[VBR]] into larger [[CBR]] files to humor players which can't handle [[VBR]]<br />
* Includes a brute-force compression optimization option to further compress files<br />
* works on directories<br />
<br />
==How It Works==<br />
'''-z switch:'''<br />
The default operation is to choose the minimum frame size to fit the data and will also minimize the data size. This is completely [[lossless]], and is equivalent to decompressing a ZIP file and recompressing with a more aggressive setting. It attempts to minimize the data by doing a brute-force search for the optimal parameters, so it takes much longer than it would normally.<br />
<br />
'''-b switch:'''<br />
Setting the -b switch will set the minimum bitrate for each frame. Using this switch will make more room in small frames for other frames' data, so it will also generally reduce the maximum bitrate as well. There is no direct control over the maximum bitrate, since there may simply be too much data to fit into a smaller frame. The exact format of the parameter is a bit odd: if the bitrate given is a valid frame bitrate, the minimum bitrate is dithered between padded and unpadded frames. If the bitrate is one more than a valid frame bitrate, then the minimum is a padded frame of bitrate one less than the given. Anything else is rounded up to the next highest unpadded bitrate. <br />
<br />
'''-r, -R switches:'''<br />
After mp3packer has chosen an output bitrate for a given frame, there is generally a range of positions to put the actual data. The data can be packed as much as possible into the previous frame, or it can be set to fill up the current frame as much as possible. Usually it is best to put as much as possible into previous frames, since this will maximize the space available for any subsequent frames. However, if the minimum bitrate is adds enough padding, there is no reason to cram the data into previous frames; it's just going to move around the padding.<br />
The default is to pack as far behind as possible if the -b switch is not given, since there is usually no problem filling up the frames. If a minimum bitrate is specified then the frames are pushed as far up as possible without affecting any of the following frames.<br />
The -r switch will attempt to always push data as far up as possible, even if a minimum bitrate is not specified. Conversely, the -R switch will push the data into previous frames as possible<br />
<br />
==Operating Systems==<br />
* Windows<br />
* Linux<br />
* should work perfectly on any other platform with an OCaml port<br />
<br />
==External links==<br />
* [http://omion.dyndns.org/mp3packer/mp3packer.html/ MP3packer: Homepage]<br />
* [http://omion.dyndns.org/mp3packer/mp3packer-1.17.rar MP3packer: Download]<br />
* [http://www.hydrogenaudio.org/forums/index.php?showtopic=40780 MP3packer: Windows GUI]<br />
* [http://omion.dyndns.org/mp3packer/mp3packer.html#changelog MP3packer: changelog]<br />
* [http://www.hydrogenaudio.org/forums/index.php?showtopic=32379 MP3packer: hydrogenaudio forum]<br />
<br />
[[Category:Software]]</div>Gottkaiserhttps://wiki.hydrogenaud.io/index.php?title=Mp3packerMp3packer2007-05-25T18:34:56Z<p>Gottkaiser: </p>
<hr />
<div>#REDIRECT [[MP3packer]]</div>Gottkaiserhttps://wiki.hydrogenaud.io/index.php?title=MP3packerMP3packer2007-05-25T18:31:12Z<p>Gottkaiser: </p>
<hr />
<div>{{Software Infobox<br />
| name = MP3packer<br />
| screenshot = [[Image:MP3packer-screenshot.png|250px|MP3packer screenshot]]<br />
| caption = rearrange and compress MP3 data<br />
| maintainer = Reed Wilson<br />
| stable_release = [http://omion.dyndns.org/mp3packer/mp3packer-1.17.rar 1.17] (23-05-2007)<br />
| preview_release = <br />
| operating_system = Windows and major Unix<br />
| use = rearrange, compress<br />
| license = GPL<br />
| website = [http://omion.dyndns.org/mp3packer/mp3packer.html/ Homepage]<br />
}}<br />
<br />
=Introduction=<br />
'''MP3packer''' is a program which can rearrange the data within an [[MP3]] to fulfill specific goals. By default, the program generates the smallest [[MP3]] possible (with the least padding). However, many people also use it to turn [[VBR]] files into [[CBR]] for use with players which don't support [[VBR]].<br />
<br />
