https://wiki.hydrogenaudio.org/api.php?action=feedcontributions&feedformat=atom&user=GigapodHydrogenaudio Knowledgebase - User contributions [en]2026-04-29T14:17:45ZUser contributionsMediaWiki 1.44.2https://wiki.hydrogenaudio.org/index.php?title=FAAC&diff=23198FAAC2012-06-26T10:36:21Z<p>Gigapod: </p>
<hr />
<div>FAAC is the oldest free and open-source AAC encoder. It is available for practically all Linux distributions. Present version (June 2012) is 1.28.<br />
<br />
== What is FAAC? ==<br />
'''FAAC''' stands for Freeware Advanced Audio Coder. FAAC is a software to compress audio in the [[AAC]] format.<br />
<br />
== Features ==<br />
<br />
Portable<br />
Reasonably fast<br />
LC, Main, LTP support<br />
DRM support through DreaM<br />
<br />
== Licensing ==<br />
Please note that although faac is available in source code format, its license is NOT the GPL. Quoting: "''FAAC is based on the original ISO MPEG reference code. The changes to this code are licensed under the LGPL license. The original license is not compatible with the LGPL, please be aware of this when using FAAC. The original license text can be found in the README file included in the download package.''" <br />
<br />
== Best Settings ==<br />
The default quality setting for faac, q=100, generates files at an average bitrate of approx. 128kbps. This quality level is good enough for casual, non-critical listening, but note that other encoders for AAC and other compressed formats may provide better quality files at similar bitrates.<br />
<br />
For better quality encoding, I suggest q=150, resulting in average bitrates around 175kbps. Based on my own (subjective) tests, at this quality level faac provides high quality artifact free music reproduction and is comparable in quality to proprietary AAC encoders at similar bitrates.<br />
<br />
Note that faac will by default wrap AAC data in an MP4 container for output files with the extensions .mp4 and .m4a.<br />
<br />
== Decoder ==<br />
A companion AAC decoder, faad2, is available.<br />
<br />
== External links ==<br />
<br />
*[http://http://www.audiocoding.com/faac.html FAAC at Audiocoding]<br />
<br />
[[Category:Software]]<br />
[[Category:Encoder/Decoder]]</div>Gigapodhttps://wiki.hydrogenaudio.org/index.php?title=FAAC&diff=23197FAAC2012-06-26T10:28:08Z<p>Gigapod: </p>
<hr />
<div>{{stub}}<br />
FAAC is the oldest free and open-source LC AAC encoder. It is available for practically all Linux distributions. Present version (June 2012) is 1.28.<br />
<br />
== What is FAAC? ==<br />
'''FAAC''' stands for Freeware Advanced Audio Coder. FAAC is a software to compress audio in the [[AAC]] format.<br />
<br />
== Features ==<br />
Features:<br />
<br />
Portable<br />
Reasonably fast<br />
LC, Main, LTP support<br />
DRM support through DreaM<br />
<br />
== Licensing ==<br />
Please note that although faac is available in source code format, its license is NOT the GPL. Quoting: "''FAAC is based on the original ISO MPEG reference code. The changes to this code are licensed under the LGPL license. The original license is not compatible with the LGPL, please be aware of this when using FAAC. The original license text can be found in the README file included in the download package.''" <br />
<br />
== Best Settings ==<br />
The default quality setting for faac, q=100, generates files at an average bitrate of approx. 128kbps. This quality level is good enough for casual, non-critical listening, but note that other encoders for aac and other compressed formats may provide better quality files at similar bitrates.<br />
<br />
For better quality encoding, I suggest q=150, resulting in average bitrates around 175kbps. Based on my own (subjective) tests, at this quality level faac provides high quality artifact free music reproduction and is comparable in quality to proprietary encoders at similar bitrates.<br />
<br />
== External links ==<br />
<br />
*[http://http://www.audiocoding.com/faac.html FAAC at Audiocoding]<br />
<br />
[[Category:Software]]<br />
[[Category:Encoder/Decoder]]</div>Gigapodhttps://wiki.hydrogenaudio.org/index.php?title=FAAC&diff=23196FAAC2012-06-26T10:22:41Z<p>Gigapod: </p>
<hr />
<div>{{stub}}<br />
FAAC is the oldest free and open-source LC AAC encoder. It is available for practically all Linux distributions. Present version (June 2012) is 1.28.<br />
<br />
== What is FAAC? ==<br />
'''FAAC''' stands for Freeware Advanced Audio Coder. FAAC is a software to compress audio in the [[AAC]] format.<br />
<br />
== Best Settings ==<br />
The default quality setting for faac, q=100, generates files at an average bitrate of approx. 128kbps. This quality level is good enough for casual, non-critical listening, but note that other encoders for aac and other compressed formats may provide better quality files at similar bitrates.<br />
<br />
For better quality encoding, I suggest q=150, resulting in average bitrates around 175kbps. Based on my own (subjective) tests, at this quality level faac provides high quality artifact free music reproduction and is comparable in quality to proprietary encoders at similar bitrates.<br />
<br />
== External links ==<br />
<br />
*[http://www.audiocoding.com/modules/wiki/?page=FAAC FAAC Wiki at Audiocoding]<br />
*[http://www.audiocoding.com/modules/mydownloads/ Source Code Download]<br />
*[http://www.audiocoding.com/ AudioCoding]<br />
*[http://www.rarewares.org/aac.html Binary downloads at RareWares]<br />
<br />
<br />
[[Category:Software]]<br />
[[Category:Encoder/Decoder]]</div>Gigapodhttps://wiki.hydrogenaudio.org/index.php?title=Resampling&diff=15859Resampling2006-12-11T23:27:14Z<p>Gigapod: </p>
<hr />
<div>==Resampling or Sample Rate Conversion==<br />
<br />
Digital audio is always sampled, which means that any digital audio file is created with a fixed sample rate (and resolution). Redbook Audio CD's sample rate is 44.1kHz (16-bits resolution). Audio on DVDs is sampled at 96kHz (24-bit resolution). Resampling or Sample Rate Conversion is required when one wants to convert a digital audio file (i.e. an analog audio signal that has already been digitized) from a given sample rate into a different sample rate (resolution can stay the same or change). Upsampling (aka interpolation) is the process of converting from a lower to higher sample rate (e.g. from 44.1kHz to 48kHz); downsampling (aka decimation) is the process of converting from a higher to a lower sample rate (e.g. from 96kHz to 48kHz).<br />
<br />
Resampling can also be used to describe resolution changes (changes in bit depth, e.g. from 16-bit to 24-bit, or from 24-bit to 16-bit).<br />
<br />
Low-quality resampling algorithms, whether upsampling or downsampling, can introduce artifacts which are clearly audible in the resampled audio file. A typical low-quality, but extremely fast resampling algorithm will just be based on linear interpolation. High-quality resampling algorithms use more CPU time, since they require translation to the frequency domain. Modern PC processors (~2GHz clock) can easily deal with very high-quality resampling in real time. Sound cards that do resampling in real time require a good DSP.<br />
<br />
Resampling is very often required and is in fact part of the audio mastering process for CDs, since professional audio equipment uses 96kHz or 192kHz for masters, whereas the RedBook Audio CD spec uses a 44.1kHz sample rate. Different media are recorded at different sample rates (CD at 44.1kHz, DAT at 48kHz, DVD audio at 96kHz, etc). Digitally mixing different sources sampled at different rates will require resampling to a common rate and resolution.<br />
<br />
Many PC audio cards (most notably the 10k1 and 10k2 based Creative Labs ones) and AC97 codecs can only input, output or internally process audio data at 48kHz and forcefully resample any digital audio data at one stage or another. Sometimes the audio software or the drivers will add a resampling step (e.g. ALSA when software mixing, see this thread: [http://www.hydrogenaudio.org/forums/index.php?showtopic=47591 ALSA sample rate conversion, ALSA uses a poor-quality linear interpolator]).<br />
<br />
==References==<br />
* Digital Audio Resampling Home Page: http://ccrma-www.stanford.edu/~jos/resample/<br />
* PeakPro 5 Sample Rate Converter Comparison with Other Audio Applications, Bias Inc., December 2005: http://www.bias-inc.com/products/peakPro5/resampling/peakResamplingWhitePaper.pdf<br />
* Sample Rate Conversion Comparisons (96kHz to 44.1kHz): http://src.infinitewave.ca/<br />
* iZotope 64-bit SRC Precise Sample Rate Conversion: http://www.izotope.com/tech/src/<br />
* Sox Sampling rate conversion. W. G. Unruh 2006: http://axion.physics.ubc.ca/soundcard/resample.html<br />
* An Analysis of Sample Rate Conversion in Sox, Andreas Wilde, 19. Dec. 2003: http://www.leute.server.de/wilde/resample.html<br />
* Lyons, Richard G. Understanding Digital Signal Processing. Indiana: Prentice Hall, March 2004: Edition: 3rd <code><nowiki>ISBN 0-13-108989-7</nowiki></code> <br />
* Secret Rabbit Code (aka libsamplerate): http://www.mega-nerd.com/SRC/index.html<br />
<br />
[[Category:Signal Processing]]</div>Gigapodhttps://wiki.hydrogenaudio.org/index.php?title=AAC_encoders&diff=15638AAC encoders2006-11-26T20:43:53Z<p>Gigapod: /* FAAC */</p>
<hr />
<div>These are some known [[AAC]] encoder implementations.<br />
<br />
==Nero AAC==<br />
<br />
A commercial implementation of both LC AAC and HE AAC, Nero AAC is distributed with Nero 6 which incorporates Nero Digital. Generally accepted to have the highest quality [[VBR]] LC AAC implementation (although [[iTunes]] CBR beats Nero's VBR at 128kbps). The codec also features the HE AAC standard for extremely low bitrates. The codec also allows for [[multichannel]] surround sound encoding. Unfortunately, unlike iTunes, it's not freeware, and requires the acquisition of the entire Nero 6 suite for usage of the AAC encoder alone.<br />
<br />
===Recomended Nero AAC Presets===<br />
<br />
NOTE: Once a preset has been selected, the "Encoding Quality" option should be changed to the "Fast" mode. Despite the name implying worse quality then high, a test undertaken by guruboolez shows that the "Fast" mode offers significant quality advantages over the "High" (see the test [http://www.hydrogenaudio.org/forums/index.php?showtopic=29924 here]). In the forthcoming release of Nero AAC 3.0 (or a release soon afterwards), the "Fast" mode will become the default and the high quality mode will be removed.<br />
<br />
====High Quality====<br />
<br />
: - VBR/Stereo - Streaming, 100-120 Kb/s (LC AAC) / Actual bitrate ~150kbps<br />
<br />
====Portable====<br />
<br />
: - VBR/Stereo - Internet, 90-100 Kb/s (LC AAC) / Actual bitrate ~128kbps<br />
<br />
====Small Filesize====<br />
<br />
: - VBR/Stereo - Portable, 50-70 Kb/s (HE AAC) / Actual bitrate ~90kbps<br />
<br />
<br />
The High Quality preset is for the archival of music, while the Small Filesize preset is for internet/streaming purposes.<br />
<br />
More information can be found in the [ftp://ftp6.nero.com/infosheets/Nero_Digital/db_nerodigital5.pdf Nero Digital PDF] and on the [http://www.nerodigital.com/ Nero Digital Website].<br />
<br />
<br />
==iTunes AAC==<br />
<br />
Another proprietaryl AAC implementation, [[iTunes]] AAC is known to be one of the highest quality medium-bitrate [[CBR]] LC AAC encoders.<br />
<br />
The codec is available for free through the [[iTunes]] Digital Jukebox.<br />
<br />
More information can be found at the [http://www.apple.com/mpeg4/aac/ Apple Website]<br />
<br />
The recommended high quality encoding setting is 160kbps, or 128kbps for portable use.<br />
<br />
<br />
==FAAC==<br />
<br />
[[FAAC]] is a free LC AAC encoder under the Lesser GPL license. Its quality has improved drastically over the last few years and FAAC is nowadays a viable alternative to the commercial encoders (although, at 128kbps or lower bitrates, not at the same quality level as some of them, according to Guruboolez's [http://www.hydrogenaudio.org/forums/index.php?showtopic=29924 last listening test]).<br />
<br />
The default quality setting is -q 100 -c 16000 (~120kbps average bitrate), for better quality encodings use -q 150 -c 22000 (~175kbps average bitrate).<br />
<br />
More information can be found at [http://www.audiocoding.com/ AudioCoding]<br />
<br />
==HHI/zPlane (Compaact!)==<br />
<br />
Compaact is one of the newest AAC encoders. Like Nero AAC, compaact is not free, however it does offer an impressive feature set. Roberto Amorim's last AAC test showed that at 128kbps, Compaact! is tied with both the FAAC and Coding Technologies (Real) encoders. Compaact! features both the LC and Main Object Types, [[CBR]], [[VBR]], [[Multichannel]], high resolution (24bit/96kHz) encoding, and command line support. Development on Compaact has stopped. <br />