It can save space by storing frame data in the smallest possible frame. Usually MP3s are already stored in the most efficient way possible. However, for high-bitrate [[CBR]] files (like --preset insane) there can be a lot of wasted space<br />
<br />
==Features==<br />
<br />
* Can make --preset insane files up to 10% smaller [[lossless|losslessly]] (depending on the [[LAME]] version used)<br />
* Squeezes out all the padding it can from any [[MP3]] (Will not produce a larger file, unless you use the -b switch or something goes wrong)<br />
* Writes valid [[LAME]] / [http://de.wikipedia.org/wiki/Xing-Header/ XING header] for proper [[VBR]] seeking<br />
* Many people also use this backwards, to losslessly turn [[VBR]] into larger [[CBR]] files to humor players which can't handle [[VBR]]<br />
* Includes a brute-force compression optimization option to further compress files<br />
* works on directories<br />
<br />
==How It Works==<br />
'''-z switch:'''<br />
The default operation is to choose the minimum frame size to fit the data and will also minimize the data size. This is completely [[lossless]], and is equivalent to decompressing a ZIP file and recompressing with a more aggressive setting. It attempts to minimize the data by doing a brute-force search for the optimal parameters, so it takes much longer than it would normally.<br />
<br />
'''-b switch:'''<br />
Setting the -b switch will set the minimum bitrate for each frame. Using this switch will make more room in small frames for other frames' data, so it will also generally reduce the maximum bitrate as well. There is no direct control over the maximum bitrate, since there may simply be too much data to fit into a smaller frame. The exact format of the parameter is a bit odd: if the bitrate given is a valid frame bitrate, the minimum bitrate is dithered between padded and unpadded frames. If the bitrate is one more than a valid frame bitrate, then the minimum is a padded frame of bitrate one less than the given. Anything else is rounded up to the next highest unpadded bitrate. <br />
<br />
'''-r, -R switches:'''<br />
After mp3packer has chosen an output bitrate for a given frame, there is generally a range of positions to put the actual data. The data can be packed as much as possible into the previous frame, or it can be set to fill up the current frame as much as possible. Usually it is best to put as much as possible into previous frames, since this will maximize the space available for any subsequent frames. However, if the minimum bitrate is adds enough padding, there is no reason to cram the data into previous frames; it's just going to move around the padding.<br />
The default is to pack as far behind as possible if the -b switch is not given, since there is usually no problem filling up the frames. If a minimum bitrate is specified then the frames are pushed as far up as possible without affecting any of the following frames.<br />
The -r switch will attempt to always push data as far up as possible, even if a minimum bitrate is not specified. Conversely, the -R switch will push the data into previous frames as possible<br />
<br />
==Operating Systems==<br />
* Windows<br />
* Linux<br />
* should work perfectly on any other platform with an OCaml port<br />
<br />
==External links==<br />
* [http://omion.dyndns.org/mp3packer/mp3packer.html/ MP3packer: Homepage]<br />
* [http://omion.dyndns.org/mp3packer/mp3packer-1.17.rar MP3packer: Download]<br />
* [http://www.hydrogenaudio.org/forums/index.php?showtopic=40780 MP3packer: Windows GUI]<br />
* [http://omion.dyndns.org/mp3packer/mp3packer.html#changelog MP3packer: changelog]<br />
* [http://www.hydrogenaudio.org/forums/index.php?showtopic=32379 MP3packer: hydrogenaudio forum]<br />
<br />
[[Category:Software]]</div>Gottkaiserhttps://wiki.hydrogenaud.io/index.php?title=File:MP3packer-screenshot.pngFile:MP3packer-screenshot.png2007-05-25T18:21:38Z<p>Gottkaiser: Screenshot of WinMP3Packer</p>
<hr />
<div>Screenshot of WinMP3Packer</div>Gottkaiserhttps://wiki.hydrogenaud.io/index.php?title=LosslessLossless2007-05-25T16:31:24Z<p>Gottkaiser: /* Popular lossless formats */ added TAK</p>
<hr />
<div>'''Lossless compression''' is a compression methodology in which the result of the compression can be restored faithfully, i.e. bit-by-bit identical with the uncompressed data.<br />