<br />
For portable encoding, try -q5 to -q6. For music archive purposes, try -q7 to -q8.<br />
<br />
More information can be found at the [http://www.compaact.com/aacPage.php?SPRACHE=UK&PAGE=compaact Compaact website].<br />
<br />
==PsyTEL==<br />
<br />
The creation of Ivan Dimkovic (who now works on Nero AAC), PsyTEL AAC was one of the first AAC encoders. Its multichannel support has bugs that make it unusable, but its stereo mode had the best quality available in its day. Since the implementation of Nero AAC, this codec has become obsolete. It's is now outclassed by both Nero AAC and [[iTunes]] - both offer higher quality and are much faster encoders.<br />
<br />
The PsyTEL encoder can be found in the AAC section of [http://www.rjamorim.com/rrw/ ReallyRareWares]<br />
<br />
===Usability (Psytel aacenc/fastenc)===<br />
<br />
; -tape<br />
; -radio<br />
; -internet<br />
; -streaming<br />
; -normal<br />
; -extreme<br />
; -archive<br />
; -ultra<br />
<br />
For music encoding. The quality ranges from -tape (lowest [[VBR]] quality) to -ultra (highest VBR quality). Ultra is considered overkill for most audio tracks, i.e: shouldn't be used except for extremely difficult music signals. Example: aacenc -extreme -if "audio file.wav"<br />
<br />
===Encoder switches (Psytel aacenc/fastenc)===<br />
<br />
; -if <br />
: Input filename. The name of the track to be encoded (must be a [[WAV]] file)<br />
<br />
; -of <br />
: Output filename. May be omitted, because encoder will automatically create the output file name from the input file name.<br />
<br />
; -br <br />
: Bitrate switch ([[CBR]] mode). Sets the number of bits utilized per second for the encoding process. Example: aacenc -br 192 -if "audio file.wav"<br />
<br />
; -vbrhi <br />
: High quality [[VBR]] mode. Can be used with -br switch to select base BitRate. If -br is not specified, it takes as default 64kbps/channel. Example: aacenc -br 192 -vbrhi -if "audio file.wav"<br />
<br />
; -vr <br />
: Lower quality [[VBR]] mode. Recommended for internet streaming. Example: aacenc -vr -if "audio file.wav"<br />
<br />
; -c <br />
: LowPassFilter cut-off (in Hertz). Not recommended. Example: aacenc -br 128 -c 15995 -if "audio file.wav"<br />
<br />
; -qual <br />
: Encoder quality level (1 to 9). 9 is usually taken as default, but you can use smaller numbers if you need high speed and high quality isn't essential. Example: aacenc -br 192 -qual 9 -if "audio file.wav"<br />
<br />
; -adif <br />
: Use adif instead of adts (default) headers. For compatibility with some decoder software and hardware players. Example: aacenc -br 192 -adif -if "audio file.wav"<br />
<br />
; -nh <br />
: No headers (raw iso aac stream). For decoder compatibility. Example: aacenc -br 192 -nh -if "audio file.wav"<br />
<br />
; -profile "x" <br />
: Choose iso aac encoding profile:<br />
:: 0 - low complexity (default, recommended)<br />
:: 1 - main (not recommended, buggy)<br />
:: 2 - main ltp (mpeg-4 only)<br />
<br />
: Only lc profile is playable on hardware players so far. Example: aacenc -br 192 -profile 2 -if "audio file.wav"<br />
<br />
; -ihsc <br />
: Improved human speech coding. Best for human voice encoding. Not recommended for low Bitrates or [CBR] coding. Example: aacenc -vbrhi -br 192 -ihsc -if "audio file.wav"<br />
<br />
; -low_ath <br />
: Tells encoder to use highest sensitivity threshold of audibility. Not recommended on Bitrates lower than 192kbps. Example: aacenc -br 192 -low_ath -if "audio file.wav"<br />
<br />
; -pns <br />
: (perceptual noise substitution) - Improves the quality at very low Bitrates. Should be used only at 64kbps or less. Example: aacenc -br 56 -pns -if "audio file.wav"<br />
<br />
<br />
==Imagine==<br />
<br />
Imagine Technology provided an [[MPEG-4]] LC AAC plugin for [[Adobe Audition]]. This plugin provided file input and output for the MPEG-4 AAC specification, defined in ISO/IEC 14496-3. After Imagine was bought by Ingenient Technologies, they stopped marketing the Audition plugin.<br />
<br />
<br />
==Coding Technologies==<br />
<br />
Coding Technologies (CT) is a Swedish/German company that works close to FhG IIS in development and research of new audio compression techniques.<br />
<br />
Thet have distinguished themselves in development of parametric coding methods, such as [[SBR]] and Parametric Stereo. SBR is the technology behind the quality boost in MP3pro and HE AAC/AACplus.<br />
<br />
They have licensed their encoding and decoding tools to several companies - E.G, Real Networks and Magix.<br />
<br />
<br />
==FhG==<br />
<br />
[http://www.iis.fraunhofer.de/amm/techinf/aac/ Audio & Multimedia MPEG-2 AAC]<br />
<br />
<br />
==Emuzed==<br />
<br />
Emuzed develops and sells various products and technologies for the PC multimedia and embedded multimedia markets. They have ported and optimized codecs for MPEG-4 ASP and AAC LC for a chip vendor preparing to offer bundled multimedia hardware and software. More info can be found at their [http://www.emuzed.com/encoders.html encoders & decoders] page.<br />
<br />
<br />
==NEC==<br />
<br />
NEC Corporation has developed an LC AAC decoding algorithm for mobile devices. They have also developed a codec named MPEG-4 AAC Ext.1, which they claim decreases bitrate while maintaining the same audio quality. The new MPEG-4 AAC Ext.1 coding technology also features high compatibility with current MPEG-4 AAC. For more information, see [http://www.neceurope.com/release.asp?parentid=671&Area=1 NEC's press release].<br />
<br />
<br />
==Panasonic==<br />
<br />
Panasonic has developed an HE AAC codec together with NEC and Coding Technologies as described in <br />
[http://www.telos-systems.com/techtalk/hosted/m4-in-30100%20(M4IF_HE_AAC_paper).pdf this MPEG Industry Forum paper].<br />
<br />
<br />
==Real/Helix Producer==<br />
<br />
RealNetworks has incorporated Coding Technologies/FhG's MPEG-4 AAC / aacPlus™ technology and software within RealNetworks’ software products. As a result, in the newest version of RealProducer 10, AAC has replaced [[ATRAC]]3 as the high bitrate audio codec, and that software can encode AAC files wrapped in the [[MP4]] container. In addition, the Producer SDK on Windows also includes HE-AAC encoding. More info can be found at [http://www.realnetworks.com/company/press/releases/2004/codingtech.html RealNetworks' press release], as well as Coding Technologies' [http://www.codingtechnologies.com/products/aacPlus.htm aacPlus page].<br />
<br />
<br />
[[Category:Encoder/Decoder]]</div>Gigapodhttps://wiki.hydrogenaudio.org/index.php?title=AAC_encoders&diff=15637AAC encoders2006-11-26T20:35:26Z<p>Gigapod: /* FAAC */</p>
<hr />
<div>These are some known [[AAC]] encoder implementations.<br />
<br />
==Nero AAC==<br />
<br />
A commercial implementation of both LC AAC and HE AAC, Nero AAC is distributed with Nero 6 which incorporates Nero Digital. Generally accepted to have the highest quality [[VBR]] LC AAC implementation (although [[iTunes]] CBR beats Nero's VBR at 128kbps). The codec also features the HE AAC standard for extremely low bitrates. The codec also allows for [[multichannel]] surround sound encoding. Unfortunately, unlike iTunes, it's not freeware, and requires the acquisition of the entire Nero 6 suite for usage of the AAC encoder alone.<br />
<br />
===Recomended Nero AAC Presets===<br />
<br />
NOTE: Once a preset has been selected, the "Encoding Quality" option should be changed to the "Fast" mode. Despite the name implying worse quality then high, a test undertaken by guruboolez shows that the "Fast" mode offers significant quality advantages over the "High" (see the test [http://www.hydrogenaudio.org/forums/index.php?showtopic=29924 here]). In the forthcoming release of Nero AAC 3.0 (or a release soon afterwards), the "Fast" mode will become the default and the high quality mode will be removed.<br />
<br />
====High Quality====<br />
<br />
: - VBR/Stereo - Streaming, 100-120 Kb/s (LC AAC) / Actual bitrate ~150kbps<br />
<br />
====Portable====<br />
<br />
: - VBR/Stereo - Internet, 90-100 Kb/s (LC AAC) / Actual bitrate ~128kbps<br />
<br />
====Small Filesize====<br />
<br />
: - VBR/Stereo - Portable, 50-70 Kb/s (HE AAC) / Actual bitrate ~90kbps<br />
<br />
<br />
The High Quality preset is for the archival of music, while the Small Filesize preset is for internet/streaming purposes.<br />
<br />
More information can be found in the [ftp://ftp6.nero.com/infosheets/Nero_Digital/db_nerodigital5.pdf Nero Digital PDF] and on the [http://www.nerodigital.com/ Nero Digital Website].<br />
<br />
<br />
==iTunes AAC==<br />
<br />
Another proprietaryl AAC implementation, [[iTunes]] AAC is known to be one of the highest quality medium-bitrate [[CBR]] LC AAC encoders.<br />
<br />
The codec is available for free through the [[iTunes]] Digital Jukebox.<br />
<br />
More information can be found at the [http://www.apple.com/mpeg4/aac/ Apple Website]<br />
<br />
The recommended high quality encoding setting is 160kbps, or 128kbps for portable use.<br />
<br />
<br />
==FAAC==<br />
<br />
[[FAAC]] is a free LC AAC encoder under the Lesser GPL license. Its quality has improved drastically over the last few years and FAAC is nowadays a viable alternative to the commercial encoders (although, at 128kbps or lower bitrates, not at the same quality level, according to Guruboolez's [http://www.hydrogenaudio.org/forums/index.php?showtopic=29924 last listening test]).<br />
<br />
More information can be found at [http://www.audiocoding.com/ AudioCoding]<br />
<br />
==HHI/zPlane (Compaact!)==<br />
<br />
Compaact is one of the newest AAC encoders. Like Nero AAC, compaact is not free, however it does offer an impressive feature set. Roberto Amorim's last AAC test showed that at 128kbps, Compaact! is tied with both the FAAC and Coding Technologies (Real) encoders. Compaact! features both the LC and Main Object Types, [[CBR]], [[VBR]], [[Multichannel]], high resolution (24bit/96kHz) encoding, and command line support. Development on Compaact has stopped. <br />
<br />
For portable encoding, try -q5 to -q6. For music archive purposes, try -q7 to -q8.<br />
<br />
More information can be found at the [http://www.compaact.com/aacPage.php?SPRACHE=UK&PAGE=compaact Compaact website].<br />
<br />
==PsyTEL==<br />
<br />
The creation of Ivan Dimkovic (who now works on Nero AAC), PsyTEL AAC was one of the first AAC encoders. Its multichannel support has bugs that make it unusable, but its stereo mode had the best quality available in its day. Since the implementation of Nero AAC, this codec has become obsolete. It's is now outclassed by both Nero AAC and [[iTunes]] - both offer higher quality and are much faster encoders.<br />
<br />
The PsyTEL encoder can be found in the AAC section of [http://www.rjamorim.com/rrw/ ReallyRareWares]<br />
<br />
===Usability (Psytel aacenc/fastenc)===<br />
<br />
; -tape<br />
; -radio<br />
; -internet<br />
; -streaming<br />
; -normal<br />
; -extreme<br />
; -archive<br />
; -ultra<br />
<br />
For music encoding. The quality ranges from -tape (lowest [[VBR]] quality) to -ultra (highest VBR quality). Ultra is considered overkill for most audio tracks, i.e: shouldn't be used except for extremely difficult music signals. Example: aacenc -extreme -if "audio file.wav"<br />
<br />
===Encoder switches (Psytel aacenc/fastenc)===<br />
<br />
; -if <br />
: Input filename. The name of the track to be encoded (must be a [[WAV]] file)<br />
<br />
; -of <br />
: Output filename. May be omitted, because encoder will automatically create the output file name from the input file name.<br />
<br />
; -br <br />
: Bitrate switch ([[CBR]] mode). Sets the number of bits utilized per second for the encoding process. Example: aacenc -br 192 -if "audio file.wav"<br />
<br />
; -vbrhi <br />
: High quality [[VBR]] mode. Can be used with -br switch to select base BitRate. If -br is not specified, it takes as default 64kbps/channel. Example: aacenc -br 192 -vbrhi -if "audio file.wav"<br />
<br />
; -vr <br />
: Lower quality [[VBR]] mode. Recommended for internet streaming. Example: aacenc -vr -if "audio file.wav"<br />
<br />
; -c <br />
: LowPassFilter cut-off (in Hertz). Not recommended. Example: aacenc -br 128 -c 15995 -if "audio file.wav"<br />
<br />
; -qual <br />