<br />
In a nutshell, it is somewhat like compressing a Waveform file with ZIP or RAR.<br />
<br />
The difference between 'mere' ZIP/RAR is that lossless compression algorithms are especially tuned and designed for the characteristics of Waveform data, thus achieving compression far greater than can be achieved by generic compression utilities.<br />
<br />
As lossless compression preserves all information of the original Waveform file, audio compressed with lossless compression will unavoidably be larger than audio compressed with [[lossy]] compression. However, this disadvantage is more than offset by lossless' ability to be [[transcoding|transcoded]] to other lossless format <u>without</u> any quality degradation.<br />
<br />
<br />
==Popular lossless formats==<br />
* [[Apple Lossless]] ([[ALAC]])<br />
<br />
* [[FLAC]]<br />
<br />
* [[Lossless Audio]] ([[LA]])<br />
<br />
* [[LPAC]]<br />
<br />
* [[MLP]]<br />
<br />
* [[Monkey's Audio]] ([[APE]])<br />
<br />
* [[OptimFROG]]<br />
<br />
* [[RKAU]]<br />
<br />
* [[Shorten]] ([[SHN]])<br />
<br />
* [[TAK]]<br />
<br />
* [[TTA]]<br />
<br />
* [[WavPack]]<br />
<br />
* [[WMA | WMA lossless]]<br />
<br />
==Oddball Formats==<br />
There are several old lossless formats that aren't really deserving of having an article all for themselves. Reasons are: lack of widespread support, lack of features, bad efficiency and, most importantly, it seems noone is really interested in them.<br />
<br />
Most of those would have disappeared by now, but they are being preserved for posterity at [[User:Rjamorim|rjamorim]]'s [http://www.rjamorim.com/rrw/ ReallyRareWares]<br />
<br />
; Advanced Digital Audio (ADA)<br />
<br />
* http://www.rjamorim.com/rrw/ada.html<br />
<br />
; Marian's a-Pac<br />
<br />
* http://www.marian.de/en/downloads#APAC<br />
* http://www.rjamorim.com/rrw/apac.html<br />
<br />
; AudioZip<br />
<br />
* http://www.rjamorim.com/rrw/audiozip.html<br />
<br />
; Dakx WAV<br />
<br />
* http://www.dakx.com/<br />
* http://www.rjamorim.com/rrw/daxwav.html<br />
<br />
; Entis Lab MIO<br />
<br />
* http://www.entis.gr.jp/eri/frame.html<br />
* http://www.rjamorim.com/rrw/mio.html<br />
<br />
; LiteWave<br />
<br />
* http://www.clearjump.com/products/LiteWave.html<br />
* http://www.rjamorim.com/rrw/litewave.html<br />
<br />
; Pegasus SPS<br />
<br />
* http://www.krishnasoft.com/sps.htm<br />
* http://www.rjamorim.com/rrw/pegasussps.html<br />
<br />
; RKaudio<br />
<br />
* http://www.msoftware.co.nz/downloads_page.php<br />
* http://rksoft.virtualave.net/rkau.html<br />
<br />
; Split2000<br />
<br />
* http://www.rjamorim.com/rrw/split2000.html<br />
<br />
; Sonarc<br />
<br />
* http://www.rjamorim.com/rrw/sonarc.html<br />
<br />
; VocPack<br />
<br />
* http://www.rjamorim.com/rrw/vocpack.html<br />
<br />
; WavArc<br />
<br />
* http://www.rjamorim.com/rrw/wavarc.html<br />
<br />
; WaveZip/MUSICompress<br />
<br />
* http://members.aol.com/_ht_a/sndspace/<br />
* http://www.rjamorim.com/rrw/wavezip.html<br />
<br />
<br />
Note that currently '''no single format can be considered best for all applications'''. Rather, the best format depends on the ''intended use'', as well as a number of other factors (such as licensing and file structure). For example, Shorten and FLAC are widely used for sharing live music because of their cross-platform support and speed. Monkey's Audio is popular among Windows users for its superior compression ratio.<br />
<br />
==Comparisons==<br />
''Note the specific assumptions and limitations of each comparison; in particular, results are sensitive to the music selected'''<br />
<br />
; http://web.inter.nl.net/users/hvdh/lossless/lossless.htm : Includes an interesting graph of encode/decode speeds vs. file size on the All Albums page<br />
<br />
; [[Lossless comparison]] : A comparision focusing more on codec features and less on absolute encoding efficiency. Also features a table comparing most popular codecs based on their features.<br />
<br />
; http://members.home.nl/w.speek/comparison.htm : Performance Comparison of Lossless Audio Compressors - Compares file size, encode speed, decode speed for [[APE]], [[FLAC]], [[LPAC]], [[WavPack]], Shorten ([[SHN]]), [[RKAU]], [[OptimFROG]], [[LA]], [[WMA | WMA Lossless]]. Updated 5-2003<br />
<br />
; http://www.bobulous.org.uk/misc/lossless_audio_2006.html : Lossless audio formats - A comparison of the rip-and-encode speed and album file size of six different lossless formats: [[WAV|uncompressed Wave]], [[FLAC]], [[WavPack]], [[SHN|Shorten]], [[APE|Monkey's Audio]], and [[OptimFROG]]. First published on 22nd May 2006.<br />
<br />
==External Links==<br />
* [http://www.losslessaudioblog.com/ The Lossless Audio Blog] Lossless Audio News & Information Site.<br />
<br />
[[Category:Codecs]]</div>Gottkaiserhttps://wiki.hydrogenaud.io/index.php?title=WinampWinamp2007-05-19T19:49:37Z<p>Gottkaiser: version update</p>
<hr />
<div>{{Software Infobox|<br />
|name = Winamp<br />
|screenshot = [[Image:Winamp-screenshot.png|250px]]<br />
|caption = Modern Winamp skin<br />
|maintainer = Team Nullsoft<br />
|stable_release = [http://www.winamp.com/player/free.php 5.35]<br />
|preview_release = None<br />
|operating_system = Windows<br />
|use = Media Player<br />
|license = freeware<br />
|website = [http://www.winamp.com/ www.winamp.com]<br />
}}<br />
<br />
{{stub}}<br />
<br />
'''Winamp''' is a free music player for Windows developed by Nullsoft. A commercial "Pro" version is also available.<br />
<br />
The main advantages of Winamp is its ease of use. In addition, it is skinnable, and extensible using plugins. As of version 5.2, it fully supports multi-user (i.e. each user of your computer may have their own skin, playlist, and other settings).<br />
<br />
You can download Winamp at http://www.winamp.com/ . While you are there, you might also check their [http://www.winamp.com/skins/ skin library] and [http://www.winamp.com/plugins/ plugin library]. And also check the very useful [http://forums.winamp.com/ community (forums)] where new plugins are announced and publicly tested, and very minor updates (i.e. 0.001 version increment) are posted.<br />
<br />
== Features ==<br />
<br />
=== Free ===<br />
* Crippled CD burning (~8x)<br />
* Crippled CD ripping (~8x)<br />
* [[AAC]], [[MP4]], [[FLAC]] ([http://win32builds.sourceforge.net/flake/index.html Flake] encoder), [[WAV]], [[WMA]] encoding<br />
* [[Transcoding]] of the different audio formats<br />
* [[ReplayGain]] support<br />
* Media library<br />
* Full Unicode support<br />
* [[ReplayGain]] scanner to apply Album Gain or Track Gain to the tags<br />
<br />
=== Pro ===<br />
The Pro version, which can be bought online, comes with some additional features compared to the free version.<br />
* Burn CDs at full speed<br />
* Rip CDs at full speed<br />
* additional MP3 encoding<br />
<br />
== Supported formats ==<br />
=== Playback ===<br />
Directly supported formats (i.e. provided with installer) include: [[MP1]], [[MP2]], [[MP3]], [[WAV]], [[AAC]], [[WMA]], [[Ogg Vorbis]], [[MIDI]], [[FLAC]], and [[Module]]<br />
<br />
Plugins also exist for many other formats, such as [[TTA]], [[WavPack]], [[Musepack]], [[TAK]]. Go to Winamp's [http://www.winamp.com/plugins/ plugin library] to download.<br />
<br />
== Supported languages ==<br />
* English<br />
<br />
== Supported platforms ==<br />
* Windows<br />
<br />
== Recommended plugins ==<br />
* '''Playlist Separator''' - provides a customizable separator line to delimit albums in a long playlist.<br />
* '''MojoMaster''' - sexy dancer visualization.<br />
<br />
== External links ==<br />
* [http://www.winamp.com Winamp: Homepage]<br />
* [http://winamp.com/player/ Winamp: Download]<br />
* [http://winamp.com/about/story.php More information]<br />
<br />
<br />
<br />
[[Category:Software]]<br />
[[Category:Media Players]]</div>Gottkaiserhttps://wiki.hydrogenaud.io/index.php?title=WinampWinamp2007-05-07T20:33:25Z<p>Gottkaiser: </p>
<hr />
<div>{{Software Infobox|<br />
|name = Winamp<br />
|screenshot = [[Image:Winamp-screenshot.png|250px]]<br />
|caption = Modern Winamp skin<br />
|maintainer = Team Nullsoft<br />
|stable_release = [http://www.winamp.com/player/free.php 5.34]<br />
|preview_release = None<br />
|operating_system = Windows<br />
|use = Media Player<br />
|license = freeware<br />
|website = [http://www.winamp.com/ www.winamp.com]<br />
}}<br />
<br />
{{stub}}<br />
<br />