: Encoder quality level (1 to 9). 9 is usually taken as default, but you can use smaller numbers if you need high speed and high quality isn't essential. Example: aacenc -br 192 -qual 9 -if "audio file.wav"<br />
<br />
; -adif <br />
: Use adif instead of adts (default) headers. For compatibility with some decoder software and hardware players. Example: aacenc -br 192 -adif -if "audio file.wav"<br />
<br />
; -nh <br />
: No headers (raw iso aac stream). For decoder compatibility. Example: aacenc -br 192 -nh -if "audio file.wav"<br />
<br />
; -profile "x" <br />
: Choose iso aac encoding profile:<br />
:: 0 - low complexity (default, recommended)<br />
:: 1 - main (not recommended, buggy)<br />
:: 2 - main ltp (mpeg-4 only)<br />
<br />
: Only lc profile is playable on hardware players so far. Example: aacenc -br 192 -profile 2 -if "audio file.wav"<br />
<br />
; -ihsc <br />
: Improved human speech coding. Best for human voice encoding. Not recommended for low Bitrates or [CBR] coding. Example: aacenc -vbrhi -br 192 -ihsc -if "audio file.wav"<br />
<br />
; -low_ath <br />
: Tells encoder to use highest sensitivity threshold of audibility. Not recommended on Bitrates lower than 192kbps. Example: aacenc -br 192 -low_ath -if "audio file.wav"<br />
<br />
; -pns <br />
: (perceptual noise substitution) - Improves the quality at very low Bitrates. Should be used only at 64kbps or less. Example: aacenc -br 56 -pns -if "audio file.wav"<br />
<br />
<br />
==Imagine==<br />
<br />
Imagine Technology provided an [[MPEG-4]] LC AAC plugin for [[Adobe Audition]]. This plugin provided file input and output for the MPEG-4 AAC specification, defined in ISO/IEC 14496-3. After Imagine was bought by Ingenient Technologies, they stopped marketing the Audition plugin.<br />
<br />
<br />
==Coding Technologies==<br />
<br />
Coding Technologies (CT) is a Swedish/German company that works close to FhG IIS in development and research of new audio compression techniques.<br />
<br />
Thet have distinguished themselves in development of parametric coding methods, such as [[SBR]] and Parametric Stereo. SBR is the technology behind the quality boost in MP3pro and HE AAC/AACplus.<br />
<br />
They have licensed their encoding and decoding tools to several companies - E.G, Real Networks and Magix.<br />
<br />
<br />
==FhG==<br />
<br />
[http://www.iis.fraunhofer.de/amm/techinf/aac/ Audio & Multimedia MPEG-2 AAC]<br />
<br />
<br />
==Emuzed==<br />
<br />
Emuzed develops and sells various products and technologies for the PC multimedia and embedded multimedia markets. They have ported and optimized codecs for MPEG-4 ASP and AAC LC for a chip vendor preparing to offer bundled multimedia hardware and software. More info can be found at their [http://www.emuzed.com/encoders.html encoders & decoders] page.<br />
<br />
<br />
==NEC==<br />
<br />
NEC Corporation has developed an LC AAC decoding algorithm for mobile devices. They have also developed a codec named MPEG-4 AAC Ext.1, which they claim decreases bitrate while maintaining the same audio quality. The new MPEG-4 AAC Ext.1 coding technology also features high compatibility with current MPEG-4 AAC. For more information, see [http://www.neceurope.com/release.asp?parentid=671&Area=1 NEC's press release].<br />
<br />
<br />
==Panasonic==<br />
<br />
Panasonic has developed an HE AAC codec together with NEC and Coding Technologies as described in <br />
[http://www.telos-systems.com/techtalk/hosted/m4-in-30100%20(M4IF_HE_AAC_paper).pdf this MPEG Industry Forum paper].<br />
<br />
<br />
==Real/Helix Producer==<br />
<br />
RealNetworks has incorporated Coding Technologies/FhG's MPEG-4 AAC / aacPlus™ technology and software within RealNetworks’ software products. As a result, in the newest version of RealProducer 10, AAC has replaced [[ATRAC]]3 as the high bitrate audio codec, and that software can encode AAC files wrapped in the [[MP4]] container. In addition, the Producer SDK on Windows also includes HE-AAC encoding. More info can be found at [http://www.realnetworks.com/company/press/releases/2004/codingtech.html RealNetworks' press release], as well as Coding Technologies' [http://www.codingtechnologies.com/products/aacPlus.htm aacPlus page].<br />
<br />
<br />
[[Category:Encoder/Decoder]]</div>Gigapodhttps://wiki.hydrogenaudio.org/index.php?title=AAC_encoders&diff=15636AAC encoders2006-11-26T20:31:25Z<p>Gigapod: /* FAAC */</p>
<hr />
<div>These are some known [[AAC]] encoder implementations.<br />
<br />
==Nero AAC==<br />
<br />
A commercial implementation of both LC AAC and HE AAC, Nero AAC is distributed with Nero 6 which incorporates Nero Digital. Generally accepted to have the highest quality [[VBR]] LC AAC implementation (although [[iTunes]] CBR beats Nero's VBR at 128kbps). The codec also features the HE AAC standard for extremely low bitrates. The codec also allows for [[multichannel]] surround sound encoding. Unfortunately, unlike iTunes, it's not freeware, and requires the acquisition of the entire Nero 6 suite for usage of the AAC encoder alone.<br />
<br />
===Recomended Nero AAC Presets===<br />
<br />
NOTE: Once a preset has been selected, the "Encoding Quality" option should be changed to the "Fast" mode. Despite the name implying worse quality then high, a test undertaken by guruboolez shows that the "Fast" mode offers significant quality advantages over the "High" (see the test [http://www.hydrogenaudio.org/forums/index.php?showtopic=29924 here]). In the forthcoming release of Nero AAC 3.0 (or a release soon afterwards), the "Fast" mode will become the default and the high quality mode will be removed.<br />
<br />
====High Quality====<br />
<br />
: - VBR/Stereo - Streaming, 100-120 Kb/s (LC AAC) / Actual bitrate ~150kbps<br />
<br />
====Portable====<br />
<br />
: - VBR/Stereo - Internet, 90-100 Kb/s (LC AAC) / Actual bitrate ~128kbps<br />
<br />
====Small Filesize====<br />
<br />
: - VBR/Stereo - Portable, 50-70 Kb/s (HE AAC) / Actual bitrate ~90kbps<br />
<br />
<br />
The High Quality preset is for the archival of music, while the Small Filesize preset is for internet/streaming purposes.<br />
<br />
More information can be found in the [ftp://ftp6.nero.com/infosheets/Nero_Digital/db_nerodigital5.pdf Nero Digital PDF] and on the [http://www.nerodigital.com/ Nero Digital Website].<br />
<br />
<br />
==iTunes AAC==<br />
<br />
Another proprietaryl AAC implementation, [[iTunes]] AAC is known to be one of the highest quality medium-bitrate [[CBR]] LC AAC encoders.<br />
<br />
The codec is available for free through the [[iTunes]] Digital Jukebox.<br />
<br />
More information can be found at the [http://www.apple.com/mpeg4/aac/ Apple Website]<br />
<br />
The recommended high quality encoding setting is 160kbps, or 128kbps for portable use.<br />
<br />
<br />
==FAAC==<br />
<br />
[[FAAC]] is a free LC AAC encoder under the Lesser GPL license. Its quality has improved drastically over the last few years and FAAC is nowadays a viable alternative to the commercial encoders (although, at 128kbps or lower bitrates, not as good as some commercial encoders, according to Guruboolez listening tests).<br />
<br />
More information can be found at [http://www.audiocoding.com/ AudioCoding]<br />
<br />
==HHI/zPlane (Compaact!)==<br />
<br />
Compaact is one of the newest AAC encoders. Like Nero AAC, compaact is not free, however it does offer an impressive feature set. Roberto Amorim's last AAC test showed that at 128kbps, Compaact! is tied with both the FAAC and Coding Technologies (Real) encoders. Compaact! features both the LC and Main Object Types, [[CBR]], [[VBR]], [[Multichannel]], high resolution (24bit/96kHz) encoding, and command line support. Development on Compaact has stopped. <br />
<br />
For portable encoding, try -q5 to -q6. For music archive purposes, try -q7 to -q8.<br />
<br />
More information can be found at the [http://www.compaact.com/aacPage.php?SPRACHE=UK&PAGE=compaact Compaact website].<br />
<br />
==PsyTEL==<br />
<br />
The creation of Ivan Dimkovic (who now works on Nero AAC), PsyTEL AAC was one of the first AAC encoders. Its multichannel support has bugs that make it unusable, but its stereo mode had the best quality available in its day. Since the implementation of Nero AAC, this codec has become obsolete. It's is now outclassed by both Nero AAC and [[iTunes]] - both offer higher quality and are much faster encoders.<br />
<br />
The PsyTEL encoder can be found in the AAC section of [http://www.rjamorim.com/rrw/ ReallyRareWares]<br />
<br />
===Usability (Psytel aacenc/fastenc)===<br />
<br />
; -tape<br />
; -radio<br />
; -internet<br />
; -streaming<br />
; -normal<br />
; -extreme<br />
; -archive<br />
; -ultra<br />
<br />
For music encoding. The quality ranges from -tape (lowest [[VBR]] quality) to -ultra (highest VBR quality). Ultra is considered overkill for most audio tracks, i.e: shouldn't be used except for extremely difficult music signals. Example: aacenc -extreme -if "audio file.wav"<br />
<br />
===Encoder switches (Psytel aacenc/fastenc)===<br />
<br />
; -if <br />
: Input filename. The name of the track to be encoded (must be a [[WAV]] file)<br />
<br />
; -of <br />
: Output filename. May be omitted, because encoder will automatically create the output file name from the input file name.<br />
<br />
; -br <br />
: Bitrate switch ([[CBR]] mode). Sets the number of bits utilized per second for the encoding process. Example: aacenc -br 192 -if "audio file.wav"<br />
<br />
; -vbrhi <br />
: High quality [[VBR]] mode. Can be used with -br switch to select base BitRate. If -br is not specified, it takes as default 64kbps/channel. Example: aacenc -br 192 -vbrhi -if "audio file.wav"<br />
<br />
; -vr <br />
: Lower quality [[VBR]] mode. Recommended for internet streaming. Example: aacenc -vr -if "audio file.wav"<br />
<br />
; -c <br />
: LowPassFilter cut-off (in Hertz). Not recommended. Example: aacenc -br 128 -c 15995 -if "audio file.wav"<br />
<br />
; -qual <br />
: Encoder quality level (1 to 9). 9 is usually taken as default, but you can use smaller numbers if you need high speed and high quality isn't essential. Example: aacenc -br 192 -qual 9 -if "audio file.wav"<br />
<br />
; -adif <br />
: Use adif instead of adts (default) headers. For compatibility with some decoder software and hardware players. Example: aacenc -br 192 -adif -if "audio file.wav"<br />
<br />
; -nh <br />
: No headers (raw iso aac stream). For decoder compatibility. Example: aacenc -br 192 -nh -if "audio file.wav"<br />
<br />
; -profile "x" <br />
: Choose iso aac encoding profile:<br />
:: 0 - low complexity (default, recommended)<br />
:: 1 - main (not recommended, buggy)<br />
:: 2 - main ltp (mpeg-4 only)<br />
<br />
: Only lc profile is playable on hardware players so far. Example: aacenc -br 192 -profile 2 -if "audio file.wav"<br />
<br />
; -ihsc <br />
: Improved human speech coding. Best for human voice encoding. Not recommended for low Bitrates or [CBR] coding. Example: aacenc -vbrhi -br 192 -ihsc -if "audio file.wav"<br />
<br />
; -low_ath <br />
: Tells encoder to use highest sensitivity threshold of audibility. Not recommended on Bitrates lower than 192kbps. Example: aacenc -br 192 -low_ath -if "audio file.wav"<br />
<br />
; -pns <br />
: (perceptual noise substitution) - Improves the quality at very low Bitrates. Should be used only at 64kbps or less. Example: aacenc -br 56 -pns -if "audio file.wav"<br />
<br />
<br />
==Imagine==<br />
<br />
Imagine Technology provided an [[MPEG-4]] LC AAC plugin for [[Adobe Audition]]. This plugin provided file input and output for the MPEG-4 AAC specification, defined in ISO/IEC 14496-3. After Imagine was bought by Ingenient Technologies, they stopped marketing the Audition plugin.<br />
<br />
<br />
==Coding Technologies==<br />
<br />
Coding Technologies (CT) is a Swedish/German company that works close to FhG IIS in development and research of new audio compression techniques.<br />
<br />
Thet have distinguished themselves in development of parametric coding methods, such as [[SBR]] and Parametric Stereo. SBR is the technology behind the quality boost in MP3pro and HE AAC/AACplus.<br />
<br />
They have licensed their encoding and decoding tools to several companies - E.G, Real Networks and Magix.<br />
<br />
<br />
==FhG==<br />
<br />
[http://www.iis.fraunhofer.de/amm/techinf/aac/ Audio & Multimedia MPEG-2 AAC]<br />
<br />
<br />
==Emuzed==<br />
<br />
Emuzed develops and sells various products and technologies for the PC multimedia and embedded multimedia markets. They have ported and optimized codecs for MPEG-4 ASP and AAC LC for a chip vendor preparing to offer bundled multimedia hardware and software. More info can be found at their [http://www.emuzed.com/encoders.html encoders & decoders] page.<br />
<br />
<br />
==NEC==<br />
<br />
NEC Corporation has developed an LC AAC decoding algorithm for mobile devices. They have also developed a codec named MPEG-4 AAC Ext.1, which they claim decreases bitrate while maintaining the same audio quality. The new MPEG-4 AAC Ext.1 coding technology also features high compatibility with current MPEG-4 AAC. For more information, see [http://www.neceurope.com/release.asp?parentid=671&Area=1 NEC's press release].<br />
<br />
<br />
==Panasonic==<br />
<br />
Panasonic has developed an HE AAC codec together with NEC and Coding Technologies as described in <br />
[http://www.telos-systems.com/techtalk/hosted/m4-in-30100%20(M4IF_HE_AAC_paper).pdf this MPEG Industry Forum paper].<br />
<br />
<br />
==Real/Helix Producer==<br />
<br />
RealNetworks has incorporated Coding Technologies/FhG's MPEG-4 AAC / aacPlus™ technology and software within RealNetworks’ software products. As a result, in the newest version of RealProducer 10, AAC has replaced [[ATRAC]]3 as the high bitrate audio codec, and that software can encode AAC files wrapped in the [[MP4]] container. In addition, the Producer SDK on Windows also includes HE-AAC encoding. More info can be found at [http://www.realnetworks.com/company/press/releases/2004/codingtech.html RealNetworks' press release], as well as Coding Technologies' [http://www.codingtechnologies.com/products/aacPlus.htm aacPlus page].<br />
<br />
<br />
[[Category:Encoder/Decoder]]</div>Gigapodhttps://wiki.hydrogenaudio.org/index.php?title=AAC_FAQ&diff=15635AAC FAQ2006-11-26T20:23:46Z<p>Gigapod: /* What hardware players can play back AAC music? */</p>
<hr />
<div>===Great, so you've given me all the technical stuff, but what is [[AAC]] really?===<br />
[[AAC]] is the culmination of the current state of the art audio encoding techniques. It is designed <br />
to improve upon and replace [[MP3]] as the defacto Audio Encoding standard. It usually offers (depending on the codec) equivalent quality to MP3 at a lower bitrate.<br />
<br />
===What is the difference between *.MP4 and *.M4A?===<br />
Besides the extension, absolutely nothing. Apple came up with extension to distiguish between files with Video and Audio (the [[MP4]] extension) and files with Audio only (the M4A extension). As far as the internal structure of the file, nothing is different.<br />
<br />
===What MPEG 4 extensions does the Apple iPod Accept?===<br />
The iPod accepts files with the MP4 extension, the M4A extension, the M4P extension (a Protected AAC file), and the M4B extension for audiobook files (which can be either protected or unprotected). It will not accept unwrapped AAC files (files with the .AAC extension).<br />
<br />
===What is the difference between LC (Low Complexity) and HE (High Efficiency)?===<br />
These are two of the various Object Types in the MPEG4 Systems Standard. LC is the most popular Object Type with all encoders/decoders supporting it. Currently, Nero, Coding Technolgies, and Panasonic have incorporated the HE AAC standard into their encoders, which allows for higher quality sound at lower bitrates then the LC Object Type does (at the same bitrate). The HE Object Type is only used for music with a bitrate of less than ~80kbps.<br />
<br />
===What's the best AAC encoder?===<br />
There is no best AAC encoder as such, because the quality of an encoding depends not only on the encoder implementation, but also on bitrates, sometimes on the specific audio sample being encoded, and finally, on the subjective perceptual judgement of the listener at playback time (which is also influenced by the equipment being used for playback).<br />
<br />
Since it is very difficult to quantify the quality of an encoder, [[Listening Tests|listening tests]] are used.<br />
<br />
Guruboolez's [http://www.hydrogenaudio.org/forums/index.php?showtopic=29924 last test] concluded that [http://www.nero.com/en/ Nero AAC] was the best AAC encoder, at 128kbps, on classical samples, at the time the test was conducted.<br />
<br />
On the other hand, a public [http://www.rjamorim.com/test/multiformat128/results.html listening test] conducted by [[User:rjamorim|rjamorim]] in mid-2004 comparing different codecs, at 128kbps, with several music styles and featuring several listeners concluded that [[iTunes]] (the only AAC codec included in the test) was better than other codecs - even VBR-enabled ones. <br />
<br />
The quality of any encoder is not linear with bitrates, and therefore these results can not be extrapolated to other higher or lower bitrates. It can also be said with great confidence that both the iTunes AAC encoder and the Nero AAC encoder, although still under development at the end of 2006, are relatively 'mature' and should not fail badly (result in any obvious artifacts) on any particular sample at an average bitrate of 128kbps (i.e. Internet Profile for Nero AAC) or above (based on Roberto's listening tests, see bottom).<br />
<br />
Beyond that, only you can decide; you may want to conduct your own private [[Listening Tests|listening tests]], or you may base your decision on other criteria besides audio quality. See the [[Audio format guide]] for more information.<br />
<br />
===Do AAC encoded files play back gaplessly?===<br />
[[Gapless]] playback is not part of the AAC standard and as such is not mandatory. However, certain companies can choose to add gapless encoding/decoding if they desire, providing it doesn't break compatibility with previous decoders. This is what Ahead have done with their Nero AAC codec. The files get encoded with information that allows the gap heard between files to be removed. This however is only possible with supported players (currently these include foobar2000 and Nero ShowTime). Currently Nero AAC and FAAC are the only encoders to have gapless encoding/decoding support.<br />
<br />
===What software players can play back AAC music?===<br />
There are now a number of software players that can play back this new format. [[foobar2000]] is considered by many to be a very high quality audio player, and it is certainly capable of playing back AAC encoded files. Other players include [http://amarok.kde.org Amarok] using [http://www.audiocoding.com/ libfaad2], Apple's [[iTunes]], [[Winamp]], [http://www.real.com/ Real Player] and [http://www.microsoft.com/windows/windowsmedia/default.aspx Windows Media Player] using the [http://corecodec.org/projects/coreaac CoreAAC filter] and [http://www.elecard.com/download/ Moonlight MP4 Demultiplexer]. Also for Directshow-based applications playback and encoding is possible using the commercial [http://www.3ivx.com/ 3ivx filter suite].<br />
<br />
===What hardware players can play back AAC music?===<br />
There are also a few hardware players that can play back AAC audio. The most famous of these is the [[Apple iPod]] series of products, all of which feature AAC playback. A number of mobile (cell) phones also support unwrapped AAC (AAC not contained in the MP4 container). Recent Pioneer HT receivers can play back AAC files on a USB key or other USB mass-storage device.<br />
<br />
==Related Links==<br />
<br />
* [[AAC]] description article<br />
* Known [[AAC implementations]].<br />
* Read the [[AAC guide]] to learn how to obtain AAC/[[MP4]] files out of [[WAV]] files and CDs.<br />
* Detailed AAC comparisons can be found at [http://www.rjamorim.com/test/ Roberto's listening tests page].<br />
<br />
<br />
[[Category:Technical]]<br />
[[Category:Codecs]]</div>Gigapodhttps://wiki.hydrogenaudio.org/index.php?title=AAC_FAQ&diff=15634AAC FAQ2006-11-26T20:15:11Z<p>Gigapod: /* What's the best AAC encoder? */</p>
<hr />
<div>===Great, so you've given me all the technical stuff, but what is [[AAC]] really?===<br />
[[AAC]] is the culmination of the current state of the art audio encoding techniques. It is designed <br />
to improve upon and replace [[MP3]] as the defacto Audio Encoding standard. It usually offers (depending on the codec) equivalent quality to MP3 at a lower bitrate.<br />
<br />
===What is the difference between *.MP4 and *.M4A?===<br />
Besides the extension, absolutely nothing. Apple came up with extension to distiguish between files with Video and Audio (the [[MP4]] extension) and files with Audio only (the M4A extension). As far as the internal structure of the file, nothing is different.<br />
<br />
===What MPEG 4 extensions does the Apple iPod Accept?===<br />
The iPod accepts files with the MP4 extension, the M4A extension, the M4P extension (a Protected AAC file), and the M4B extension for audiobook files (which can be either protected or unprotected). It will not accept unwrapped AAC files (files with the .AAC extension).<br />
<br />
===What is the difference between LC (Low Complexity) and HE (High Efficiency)?===<br />
These are two of the various Object Types in the MPEG4 Systems Standard. LC is the most popular Object Type with all encoders/decoders supporting it. Currently, Nero, Coding Technolgies, and Panasonic have incorporated the HE AAC standard into their encoders, which allows for higher quality sound at lower bitrates then the LC Object Type does (at the same bitrate). The HE Object Type is only used for music with a bitrate of less than ~80kbps.<br />
<br />
===What's the best AAC encoder?===<br />
There is no best AAC encoder as such, because the quality of an encoding depends not only on the encoder implementation, but also on bitrates, sometimes on the specific audio sample being encoded, and finally, on the subjective perceptual judgement of the listener at playback time (which is also influenced by the equipment being used for playback).<br />
<br />
Since it is very difficult to quantify the quality of an encoder, [[Listening Tests|listening tests]] are used.<br />
<br />
Guruboolez's [http://www.hydrogenaudio.org/forums/index.php?showtopic=29924 last test] concluded that [http://www.nero.com/en/ Nero AAC] was the best AAC encoder, at 128kbps, on classical samples, at the time the test was conducted.<br />
<br />
On the other hand, a public [http://www.rjamorim.com/test/multiformat128/results.html listening test] conducted by [[User:rjamorim|rjamorim]] in mid-2004 comparing different codecs, at 128kbps, with several music styles and featuring several listeners concluded that [[iTunes]] (the only AAC codec included in the test) was better than other codecs - even VBR-enabled ones. <br />
<br />
The quality of any encoder is not linear with bitrates, and therefore these results can not be extrapolated to other higher or lower bitrates. It can also be said with great confidence that both the iTunes AAC encoder and the Nero AAC encoder, although still under development at the end of 2006, are relatively 'mature' and should not fail badly (result in any obvious artifacts) on any particular sample at an average bitrate of 128kbps (i.e. Internet Profile for Nero AAC) or above (based on Roberto's listening tests, see bottom).<br />
<br />
Beyond that, only you can decide; you may want to conduct your own private [[Listening Tests|listening tests]], or you may base your decision on other criteria besides audio quality. See the [[Audio format guide]] for more information.<br />
<br />
===Do AAC encoded files play back gaplessly?===<br />
[[Gapless]] playback is not part of the AAC standard and as such is not mandatory. However, certain companies can choose to add gapless encoding/decoding if they desire, providing it doesn't break compatibility with previous decoders. This is what Ahead have done with their Nero AAC codec. The files get encoded with information that allows the gap heard between files to be removed. This however is only possible with supported players (currently these include foobar2000 and Nero ShowTime). Currently Nero AAC and FAAC are the only encoders to have gapless encoding/decoding support.<br />
<br />
===What software players can play back AAC music?===<br />
There are now a number of software players that can play back this new format. [[foobar2000]] is considered by many to be a very high quality audio player, and it is certainly capable of playing back AAC encoded files. Other players include [http://amarok.kde.org Amarok] using [http://www.audiocoding.com/ libfaad2], Apple's [[iTunes]], [[Winamp]], [http://www.real.com/ Real Player] and [http://www.microsoft.com/windows/windowsmedia/default.aspx Windows Media Player] using the [http://corecodec.org/projects/coreaac CoreAAC filter] and [http://www.elecard.com/download/ Moonlight MP4 Demultiplexer]. Also for Directshow-based applications playback and encoding is possible using the commercial [http://www.3ivx.com/ 3ivx filter suite].<br />