'''Winamp''' is a free music player for Windows developed by Nullsoft. A commercial "Pro" version is also available.<br />
<br />
The main advantages of Winamp is its ease of use. In addition, it is skinnable, and extensible using plugins. As of version 5.2, it fully supports multi-user (i.e. each user of your computer may have their own skin, playlist, and other settings).<br />
<br />
You can download Winamp at http://www.winamp.com/ . While you are there, you might also check their [http://www.winamp.com/skins/ skin library] and [http://www.winamp.com/plugins/ plugin library]. And also check the very useful [http://forums.winamp.com/ community (forums)] where new plugins are announced and publicly tested, and very minor updates (i.e. 0.001 version increment) are posted.<br />
<br />
=== Functions ===<br />
----<br />
==== Free ====<br />
* Crippled CD burning (~8x)<br />
* Crippled CD ripping (~8x)<br />
* [[AAC]], [[MP4]], [[FLAC]] ([http://win32builds.sourceforge.net/flake/index.html Flake] encoder), [[WAV]], [[WMA]] encoding<br />
* [[Transcoding]] of the different audio formats<br />
* [[ReplayGain]] support<br />
* Media library<br />
* Full Unicode support<br />
* [[ReplayGain]] scanner to apply Album Gain or Track Gain to the tags<br />
<br />
==== Pro ====<br />
The Pro version, which can be bought online, comes with some additional features compared to the free version.<br />
* Burn CDs at full speed<br />
* Rip CDs at full speed<br />
* additional MP3 encoding<br />
<br />
<br />
=== Supported formats ===<br />
----<br />
==== Playback ====<br />
Directly supported formats (i.e. provided with installer) include: [[MP1]], [[MP2]], [[MP3]], [[WAV]], [[AAC]], [[WMA]], [[Ogg Vorbis]], [[MIDI]], [[FLAC]], and [[Module]]<br />
<br />
Plugins also exist for many other formats, such as [[TTA]], [[WavPack]], [[Musepack]], [[TAK]]. Go to Winamp's [http://www.winamp.com/plugins/ plugin library] to download.<br />
<br />
=== Supported languages ===<br />
----<br />
* English<br />
<br />
<br />
=== Supported platforms ===<br />
----<br />
* Windows<br />
<br />
<br />
=== Some recommended plugins ===<br />
----<br />
* '''Playlist Separator''' - provides a customizable separator line to delimit albums in a long playlist.<br />
* '''MojoMaster''' - sexy dancer visualization.<br />
<br />
<br />
=== External links ===<br />
----<br />
* [http://www.winamp.com Winamp: Homepage]<br />
<br />
* [http://winamp.com/player/ Winamp: Download]<br />
<br />
* [http://winamp.com/about/story.php More information]<br />
<br />
<br />
<br />
[[Category:Software]]<br />
[[Category:Media Players]]</div>Gottkaiserhttps://wiki.hydrogenaud.io/index.php?title=WinampWinamp2007-05-07T20:32:01Z<p>Gottkaiser: added Software Infobox</p>
<hr />
<div>{{Software Infobox|<br />
|name = Winamp<br />
|screenshot = [[Image:Winamp-screenshot.png|250px]]<br />
|caption = Modern Winamp skin<br />
|maintainer = Team Nullsoft<br />
|stable_release = [http://www.winamp.com/player/free.php 5.34]<br />
|preview_release = None<br />
|operating_system = Windows<br />
|use = Media Player<br />
|license = freeware<br />
|website = [http://www.winamp.com/ www.winamp.com]<br />
}}<br />
<br />
{{stub}}<br />
<br />
'''Winamp''' is a free music player for Windows developed by Nullsoft. A commercial "Pro" version is also available.<br />
<br />
The main advantages of Winamp is its ease of use. In addition, it is skinnable, and extensible using plugins. As of version 5.2, it fully supports multi-user (i.e. each user of your computer may have their own skin, playlist, and other settings).<br />
<br />
You can download Winamp at http://www.winamp.com/ . While you are there, you might also check their [http://www.winamp.com/skins/ skin library] and [http://www.winamp.com/plugins/ plugin library]. And also check the very useful [http://forums.winamp.com/ community (forums)] where new plugins are announced and publicly tested, and very minor updates (i.e. 0.001 version increment) are posted.<br />
<br />
=== Functions ===<br />
----<br />
==== Free ====<br />
* Crippled CD burning (~8x)<br />
* Crippled CD ripping (~8x)<br />
* [[AAC]], [[MP4]], [[FLAC]] ([http://win32builds.sourceforge.net/flake/index.html Flake] encoder), [[WAV]], [[WMA]] encoding<br />
* [[Transcoding]] of the different audio formats<br />
* [[ReplayGain]] support<br />
* Media library<br />
* Full Unicode support<br />
* [[ReplayGain]] scanner to apply Album Gain or Track Gain to the tags<br />
<br />
==== Pro ====<br />
The Pro version, which can be bought online, comes with some additional features compared to the free version.<br />
* Burn CDs at full speed<br />
* Rip CDs at full speed<br />
* additional MP3 encoding<br />
<br />
<br />
=== Supported formats ===<br />
----<br />
==== Playback ====<br />
Directly supported formats (i.e. provided with installer) include: [[MP1]], [[MP2]], [[MP3]], [[WAV]], [[AAC]], [[WMA]], [[Ogg Vorbis]], [[MIDI]], [[FLAC]], and [[Module]]<br />
<br />
Plugins also exist for many other formats, such as [[TTA]], [[WavPack]] and [[Musepack]]. Go to Winamp's [http://www.winamp.com/plugins/ plugin library] to download.<br />
<br />
=== Supported languages ===<br />
----<br />
* English<br />
<br />
<br />
=== Supported platforms ===<br />
----<br />
* Windows<br />
<br />
<br />
=== Some recommended plugins ===<br />
----<br />
* '''Playlist Separator''' - provides a customizable separator line to delimit albums in a long playlist.<br />
* '''MojoMaster''' - sexy dancer visualization.<br />
<br />
<br />
=== External links ===<br />
----<br />
* [http://www.winamp.com Winamp: Homepage]<br />
<br />
* [http://winamp.com/player/ Winamp: Download]<br />
<br />
* [http://winamp.com/about/story.php More information]<br />
<br />
<br />
<br />
[[Category:Software]]<br />
[[Category:Media Players]]</div>Gottkaiserhttps://wiki.hydrogenaud.io/index.php?title=File:Winamp-screenshot.pngFile:Winamp-screenshot.png2007-05-07T20:24:45Z<p>Gottkaiser: Winamp modern skin screenshot</p>
<hr />
<div>Winamp modern skin screenshot</div>Gottkaiserhttps://wiki.hydrogenaud.io/index.php?title=TAKTAK2007-05-02T22:48:03Z<p>Gottkaiser: /* Recommended Settings */</p>
<hr />
<div>{{stub}}<br />
{{Codec Infobox<br />
| name = TAK<br />
| logo = [[Image:TAK-logo.png]]<br />
| type = lossless<br />
| purpose = lossless audio compression.<br />
| maintainer = Thomas Becker <br />
| recommended_encoder = TAK encoder<br />
| recommended_text = TAK v1.0.1<br />
| website = [http://www.thbeck.de/Tak/Tak.html ThBeck.de/Tak/Tak.html] ''(german)''<br />
}}<br />
== General aspects of the format ==<br />
'''TAK''' ('''T'''om's lossless '''A'''udio '''K'''ompressor) is a lossless audio compressor which promises compression performance similar to [[Monkey's Audio]] "High" and decompression speed similar to [[FLAC]]. While the basic format is ready and safe to use, many features such as internal tagging, playback support for other players are yet to be implemented.<br />
<br />
=== Features ===<br />
* High compression<br />
* Fast compression and decompression speed<br />
* Streaming support<br />
* Error tolerance<br />
** Single bit error will never affect more than 250 ms<br />
* Error detection<br />
** Each frame protected by a 24-bit checksum (CRC)<br />
<br />
=== Pros ===<br />
* Fast encoding speed<br />
** TAK "Extra" encodes as fast as [[FLAC]] -8 while providing better compression<br />
** TAK "Turbo" encodes several times faster than [[FLAC]] -8 while providing better compression<br />
* Fast decompression speed (on par with [[FLAC]] / [[WavPack]])<br />
* Good compression levels (on par with [[Monkey's Audio]] High)<br />
* Error Robustness<br />
* Fast Seeking<br />
<br />
=== Cons ===<br />
* Closed Source (at the moment)<br />
* No hardware support<br />
* Very limited software support (Only [[Winamp]] & [[Foobar2000]] plugins at the moment)<br />
<br />
<br />
== Hardware and software that support TAK ==<br />
=== Hardware ===<br />
* None<br />
<br />
=== Software ===<br />
* TAK SDK 1.0.3 - Software Development Kit for TAK [http://www.hydrogenaudio.org/forums/index.php?showtopic=54159 here]<br />
* Winamp Plugin 1.0.2 [http://www.hydrogenaudio.org/forums/index.php?showtopic=54159 here]<br />
* foo_input_tak, TAK decoder for [[Foobar2000]] [http://www.hydrogenaudio.org/forums/index.php?showtopic=54087 here]<br />
* [[MP3tag]] - universal tag editor with support for TAK<br />
<br />
<br />
== Recommended Settings ==<br />
* Default compression: "default" is the most attractive setting. (At compression levels close to [[Monkey's Audio]] High (<0.4% difference), it is able to encode more quickly.)<br />