<br />
===What hardware players can play back AAC music?===<br />
There are also a few hardware players that can play back AAC audio. The most famous of these is the [[Apple iPod]] series of products, all of which feature AAC playback. A number of mobile (cell) phones also support unwrapped AAC (AAC not contained in the MP4 container). Recent Pioneer HT receivers can play back AAC files on a USB key or other mass-storage device.<br />
<br />
==Related Links==<br />
<br />
* [[AAC]] description article<br />
* Known [[AAC implementations]].<br />
* Read the [[AAC guide]] to learn how to obtain AAC/[[MP4]] files out of [[WAV]] files and CDs.<br />
* Detailed AAC comparisons can be found at [http://www.rjamorim.com/test/ Roberto's listening tests page].<br />
<br />
<br />
[[Category:Technical]]<br />
[[Category:Codecs]]</div>Gigapodhttps://wiki.hydrogenaudio.org/index.php?title=AAC_FAQ&diff=15633AAC FAQ2006-11-26T20:14:02Z<p>Gigapod: /* What's the best AAC encoder? */</p>
<hr />
<div>===Great, so you've given me all the technical stuff, but what is [[AAC]] really?===<br />
[[AAC]] is the culmination of the current state of the art audio encoding techniques. It is designed <br />
to improve upon and replace [[MP3]] as the defacto Audio Encoding standard. It usually offers (depending on the codec) equivalent quality to MP3 at a lower bitrate.<br />
<br />
===What is the difference between *.MP4 and *.M4A?===<br />
Besides the extension, absolutely nothing. Apple came up with extension to distiguish between files with Video and Audio (the [[MP4]] extension) and files with Audio only (the M4A extension). As far as the internal structure of the file, nothing is different.<br />
<br />
===What MPEG 4 extensions does the Apple iPod Accept?===<br />
The iPod accepts files with the MP4 extension, the M4A extension, the M4P extension (a Protected AAC file), and the M4B extension for audiobook files (which can be either protected or unprotected). It will not accept unwrapped AAC files (files with the .AAC extension).<br />
<br />
===What is the difference between LC (Low Complexity) and HE (High Efficiency)?===<br />
These are two of the various Object Types in the MPEG4 Systems Standard. LC is the most popular Object Type with all encoders/decoders supporting it. Currently, Nero, Coding Technolgies, and Panasonic have incorporated the HE AAC standard into their encoders, which allows for higher quality sound at lower bitrates then the LC Object Type does (at the same bitrate). The HE Object Type is only used for music with a bitrate of less than ~80kbps.<br />
<br />
===What's the best AAC encoder?===<br />
There is no best AAC encoder as such, because the quality of an encoding depends not only on the encoder implementation, but also on bitrates, sometimes on the specific audio sample being encoded, and finally, on the subjective perceptual judgement of the listener at playback time (which is also influenced by the equipment being used for playback).<br />
<br />
Since it is very difficult to quantify the quality of an encoder, [[Listening Tests|listening tests]] are used.<br />
<br />
Guruboolez's [http://www.hydrogenaudio.org/forums/index.php?showtopic=29924 last test] concluded that [http://www.nero.com/en/ Nero AAC] was the best AAC encoder, at 128kbps, on classical samples, at the time the test was conducted.<br />
<br />
On the other hand, a public [http://www.rjamorim.com/test/multiformat128/results.html listening test] conducted by [[User:rjamorim|rjamorim]] in mid-2004 comparing different codecs, at 128kbps, with several music styles and featuring several listeners concluded that [[iTunes]] (the only AAC codec included in the test) was better than other codecs - even VBR-enabled ones. <br />
<br />
The quality of any encoder is not linear with bitrates, and therefore these results can not be extrapolated to other higher or lower bitrates. It can also be said with great confidence that both the iTunes AAC encoder and the Nero AAC encoder, although still under development at the end of 2006, are relatively 'mature' and should not fail badly (result in any obvious artifacts) on any particular sample at an average bitrate of 128kbps (i.e. Internet Profile for Nero AAC) or above (based on Roberto's listening tests, see bottom).<br />
<br />
Beyond that, only you can decide; you may want to conduct your own private [[Listening Tests|listening tests]], or you may base yourself on other parameters besides audio quality. See the [[Audio format guide]] for more information.<br />
<br />
===Do AAC encoded files play back gaplessly?===<br />
[[Gapless]] playback is not part of the AAC standard and as such is not mandatory. However, certain companies can choose to add gapless encoding/decoding if they desire, providing it doesn't break compatibility with previous decoders. This is what Ahead have done with their Nero AAC codec. The files get encoded with information that allows the gap heard between files to be removed. This however is only possible with supported players (currently these include foobar2000 and Nero ShowTime). Currently Nero AAC and FAAC are the only encoders to have gapless encoding/decoding support.<br />
<br />
===What software players can play back AAC music?===<br />
There are now a number of software players that can play back this new format. [[foobar2000]] is considered by many to be a very high quality audio player, and it is certainly capable of playing back AAC encoded files. Other players include [http://amarok.kde.org Amarok] using [http://www.audiocoding.com/ libfaad2], Apple's [[iTunes]], [[Winamp]], [http://www.real.com/ Real Player] and [http://www.microsoft.com/windows/windowsmedia/default.aspx Windows Media Player] using the [http://corecodec.org/projects/coreaac CoreAAC filter] and [http://www.elecard.com/download/ Moonlight MP4 Demultiplexer]. Also for Directshow-based applications playback and encoding is possible using the commercial [http://www.3ivx.com/ 3ivx filter suite].<br />
<br />
===What hardware players can play back AAC music?===<br />
There are also a few hardware players that can play back AAC audio. The most famous of these is the [[Apple iPod]] series of products, all of which feature AAC playback. A number of mobile (cell) phones also support unwrapped AAC (AAC not contained in the MP4 container). Recent Pioneer HT receivers can play back AAC files on a USB key or other mass-storage device.<br />
<br />
==Related Links==<br />
<br />
* [[AAC]] description article<br />
* Known [[AAC implementations]].<br />
* Read the [[AAC guide]] to learn how to obtain AAC/[[MP4]] files out of [[WAV]] files and CDs.<br />
* Detailed AAC comparisons can be found at [http://www.rjamorim.com/test/ Roberto's listening tests page].<br />
<br />
<br />
[[Category:Technical]]<br />
[[Category:Codecs]]</div>Gigapodhttps://wiki.hydrogenaudio.org/index.php?title=AAC_FAQ&diff=15632AAC FAQ2006-11-26T20:12:53Z<p>Gigapod: /* What's the best AAC encoder? */</p>
<hr />
<div>===Great, so you've given me all the technical stuff, but what is [[AAC]] really?===<br />
[[AAC]] is the culmination of the current state of the art audio encoding techniques. It is designed <br />
to improve upon and replace [[MP3]] as the defacto Audio Encoding standard. It usually offers (depending on the codec) equivalent quality to MP3 at a lower bitrate.<br />
<br />
===What is the difference between *.MP4 and *.M4A?===<br />
Besides the extension, absolutely nothing. Apple came up with extension to distiguish between files with Video and Audio (the [[MP4]] extension) and files with Audio only (the M4A extension). As far as the internal structure of the file, nothing is different.<br />
<br />
===What MPEG 4 extensions does the Apple iPod Accept?===<br />
The iPod accepts files with the MP4 extension, the M4A extension, the M4P extension (a Protected AAC file), and the M4B extension for audiobook files (which can be either protected or unprotected). It will not accept unwrapped AAC files (files with the .AAC extension).<br />
<br />
===What is the difference between LC (Low Complexity) and HE (High Efficiency)?===<br />
These are two of the various Object Types in the MPEG4 Systems Standard. LC is the most popular Object Type with all encoders/decoders supporting it. Currently, Nero, Coding Technolgies, and Panasonic have incorporated the HE AAC standard into their encoders, which allows for higher quality sound at lower bitrates then the LC Object Type does (at the same bitrate). The HE Object Type is only used for music with a bitrate of less than ~80kbps.<br />
<br />
===What's the best AAC encoder?===<br />
There is no best AAC encoder as such, because the quality of an encoding depends not only on the encoder implementation, but also on bitrates, sometimes on the specific audio sample being encoded, and finally, on the subjective perceptual judgement of the listener at playback time (which is also influenced by the equipment being used for playback).<br />
<br />
Since it is very difficult to quantify the quality of an encoder, [[Listening Tests] listening tests] are used.<br />
<br />
Guruboolez's [http://www.hydrogenaudio.org/forums/index.php?showtopic=29924 last test] concluded that [http://www.nero.com/en/ Nero AAC] was the best AAC encoder, at 128kbps, on classical samples, at the time the test was conducted.<br />
<br />
On the other hand, a public [http://www.rjamorim.com/test/multiformat128/results.html listening test] conducted by [[User:rjamorim|rjamorim]] in mid-2004 comparing different codecs, at 128kbps, with several music styles and featuring several listeners concluded that [[iTunes]] (the only AAC codec included in the test) was better than other codecs - even VBR-enabled ones. <br />
<br />
The quality of any encoder is not linear with bitrates, and therefore these results can not be extrapolated to other higher or lower bitrates. It can also be said with great confidence that both the iTunes AAC encoder and the Nero AAC encoder, although still under development at the end of 2006, are relatively 'mature' and should not fail badly (result in any obvious artifacts) on any particular sample at an average bitrate of 128kbps (i.e. Internet Profile for Nero AAC) or above (based on Roberto's listening tests, see bottom).<br />
<br />
Beyond that, only you can decide; you may want to conduct your own private [[Listening Tests] listening tests], or you may base yourself on other parameters besides audio quality. See the [[Audio format guide]] for more information.<br />
<br />
===Do AAC encoded files play back gaplessly?===<br />
[[Gapless]] playback is not part of the AAC standard and as such is not mandatory. However, certain companies can choose to add gapless encoding/decoding if they desire, providing it doesn't break compatibility with previous decoders. This is what Ahead have done with their Nero AAC codec. The files get encoded with information that allows the gap heard between files to be removed. This however is only possible with supported players (currently these include foobar2000 and Nero ShowTime). Currently Nero AAC and FAAC are the only encoders to have gapless encoding/decoding support.<br />
<br />
===What software players can play back AAC music?===<br />
There are now a number of software players that can play back this new format. [[foobar2000]] is considered by many to be a very high quality audio player, and it is certainly capable of playing back AAC encoded files. Other players include [http://amarok.kde.org Amarok] using [http://www.audiocoding.com/ libfaad2], Apple's [[iTunes]], [[Winamp]], [http://www.real.com/ Real Player] and [http://www.microsoft.com/windows/windowsmedia/default.aspx Windows Media Player] using the [http://corecodec.org/projects/coreaac CoreAAC filter] and [http://www.elecard.com/download/ Moonlight MP4 Demultiplexer]. Also for Directshow-based applications playback and encoding is possible using the commercial [http://www.3ivx.com/ 3ivx filter suite].<br />
<br />
===What hardware players can play back AAC music?===<br />
There are also a few hardware players that can play back AAC audio. The most famous of these is the [[Apple iPod]] series of products, all of which feature AAC playback. A number of mobile (cell) phones also support unwrapped AAC (AAC not contained in the MP4 container). Recent Pioneer HT receivers can play back AAC files on a USB key or other mass-storage device.<br />
<br />
==Related Links==<br />
<br />
* [[AAC]] description article<br />