takc -e [input file]<br />
* Highest compression: "Extra" preset with "Maximum" switch. (This will create files which are comparable in size to file created using [[Monkey's Audio]] High.Decompression speed is comparable to [[WavPack]] Normal.)<br />
takc -e -p4m [input file]<br />
* Fastest compression: "Turbo" preset (This will create files which are comparable in size to [[Monkey's Audio]] Fast or [[WavPack]] High. Decompression speed is comparable to [[FLAC]] 0.)<br />
takc -e -p0 [input file]<br />
<br />
'''TAK performance Graph'''<br />
[[Image:TAK_performance_graph.png|center|TAK performance Graph]]<br />
<br />
== using TAK ==<br />
=== TAK with Foobar2000 ===<br />
* Copy the takc.exe to your [[foobar2000]] directory<br />
* Go to File -> Preferences -> Tools -> Converter<br />
* Set it up as shown:<br />
[[Image:tak.PNG|frame|center|TAK Encoder with Foobar]]<br />
NOTE: replace the -p4 with the desired compression level.<br />
<br />
* Use [[APEv2_specification|APEv2]] tagging (will be used as internal tagging)<br />
<br />
=== TAK with EAC ===<br />
[[EAC_and_TAK|Wiki guide to use TAK with EAC]]<br />
<br />
<br />
== Future Features ==<br />
* Unicode support<br />
* Piping support<br />
* MD5 audio checksums for verification and identification<br />
* A German version<br />
* Embedded cue sheets<br />
* Embedded cover art<br />
* Multichannel audio<br />
<br />
<br />
== Frequently asked questions ==<br />
* Is the codec safe for use?<br />
: Yes. To check, convert a WAVE to TAK and back and compare the two (or use foobar's bitcompare tool).<br />
<br />
* Why should I use TAK?<br />
: TAK offers high compression ratios with great decoding rates.<br />
<br />
* What can I compress with TAK?<br />
: TAK 1.0 can compress any integer-format (up to 24 bits per channel) PCM Windows Waveform file (.wav). Since piping support has not been added yet, you must convert your lossless files to wav first to convert to TAK.<br />
<br />
* What about hardware support?<br />
: None at the moment. Although, Turbo, Fast and Normal are the candidates for hardware playback.<br />
<br />
* When will the source be opened?<br />
: Yes, TAK will be open-source, as soon as the code is ported to C or C++ and documented. However, Thomas has mentioned that he would like to improve the codec before opening the source.<br />
<br />
<br />
== External links ==<br />
* [http://www.thbeck.de/Tak/Tak.html TAK: Homepage] (german)<br />
* [http://www.hydrogenaudio.org/forums/index.php?showtopic=54160 TAK: Release Announcement / Discussion Thread on HA] (english)<br />
* [http://www.hydrogenaudio.org/forums/index.php?showtopic=54159 TAK: Download]<br />
* [http://synthetic-soul.co.uk/comparison/lossless/ Comparison with Other Codecs (by Synthetic Soul)]<br />
* [http://flac.sourceforge.net/comparison.html An Updated Comparison (from FLAC Homepage)]<br />
<br />
[[Category:Lossless]]<br />
[[Category:Encoder/Decoder]]</div>Gottkaiserhttps://wiki.hydrogenaud.io/index.php?title=TAKTAK2007-05-02T22:45:27Z<p>Gottkaiser: /* Recommended Settings */ updated TAK_performance_graph.png</p>
<hr />
<div>{{stub}}<br />
{{Codec Infobox<br />
| name = TAK<br />
| logo = [[Image:TAK-logo.png]]<br />
| type = lossless<br />
| purpose = lossless audio compression.<br />
| maintainer = Thomas Becker <br />
| recommended_encoder = TAK encoder<br />
| recommended_text = TAK v1.0.1<br />
| website = [http://www.thbeck.de/Tak/Tak.html ThBeck.de/Tak/Tak.html] ''(german)''<br />
}}<br />
== General aspects of the format ==<br />
'''TAK''' ('''T'''om's lossless '''A'''udio '''K'''ompressor) is a lossless audio compressor which promises compression performance similar to [[Monkey's Audio]] "High" and decompression speed similar to [[FLAC]]. While the basic format is ready and safe to use, many features such as internal tagging, playback support for other players are yet to be implemented.<br />
<br />
=== Features ===<br />
* High compression<br />
* Fast compression and decompression speed<br />
* Streaming support<br />
* Error tolerance<br />
** Single bit error will never affect more than 250 ms<br />
* Error detection<br />
** Each frame protected by a 24-bit checksum (CRC)<br />
<br />
=== Pros ===<br />
* Fast encoding speed<br />