* Known [[AAC implementations]].<br />
* Read the [[AAC guide]] to learn how to obtain AAC/[[MP4]] files out of [[WAV]] files and CDs.<br />
* Detailed AAC comparisons can be found at [http://www.rjamorim.com/test/ Roberto's listening tests page].<br />
<br />
<br />
[[Category:Technical]]<br />
[[Category:Codecs]]</div>Gigapodhttps://wiki.hydrogenaudio.org/index.php?title=AAC_FAQ&diff=15631AAC FAQ2006-11-26T19:47:52Z<p>Gigapod: /* What's the best AAC encoder? */</p>
<hr />
<div>===Great, so you've given me all the technical stuff, but what is [[AAC]] really?===<br />
[[AAC]] is the culmination of the current state of the art audio encoding techniques. It is designed <br />
to improve upon and replace [[MP3]] as the defacto Audio Encoding standard. It usually offers (depending on the codec) equivalent quality to MP3 at a lower bitrate.<br />
<br />
===What is the difference between *.MP4 and *.M4A?===<br />
Besides the extension, absolutely nothing. Apple came up with extension to distiguish between files with Video and Audio (the [[MP4]] extension) and files with Audio only (the M4A extension). As far as the internal structure of the file, nothing is different.<br />
<br />
===What MPEG 4 extensions does the Apple iPod Accept?===<br />
The iPod accepts files with the MP4 extension, the M4A extension, the M4P extension (a Protected AAC file), and the M4B extension for audiobook files (which can be either protected or unprotected). It will not accept unwrapped AAC files (files with the .AAC extension).<br />
<br />
===What is the difference between LC (Low Complexity) and HE (High Efficiency)?===<br />
These are two of the various Object Types in the MPEG4 Systems Standard. LC is the most popular Object Type with all encoders/decoders supporting it. Currently, Nero, Coding Technolgies, and Panasonic have incorporated the HE AAC standard into their encoders, which allows for higher quality sound at lower bitrates then the LC Object Type does (at the same bitrate). The HE Object Type is only used for music with a bitrate of less than ~80kbps.<br />
<br />
===What's the best AAC encoder?===<br />
There is no best AAC encoder as such, because the quality of an encoding depends not only on the encoder implementation, but also on bitrates, sometimes on the specific audio sample being encoded, and finally, on the subjective perceptual judgement of the listener at playback time (which is also influenced by the equipment being used for playback).<br />
<br />
As such, since it is very difficult to quantify the quality of an encoder, [[Listening Tests]] are used.<br />
<br />
Guruboolez's [http://www.hydrogenaudio.org/forums/index.php?showtopic=29924 last test] concluded that [http://www.nero.com/en/ Nero AAC] was the best AAC encoder, at 128kbps, on classical samples, at the time the test was conducted.<br />
<br />
On the other hand, a public [http://www.rjamorim.com/test/multiformat128/results.html listening test] conducted by [[User:rjamorim|rjamorim]] in mid-2004 comparing different codecs, at 128kbps, with several music styles and featuring several listeners concluded that [[iTunes]] (the only AAC codec included in the test) was better than other codecs - even VBR-enabled ones. <br />
<br />
The quality of any encoder is not linear with bitrates, and therefore these results can not be extrapolated to other higher or lower bitrates. It can also be said with great confidence that both the iTunes AAC encoder and the Nero AAC encoder, although still under development at the end of 2006, are relatively 'mature' and should not fail badly (result in any obvious artifacts) on any particular sample at an average bitrate of 128kbps (i.e. Internet Profile for Nero AAC) or above (based on Roberto's listening tests, see bottom).<br />
<br />
Beyond that, only you can decide; you may want to conduct your own [[Listening Tests]], or you may base yourself on other parameters besides audio quality. See the [[Audio format guide]] for more information.<br />
<br />
===Do AAC encoded files play back gaplessly?===<br />
[[Gapless]] playback is not part of the AAC standard and as such is not mandatory. However, certain companies can choose to add gapless encoding/decoding if they desire, providing it doesn't break compatibility with previous decoders. This is what Ahead have done with their Nero AAC codec. The files get encoded with information that allows the gap heard between files to be removed. This however is only possible with supported players (currently these include foobar2000 and Nero ShowTime). Currently Nero AAC and FAAC are the only encoders to have gapless encoding/decoding support.<br />
<br />
===What software players can play back AAC music?===<br />
There are now a number of software players that can play back this new format. [[foobar2000]] is considered by many to be a very high quality audio player, and it is certainly capable of playing back AAC encoded files. Other players include [http://amarok.kde.org Amarok] using [http://www.audiocoding.com/ libfaad2], Apple's [[iTunes]], [[Winamp]], [http://www.real.com/ Real Player] and [http://www.microsoft.com/windows/windowsmedia/default.aspx Windows Media Player] using the [http://corecodec.org/projects/coreaac CoreAAC filter] and [http://www.elecard.com/download/ Moonlight MP4 Demultiplexer]. Also for Directshow-based applications playback and encoding is possible using the commercial [http://www.3ivx.com/ 3ivx filter suite].<br />
<br />
===What hardware players can play back AAC music?===<br />
There are also a few hardware players that can play back AAC audio. The most famous of these is the [[Apple iPod]] series of products, all of which feature AAC playback. A number of mobile (cell) phones also support unwrapped AAC (AAC not contained in the MP4 container). Recent Pioneer HT receivers can play back AAC files on a USB key or other mass-storage device.<br />
<br />
==Related Links==<br />
<br />
* [[AAC]] description article<br />
* Known [[AAC implementations]].<br />
* Read the [[AAC guide]] to learn how to obtain AAC/[[MP4]] files out of [[WAV]] files and CDs.<br />
* Detailed AAC comparisons can be found at [http://www.rjamorim.com/test/ Roberto's listening tests page].<br />
<br />
<br />
[[Category:Technical]]<br />
[[Category:Codecs]]</div>Gigapodhttps://wiki.hydrogenaudio.org/index.php?title=AAC_FAQ&diff=15630AAC FAQ2006-11-26T19:47:11Z<p>Gigapod: /* What's the best AAC encoder? */</p>
<hr />
<div>===Great, so you've given me all the technical stuff, but what is [[AAC]] really?===<br />
[[AAC]] is the culmination of the current state of the art audio encoding techniques. It is designed <br />
to improve upon and replace [[MP3]] as the defacto Audio Encoding standard. It usually offers (depending on the codec) equivalent quality to MP3 at a lower bitrate.<br />
<br />
===What is the difference between *.MP4 and *.M4A?===<br />
Besides the extension, absolutely nothing. Apple came up with extension to distiguish between files with Video and Audio (the [[MP4]] extension) and files with Audio only (the M4A extension). As far as the internal structure of the file, nothing is different.<br />
<br />
===What MPEG 4 extensions does the Apple iPod Accept?===<br />
The iPod accepts files with the MP4 extension, the M4A extension, the M4P extension (a Protected AAC file), and the M4B extension for audiobook files (which can be either protected or unprotected). It will not accept unwrapped AAC files (files with the .AAC extension).<br />
<br />
===What is the difference between LC (Low Complexity) and HE (High Efficiency)?===<br />
These are two of the various Object Types in the MPEG4 Systems Standard. LC is the most popular Object Type with all encoders/decoders supporting it. Currently, Nero, Coding Technolgies, and Panasonic have incorporated the HE AAC standard into their encoders, which allows for higher quality sound at lower bitrates then the LC Object Type does (at the same bitrate). The HE Object Type is only used for music with a bitrate of less than ~80kbps.<br />
<br />
===What's the best AAC encoder?===<br />
There is no best AAC encoder as such, because the quality of an encoding depends not only on the encoder implementation, but also on bitrates, sometimes on the specific audio sample being encoded, and finally, on the subjective perceptual judgement of the listener at playback time (which is also influenced by the equipment being used for playback).<br />
<br />
As such, since it is very difficult to quantify the quality of an encoder, [[listening tests]] are used.<br />
<br />
Guruboolez's [http://www.hydrogenaudio.org/forums/index.php?showtopic=29924 last test] concluded that [http://www.nero.com/en/ Nero AAC] was the best AAC encoder, at 128kbps, on classical samples, at the time the test was conducted.<br />
<br />
On the other hand, a public [http://www.rjamorim.com/test/multiformat128/results.html listening test] conducted by [[User:rjamorim|rjamorim]] in mid-2004 comparing different codecs, at 128kbps, with several music styles and featuring several listeners concluded that [[iTunes]] (the only AAC codec included in the test) was better than other codecs - even VBR-enabled ones. <br />
<br />
The quality of any encoder is not linear with bitrates, and therefore these results can not be extrapolated to other higher or lower bitrates. It can also be said with great confidence that both the iTunes AAC encoder and the Nero AAC encoder, although still under development at the end of 2006, are relatively 'mature' and should not fail badly (result in any obvious artifacts) on any particular sample at an average bitrate of 128kbps (i.e. Internet Profile for Nero AAC) or above (based on Roberto's listening tests, see bottom).<br />
<br />
Beyond that, only you can decide; you may want to conduct your own [[Listening Tests]], or you may base yourself on other parameters besides audio quality. See the [[Audio format guide]] for more information.<br />
<br />
===Do AAC encoded files play back gaplessly?===<br />
[[Gapless]] playback is not part of the AAC standard and as such is not mandatory. However, certain companies can choose to add gapless encoding/decoding if they desire, providing it doesn't break compatibility with previous decoders. This is what Ahead have done with their Nero AAC codec. The files get encoded with information that allows the gap heard between files to be removed. This however is only possible with supported players (currently these include foobar2000 and Nero ShowTime). Currently Nero AAC and FAAC are the only encoders to have gapless encoding/decoding support.<br />
<br />
===What software players can play back AAC music?===<br />
There are now a number of software players that can play back this new format. [[foobar2000]] is considered by many to be a very high quality audio player, and it is certainly capable of playing back AAC encoded files. Other players include [http://amarok.kde.org Amarok] using [http://www.audiocoding.com/ libfaad2], Apple's [[iTunes]], [[Winamp]], [http://www.real.com/ Real Player] and [http://www.microsoft.com/windows/windowsmedia/default.aspx Windows Media Player] using the [http://corecodec.org/projects/coreaac CoreAAC filter] and [http://www.elecard.com/download/ Moonlight MP4 Demultiplexer]. Also for Directshow-based applications playback and encoding is possible using the commercial [http://www.3ivx.com/ 3ivx filter suite].<br />
<br />
===What hardware players can play back AAC music?===<br />
There are also a few hardware players that can play back AAC audio. The most famous of these is the [[Apple iPod]] series of products, all of which feature AAC playback. A number of mobile (cell) phones also support unwrapped AAC (AAC not contained in the MP4 container). Recent Pioneer HT receivers can play back AAC files on a USB key or other mass-storage device.<br />
<br />
==Related Links==<br />
<br />
* [[AAC]] description article<br />
* Known [[AAC implementations]].<br />
* Read the [[AAC guide]] to learn how to obtain AAC/[[MP4]] files out of [[WAV]] files and CDs.<br />
* Detailed AAC comparisons can be found at [http://www.rjamorim.com/test/ Roberto's listening tests page].<br />
<br />
<br />
[[Category:Technical]]<br />
[[Category:Codecs]]</div>Gigapodhttps://wiki.hydrogenaudio.org/index.php?title=AAC_FAQ&diff=15629AAC FAQ2006-11-26T19:45:53Z<p>Gigapod: /* What's the best AAC encoder? */</p>
<hr />
<div>===Great, so you've given me all the technical stuff, but what is [[AAC]] really?===<br />
[[AAC]] is the culmination of the current state of the art audio encoding techniques. It is designed <br />