** TAK "Extra" encodes as fast as [[FLAC]] -8 while providing better compression<br />
** TAK "Turbo" encodes several times faster than [[FLAC]] -8 while providing better compression<br />
* Fast decompression speed (on par with [[FLAC]] / [[WavPack]])<br />
* Good compression levels (on par with [[Monkey's Audio]] High)<br />
* Error Robustness<br />
* Fast Seeking<br />
<br />
=== Cons ===<br />
* Closed Source (at the moment)<br />
* No hardware support<br />
* Very limited software support (Only [[Winamp]] & [[Foobar2000]] plugins at the moment)<br />
<br />
<br />
== Hardware and software that support TAK ==<br />
=== Hardware ===<br />
* None<br />
<br />
=== Software ===<br />
* TAK SDK 1.0.3 - Software Development Kit for TAK [http://www.hydrogenaudio.org/forums/index.php?showtopic=54159 here]<br />
* Winamp Plugin 1.0.2 [http://www.hydrogenaudio.org/forums/index.php?showtopic=54159 here]<br />
* foo_input_tak, TAK decoder for [[Foobar2000]] [http://www.hydrogenaudio.org/forums/index.php?showtopic=54087 here]<br />
* [[MP3tag]] - universal tag editor with support for TAK<br />
<br />
<br />
== Recommended Settings ==<br />
* Default compression: "default" is the most attractive setting. (At compression levels close to [[Monkey's Audio]] High (<0.4% difference), it is able to encode more quickly.)<br />
takc -e [input file]<br />
* highest compression: "Extra" preset with "Maximum" switch. (This will create files which are comparable in size to file created using Monkey's Audio High.Decompression speed is comparable to [[WavPack]] Normal.)<br />
takc -e -p4m [input file]<br />
* Fastest compression: "Turbo" preset (This will create files which are comparable in size to [[Monkey's Audio]] Fast or [[WavPack]] High. Decompression speed is comparable to [[FLAC]] 0.)<br />
takc -e -p0 [input file]<br />
<br />
'''TAK performance Graph'''<br />
[[Image:TAK_performance_graph.png|center|TAK performance Graph]]<br />
<br />
== using TAK ==<br />
=== TAK with Foobar2000 ===<br />
* Copy the takc.exe to your [[foobar2000]] directory<br />
* Go to File -> Preferences -> Tools -> Converter<br />
* Set it up as shown:<br />
[[Image:tak.PNG|frame|center|TAK Encoder with Foobar]]<br />
NOTE: replace the -p4 with the desired compression level.<br />
<br />
* Use [[APEv2_specification|APEv2]] tagging (will be used as internal tagging)<br />
<br />
=== TAK with EAC ===<br />
[[EAC_and_TAK|Wiki guide to use TAK with EAC]]<br />
<br />
<br />
== Future Features ==<br />
* Unicode support<br />
* Piping support<br />
* MD5 audio checksums for verification and identification<br />
* A German version<br />
* Embedded cue sheets<br />
* Embedded cover art<br />
* Multichannel audio<br />
<br />
<br />
== Frequently asked questions ==<br />
* Is the codec safe for use?<br />
: Yes. To check, convert a WAVE to TAK and back and compare the two (or use foobar's bitcompare tool).<br />
<br />
* Why should I use TAK?<br />
: TAK offers high compression ratios with great decoding rates.<br />
<br />
* What can I compress with TAK?<br />
: TAK 1.0 can compress any integer-format (up to 24 bits per channel) PCM Windows Waveform file (.wav). Since piping support has not been added yet, you must convert your lossless files to wav first to convert to TAK.<br />
<br />
* What about hardware support?<br />
: None at the moment. Although, Turbo, Fast and Normal are the candidates for hardware playback.<br />
<br />
* When will the source be opened?<br />
: Yes, TAK will be open-source, as soon as the code is ported to C or C++ and documented. However, Thomas has mentioned that he would like to improve the codec before opening the source.<br />
<br />
<br />
== External links ==<br />
* [http://www.thbeck.de/Tak/Tak.html TAK: Homepage] (german)<br />
* [http://www.hydrogenaudio.org/forums/index.php?showtopic=54160 TAK: Release Announcement / Discussion Thread on HA] (english)<br />
* [http://www.hydrogenaudio.org/forums/index.php?showtopic=54159 TAK: Download]<br />
* [http://synthetic-soul.co.uk/comparison/lossless/ Comparison with Other Codecs (by Synthetic Soul)]<br />
* [http://flac.sourceforge.net/comparison.html An Updated Comparison (from FLAC Homepage)]<br />
<br />
[[Category:Lossless]]<br />
[[Category:Encoder/Decoder]]</div>Gottkaiser