to improve upon and replace [[MP3]] as the defacto Audio Encoding standard. It usually offers (depending on the codec) equivalent quality to MP3 at a lower bitrate.<br />
<br />
===What is the difference between *.MP4 and *.M4A?===<br />
Besides the extension, absolutely nothing. Apple came up with extension to distiguish between files with Video and Audio (the [[MP4]] extension) and files with Audio only (the M4A extension). As far as the internal structure of the file, nothing is different.<br />
<br />
===What MPEG 4 extensions does the Apple iPod Accept?===<br />
The iPod accepts files with the MP4 extension, the M4A extension, the M4P extension (a Protected AAC file), and the M4B extension for audiobook files (which can be either protected or unprotected). It will not accept unwrapped AAC files (files with the .AAC extension).<br />
<br />
===What is the difference between LC (Low Complexity) and HE (High Efficiency)?===<br />
These are two of the various Object Types in the MPEG4 Systems Standard. LC is the most popular Object Type with all encoders/decoders supporting it. Currently, Nero, Coding Technolgies, and Panasonic have incorporated the HE AAC standard into their encoders, which allows for higher quality sound at lower bitrates then the LC Object Type does (at the same bitrate). The HE Object Type is only used for music with a bitrate of less than ~80kbps.<br />
<br />
===What's the best AAC encoder?===<br />
There is no best AAC encoder as such, because the quality of an encoding depends not only on the encoder implementation, but also on bitrates, sometimes on the specific audio sample being encoded, and finally, on the subjective perceptual judgement of the listener at playback time (which is also influenced by the equipment being used for playback).<br />
<br />
As such, since it is very difficult to quantify the quality of an encoder, [[Listening Tests][listening tests]] are used.<br />
<br />
Guruboolez's [http://www.hydrogenaudio.org/forums/index.php?showtopic=29924 last test] concluded that [http://www.nero.com/en/ Nero AAC] was the best AAC encoder, at 128kbps, on classical samples, at the time the test was conducted.<br />
<br />
On the other hand, a public [http://www.rjamorim.com/test/multiformat128/results.html listening test] conducted by [[User:rjamorim|rjamorim]] in mid-2004 comparing different codecs, at 128kbps, with several music styles and featuring several listeners concluded that [[iTunes]] (the only AAC codec included in the test) was better than other codecs - even VBR-enabled ones. <br />
<br />
The quality of any encoder is not linear with bitrates, and therefore these results can not be extrapolated to other higher or lower bitrates. It can also be said with great confidence that both the iTunes AAC encoder and the Nero AAC encoder, although still under development at the end of 2006, are relatively 'mature' and should not fail badly (result in any obvious artifacts) on any particular sample at an average bitrate of 128kbps (i.e. Internet Profile for Nero AAC) or above (based on Roberto's listening tests, see bottom).<br />
<br />
Beyond that, only you can decide; you may want to conduct your own [[Listening Tests][listening tests]], or you may base yourself on other parameters besides audio quality. See the [[Audio format guide]] for more information.<br />
<br />
===Do AAC encoded files play back gaplessly?===<br />
[[Gapless]] playback is not part of the AAC standard and as such is not mandatory. However, certain companies can choose to add gapless encoding/decoding if they desire, providing it doesn't break compatibility with previous decoders. This is what Ahead have done with their Nero AAC codec. The files get encoded with information that allows the gap heard between files to be removed. This however is only possible with supported players (currently these include foobar2000 and Nero ShowTime). Currently Nero AAC and FAAC are the only encoders to have gapless encoding/decoding support.<br />
<br />
===What software players can play back AAC music?===<br />
There are now a number of software players that can play back this new format. [[foobar2000]] is considered by many to be a very high quality audio player, and it is certainly capable of playing back AAC encoded files. Other players include [http://amarok.kde.org Amarok] using [http://www.audiocoding.com/ libfaad2], Apple's [[iTunes]], [[Winamp]], [http://www.real.com/ Real Player] and [http://www.microsoft.com/windows/windowsmedia/default.aspx Windows Media Player] using the [http://corecodec.org/projects/coreaac CoreAAC filter] and [http://www.elecard.com/download/ Moonlight MP4 Demultiplexer]. Also for Directshow-based applications playback and encoding is possible using the commercial [http://www.3ivx.com/ 3ivx filter suite].<br />
<br />
===What hardware players can play back AAC music?===<br />
There are also a few hardware players that can play back AAC audio. The most famous of these is the [[Apple iPod]] series of products, all of which feature AAC playback. A number of mobile (cell) phones also support unwrapped AAC (AAC not contained in the MP4 container). Recent Pioneer HT receivers can play back AAC files on a USB key or other mass-storage device.<br />
<br />
==Related Links==<br />
<br />
* [[AAC]] description article<br />
* Known [[AAC implementations]].<br />
* Read the [[AAC guide]] to learn how to obtain AAC/[[MP4]] files out of [[WAV]] files and CDs.<br />
* Detailed AAC comparisons can be found at [http://www.rjamorim.com/test/ Roberto's listening tests page].<br />
<br />
<br />
[[Category:Technical]]<br />
[[Category:Codecs]]</div>Gigapodhttps://wiki.hydrogenaudio.org/index.php?title=AAC_FAQ&diff=15628AAC FAQ2006-11-26T18:56:21Z<p>Gigapod: /* What hardware players can play back AAC music? */</p>
<hr />
<div>===Great, so you've given me all the technical stuff, but what is [[AAC]] really?===<br />
[[AAC]] is the culmination of the current state of the art audio encoding techniques. It is designed <br />
to improve upon and replace [[MP3]] as the defacto Audio Encoding standard. It usually offers (depending on the codec) equivalent quality to MP3 at a lower bitrate.<br />
<br />
===What is the difference between *.MP4 and *.M4A?===<br />
Besides the extension, absolutely nothing. Apple came up with extension to distiguish between files with Video and Audio (the [[MP4]] extension) and files with Audio only (the M4A extension). As far as the internal structure of the file, nothing is different.<br />
<br />
===What MPEG 4 extensions does the Apple iPod Accept?===<br />
The iPod accepts files with the MP4 extension, the M4A extension, the M4P extension (a Protected AAC file), and the M4B extension for audiobook files (which can be either protected or unprotected). It will not accept unwrapped AAC files (files with the .AAC extension).<br />
<br />
===What is the difference between LC (Low Complexity) and HE (High Efficiency)?===<br />
These are two of the various Object Types in the MPEG4 Systems Standard. LC is the most popular Object Type with all encoders/decoders supporting it. Currently, Nero, Coding Technolgies, and Panasonic have incorporated the HE AAC standard into their encoders, which allows for higher quality sound at lower bitrates then the LC Object Type does (at the same bitrate). The HE Object Type is only used for music with a bitrate of less than ~80kbps.<br />
<br />
===What's the best AAC encoder?===<br />
There is no best AAC encoder as such. It can be said with reasonable confidence (based on guruboolez's last test, [http://www.hydrogenaudio.org/forums/index.php?showtopic=29924 hear]) that [http://www.nero.com/en/ Nero AAC] is the best AAC encoder at 128kbps on classical samples. On the other hand, a public [http://www.rjamorim.com/test/multiformat128/results.html listening test] conduced by [[User:rjamorim|rjamorim]] in mid-2004 comparing several music styles and featuring several listeners concluded that [[iTunes]] at 128kbps is better than other codecs at the same bitrate - even VBR-enabled ones. <br />
<br />
Anyway, the quality of any encoder is not linear and therefore these results can not be extrapolated to other bitrates. It can also be said with great confidence that both the iTunes encoder and the Nero encoder are 'mature' and should not fail badly on any particular sample at an average bitrate of 128kbps (i.e. Internet Profile for Nero AAC) or above (based on Roberto's listening tests, see bottom). Beyond that, only you can decide through [[ABX]] testing. See the [[Audio format guide]] <br />
for more information.<br />
<br />
===Do AAC encoded files play back gaplessly?===<br />
[[Gapless]] playback is not part of the AAC standard and as such is not mandatory. However, certain companies can choose to add gapless encoding/decoding if they desire, providing it doesn't break compatibility with previous decoders. This is what Ahead have done with their Nero AAC codec. The files get encoded with information that allows the gap heard between files to be removed. This however is only possible with supported players (currently these include foobar2000 and Nero ShowTime). Currently Nero AAC and FAAC are the only encoders to have gapless encoding/decoding support.<br />
<br />
===What software players can play back AAC music?===<br />
There are now a number of software players that can play back this new format. [[foobar2000]] is considered by many to be a very high quality audio player, and it is certainly capable of playing back AAC encoded files. Other players include [http://amarok.kde.org Amarok] using [http://www.audiocoding.com/ libfaad2], Apple's [[iTunes]], [[Winamp]], [http://www.real.com/ Real Player] and [http://www.microsoft.com/windows/windowsmedia/default.aspx Windows Media Player] using the [http://corecodec.org/projects/coreaac CoreAAC filter] and [http://www.elecard.com/download/ Moonlight MP4 Demultiplexer]. Also for Directshow-based applications playback and encoding is possible using the commercial [http://www.3ivx.com/ 3ivx filter suite].<br />
<br />
===What hardware players can play back AAC music?===<br />
There are also a few hardware players that can play back AAC audio. The most famous of these is the [[Apple iPod]] series of products, all of which feature AAC playback. A number of mobile (cell) phones also support unwrapped AAC (AAC not contained in the MP4 container). Recent Pioneer HT receivers can play back AAC files on a USB key or other mass-storage device.<br />
<br />
==Related Links==<br />
<br />
* [[AAC]] description article<br />
* Known [[AAC implementations]].<br />
* Read the [[AAC guide]] to learn how to obtain AAC/[[MP4]] files out of [[WAV]] files and CDs.<br />
* Detailed AAC comparisons can be found at [http://www.rjamorim.com/test/ Roberto's listening tests page].<br />
<br />
<br />
[[Category:Technical]]<br />
[[Category:Codecs]]</div>Gigapodhttps://wiki.hydrogenaudio.org/index.php?title=AAC_FAQ&diff=15627AAC FAQ2006-11-26T18:54:55Z<p>Gigapod: /* What software players can play back AAC music? */</p>
<hr />
<div>===Great, so you've given me all the technical stuff, but what is [[AAC]] really?===<br />
[[AAC]] is the culmination of the current state of the art audio encoding techniques. It is designed <br />
to improve upon and replace [[MP3]] as the defacto Audio Encoding standard. It usually offers (depending on the codec) equivalent quality to MP3 at a lower bitrate.<br />
<br />
===What is the difference between *.MP4 and *.M4A?===<br />
Besides the extension, absolutely nothing. Apple came up with extension to distiguish between files with Video and Audio (the [[MP4]] extension) and files with Audio only (the M4A extension). As far as the internal structure of the file, nothing is different.<br />
<br />
===What MPEG 4 extensions does the Apple iPod Accept?===<br />
The iPod accepts files with the MP4 extension, the M4A extension, the M4P extension (a Protected AAC file), and the M4B extension for audiobook files (which can be either protected or unprotected). It will not accept unwrapped AAC files (files with the .AAC extension).<br />
<br />
===What is the difference between LC (Low Complexity) and HE (High Efficiency)?===<br />
These are two of the various Object Types in the MPEG4 Systems Standard. LC is the most popular Object Type with all encoders/decoders supporting it. Currently, Nero, Coding Technolgies, and Panasonic have incorporated the HE AAC standard into their encoders, which allows for higher quality sound at lower bitrates then the LC Object Type does (at the same bitrate). The HE Object Type is only used for music with a bitrate of less than ~80kbps.<br />
<br />
===What's the best AAC encoder?===<br />
There is no best AAC encoder as such. It can be said with reasonable confidence (based on guruboolez's last test, [http://www.hydrogenaudio.org/forums/index.php?showtopic=29924 hear]) that [http://www.nero.com/en/ Nero AAC] is the best AAC encoder at 128kbps on classical samples. On the other hand, a public [http://www.rjamorim.com/test/multiformat128/results.html listening test] conduced by [[User:rjamorim|rjamorim]] in mid-2004 comparing several music styles and featuring several listeners concluded that [[iTunes]] at 128kbps is better than other codecs at the same bitrate - even VBR-enabled ones. <br />
<br />
Anyway, the quality of any encoder is not linear and therefore these results can not be extrapolated to other bitrates. It can also be said with great confidence that both the iTunes encoder and the Nero encoder are 'mature' and should not fail badly on any particular sample at an average bitrate of 128kbps (i.e. Internet Profile for Nero AAC) or above (based on Roberto's listening tests, see bottom). Beyond that, only you can decide through [[ABX]] testing. See the [[Audio format guide]] <br />
for more information.<br />
<br />
===Do AAC encoded files play back gaplessly?===<br />
[[Gapless]] playback is not part of the AAC standard and as such is not mandatory. However, certain companies can choose to add gapless encoding/decoding if they desire, providing it doesn't break compatibility with previous decoders. This is what Ahead have done with their Nero AAC codec. The files get encoded with information that allows the gap heard between files to be removed. This however is only possible with supported players (currently these include foobar2000 and Nero ShowTime). Currently Nero AAC and FAAC are the only encoders to have gapless encoding/decoding support.<br />
<br />
===What software players can play back AAC music?===<br />
There are now a number of software players that can play back this new format. [[foobar2000]] is considered by many to be a very high quality audio player, and it is certainly capable of playing back AAC encoded files. Other players include [http://amarok.kde.org Amarok] using [http://www.audiocoding.com/ libfaad2], Apple's [[iTunes]], [[Winamp]], [http://www.real.com/ Real Player] and [http://www.microsoft.com/windows/windowsmedia/default.aspx Windows Media Player] using the [http://corecodec.org/projects/coreaac CoreAAC filter] and [http://www.elecard.com/download/ Moonlight MP4 Demultiplexer]. Also for Directshow-based applications playback and encoding is possible using the commercial [http://www.3ivx.com/ 3ivx filter suite].<br />
<br />
===What hardware players can play back AAC music?===<br />
There are also a few hardware players that can play back AAC audio. The most famous of these is the [[Apple iPod]] series of products, all of which feature AAC playback. A number of mobile (cell) phones also support unwrapped AAC (AAC not contained in the MP4 container).<br />
<br />
<br />
==Related Links==<br />
<br />
* [[AAC]] description article<br />
* Known [[AAC implementations]].<br />
* Read the [[AAC guide]] to learn how to obtain AAC/[[MP4]] files out of [[WAV]] files and CDs.<br />
* Detailed AAC comparisons can be found at [http://www.rjamorim.com/test/ Roberto's listening tests page].<br />
<br />
<br />
[[Category:Technical]]<br />
[[Category:Codecs]]</div>Gigapodhttps://wiki.hydrogenaudio.org/index.php?title=Perceptual_Noise_Substitution&diff=15622Perceptual Noise Substitution2006-11-26T13:52:43Z<p>Gigapod: </p>
<hr />
<div>Part of the MPEG-4 Standard ISO/IEC 14496-3 Audio (MPEG-4 General Audio) and not available within MPEG-2 AAC i.e. PNS is one of the new tools introduced in MPEG-4 AAC.<br />
<br />
Based on the fact that the exact spectral components of noise in audio signals is not ''perceptible'', hence the original noise can be ''substituted'' by randomly generated noise over an appropriate spectral region at an equivalent power level during decoding, i.o.w. "noise always sounds the same¨.<br />
<br />
==References==<br />
* MPEG-4 Advanced Audio Coding by Peter Doliwa, http://www.ibr.cs.tu-bs.de/courses/ss04/skm/mpeg-4-aac.pdf<br />
<br />
<br />
{{stub}}<br />
[[Category:Technical]]</div>Gigapodhttps://wiki.hydrogenaudio.org/index.php?title=Perceptual_Noise_Substitution&diff=15621Perceptual Noise Substitution2006-11-26T13:52:29Z<p>Gigapod: </p>
<hr />
<div>Part of the MPEG-4 Standard ISO/IEC 14496-3 Audio (MPEG-4 General Audio) and not available within MPEG-2 AAC i.e. PNS is one of the new tools introduced in MPEG-4 AAC.<br />
<br />
Based on the fact that the exact spectral components of noise in audio signals is not ''perceptible'', hence the original noise can be ''substituted'' by randomly generated noise over an appropriate spectral region at an equivalent power level during decoding, i.o.w. "noise always sounds the same¨.<br />
<br />
==References==<br />
* MPEG-4 Advanced Audio Coding by Peter Doliwa, http://www.ibr.cs.tu-bs.de/courses/ss04/skm/mpeg-4-aac.pdf<br />
<br />
{{stub}}<br />
[[Category:Technical]]</div>Gigapodhttps://wiki.hydrogenaudio.org/index.php?title=Perceptual_Noise_Substitution&diff=15619Perceptual Noise Substitution2006-11-26T13:31:30Z<p>Gigapod: </p>
<hr />
<div>Part of the MPEG-4 Standard ISO/IEC 14496-3 Audio (MPEG-4 General Audio) and not available within MPEG-2 AAC.<br />
<br />
Based on the fact that the exact spectral components of noise in audio signals is not ''perceptible'', hence the original noise can be ''substituted'' by randomly generated noise over an appropriate spectral region at an equivalent power level, i.o.w. "noise always sounds the same¨.<br />
<br />
{{stub}}<br />
[[Category:Technical]]</div>Gigapodhttps://wiki.hydrogenaudio.org/index.php?title=Perceptual_Noise_Substitution&diff=15618Perceptual Noise Substitution2006-11-26T13:29:50Z<p>Gigapod: </p>
<hr />
<div>Part of the MPEG-4 Standard ISO/IEC 14496-3 Audio (MPEG-4 General Audio) and not available within MPEG-2 AAC.<br />
<br />
Based on the fact that the exact spectral components of noise in audio signals is not perceptible, hence the original noise can be replaced by randomly generated noise over an appropriate spectral region at an equivalent power level, i.o.w. "noise always sounds the same¨.<br />
<br />
{{stub}}<br />
[[Category:Technical]]</div>Gigapodhttps://wiki.hydrogenaudio.org/index.php?title=Temporal_Noise_Shaping&diff=15617Temporal Noise Shaping2006-11-26T12:29:48Z<p>Gigapod: </p>
<hr />
<div>Temporal Noise Shaping is a technique for reshaping the quantization noise over time (hence the term "Temporal"). The objective is to reduce artifacts in transient and speech signals.<br />
TNS was introduced into MPEG-2 AAC.<br />
TNS is not available in MP3.<br />
TNS is not applied for short blocks, only for long blocks.<br />
{{stub}}<br />
[[Category:Technical]]</div>Gigapodhttps://wiki.hydrogenaudio.org/index.php?title=Perceptual_Noise_Substitution&diff=15616Perceptual Noise Substitution2006-11-26T12:17:28Z<p>Gigapod: </p>
<hr />
<div>{{stub}}<br />
[[Category:Technical]]</div>Gigapodhttps://wiki.hydrogenaudio.org/index.php?title=Glossary_Of_Audio_Terms&diff=15615Glossary Of Audio Terms2006-11-26T12:17:06Z<p>Gigapod: /* P */</p>
<hr />
<div>__NOTOC__<br />
<br />
==A==<br />
<br />
* [[AAC]]<br />
* [[ABR]]<br />
* [[ABX]]/[[ABX|ABX testing]]<br />
* [[AC3]]<br />
* [[Aliasing]]<br />
* [[AltPresets]]<br />
* [[Ambisonics]] <br />
* [[Amplitude]]<br />
* [[APE]] ([[Monkey's Audio]])<br />
* [[APE Tags]]<br />
* [[Artifact]]/[[Artifact|Distortion]]<br />
* [[ATH]] (Absolute Threshold of Hearing)<br />
* [[ASIO]] (Audio Stream Input Output)<br />
* [[ASPI]] (CD-ROM Installation)<br />
<br />
==B==<br />
<br />
* [[Bandpass filter]]<br />
* [[Bandstop filter]]/[[Bandstop filter|Band reject]]<br />
* [[Bandwidth]]<br />
* [[Bark]]<br />
* [[Bit reservoir]]<br />
* [[Bitrate]]<br />
* [[Bit depth|Bits per sample]] / [[Bit depth]]<br />
* [[Blind test]]<br />
* [[Block switching]]<br />
* [[BSAC]] (Bit-sliced arithmetic coding)<br />
<br />
==C==<br />
<br />
* [[CBR]]/[[Constant bitrate]]/[[CBR|Constant bitrate coding]]<br />
* [[Channel coupling]]<br />
* [[Clipping]]<br />
* [[Codec]]<br />
* [[Container format]]<br />
* [[Critical band]]<br />
* [[C1/C2 errors]]<br />
<br />
==D==<br />
<br />
* [[dB]]/[[dB|Decibel]]<br />
* [[DC coefficient]]<br />
* [[DCT]]<br />
* [[DCT coefficient]]<br />
* [[Digital Radio Mondiale|DRM]] (Digital Radio Mondiale)<br />
* [[Digital Rights Management|DRM]] (Digital Rights Management)<br />
* [[Dither]] <br />
* [[DSP]]<br />
* [[DTS]]<br />
<br />
==F==<br />
<br />
* [[FFT]]<br />
* [[Filterbank]]<br />
* [[FIR filter]] (Finite Impulse Response Filters)<br />
* [[FLAC]]<br />
* [[Frequency]]<br />
* [[Frequency domain]]<br />
<br />
==H==<br />
<br />
* [[Harmonics]]<br />
* [[Highpass]]<br />
* [[Huffman coding]]<br />
* [[HDMI]] (High Definition Multimedia Interface)<br />
<br />
==I==<br />
<br />
* [[ID3]]<br />
* [[IIR filter]] (Infinite Impulse Response Filters)<br />
* [[Impulse]]<br />
* [[Intensity stereo]]<br />
* [[Inverse mix]]<br />
<br />
==J==<br />
<br />
* [[Joint stereo]]<br />
<br />
==L==<br />
<br />
* [[LFE]] (Low Frequency Extension)<br />
* [[LPAC]]<br />
* [[LPC]] (Linear Prediction Coding)<br />
* [[Long block]]<br />
* [[Lossless]]<br />
* [[Lossy]]<br />
* [[Lowpass]]<br />
* [[ATH|LTQ]] (Level of Threshold in Quiet)<br />
<br />
==M==<br />
<br />
* [[Masking]]<br />
* [[MDCT]]<br />
* [[Metadata]]<br />
* [[Mid-side stereo]]<br />
* [[Monkey's Audio]]<br />
* [[MP3]] (MPEG 1 Audio Layer 3)<br />
* [[MP3Pro]]<br />
* [[Musepack|MPC]]/[[Musepack|MP+]]/[[Musepack]]/[[Musepack|Mpeg Plus]]<br />
* [[Mpeg]] (Motion Picture Expert Group)<br />
* [[MPEG-4]]<br />
* [[Multichannel]]<br />
<br />
==N==<br />
<br />
* [[Neural network]]<br />
* [[Noise shaping]]<br />
* [[Noise normalization]] <br />
* [[Notch filter]]<br />
* [[Nyquist rate]]/[[Nyquist frequency]]/[[Nyquist sampling theorem]]<br />
<br />
==O==<br />
<br />
* [[Ogg]]<br />
* [[Ogg Vorbis]]<br />
* [[OptimFROG]]<br />
<br />
==P==<br />
<br />
* [[Parametric stereo]]/[[PS]]<br />
* [[PCM]]<br />
* [[PNS]]/[[Perceptual Noise Substitution]]<br />
* [[Point stereo]] <br />
* [[Pre echo]]/[[Pre echo|Post echo]]<br />
* [[Psychoacoustic]]<br />
* [[Psychoacoustic#Psychoacoustic model|Psychoacoustic model]]<br />
* [[Pre-masking]]/[[Pre-masking|Post-masking]]<br />
<br />
==Q==<br />
<br />
* [[Quantize]]/[[Quantizer]]/[[Quantization]]<br />
* [[Quantization noise]]<br />
<br />
==R==<br />
<br />
* [[Range coding]]<br />
* [[Resampling]]<br />
* [[Rice coding]]<br />
* [[Ringing]]<br />
<br />
==S==<br />
<br />
* [[Sampling rate]]/[[Sampling rate|Sample rate]]/[[Sampling rate|Sampling frequency]]<br />
* [[SBR]] (Spectral Band Replication)<br />
* [[Scale factor]]/[[Scale factor|Scale factor band]]<br />
* [[Short block]]<br />
* [[Shorten]] ([[SHN]])<br />
* [[Sigma Delta Modulation]]<br />
* [[SNR]] (Signal-to-Noise Ratio)<br />
* [[Spectrogram]]<br />
* [[S/PDIF]] (Sony-Phillips Digital Interface)<br />
* [[Streaming]]<br />
* [[Subband]]<br />
* [[SZIP]]<br />
<br />
==T==<br />
<br />
* [[TTA]] '''T'''rue '''T'''ap '''A'''udio<br />
* [[Temporal accuracy]]<br />
* [[Temporal smearing]]<br />
* [[Temporal Noise Shaping]]<br />
* [[Time domain]]<br />
* [[Temporal Noise Shaping|TNS]]<br />
* [[Tonality]]/[[Tonality|Tonal signals]]/[[Tonality|Tonality estimation]]<br />
* [[Transcoding]]<br />
* [[Transform]]<br />
* [[Transient]]<br />
* [[Transient smearing]]<br />
* [[Transparency]]<br />
* [[Tremor]]<br />
<br />
==V==<br />
<br />
* [[VBR]] (Variable Bitrate/Variable Bitrate Coding)<br />
* [[Vector quantization]]<br />
<br />
==W==<br />
<br />
* [[WAV]]<br />
* [[Wavelet]]s<br />
* [[WavPack]]<br />
* [[Window function]] <br />
* [[WMA]]<br />
<br />
<br />
[[Category:Guides]]</div>Gigapodhttps://wiki.hydrogenaudio.org/index.php?title=Glossary_Of_Audio_Terms&diff=15614Glossary Of Audio Terms2006-11-26T12:16:41Z<p>Gigapod: /* P */</p>
<hr />
<div>__NOTOC__<br />
<br />
==A==<br />
<br />
* [[AAC]]<br />
* [[ABR]]<br />
* [[ABX]]/[[ABX|ABX testing]]<br />
* [[AC3]]<br />
* [[Aliasing]]<br />
* [[AltPresets]]<br />
* [[Ambisonics]] <br />
* [[Amplitude]]<br />
* [[APE]] ([[Monkey's Audio]])<br />
* [[APE Tags]]<br />
* [[Artifact]]/[[Artifact|Distortion]]<br />
* [[ATH]] (Absolute Threshold of Hearing)<br />
* [[ASIO]] (Audio Stream Input Output)<br />
* [[ASPI]] (CD-ROM Installation)<br />
<br />
==B==<br />
<br />
* [[Bandpass filter]]<br />
* [[Bandstop filter]]/[[Bandstop filter|Band reject]]<br />
* [[Bandwidth]]<br />
* [[Bark]]<br />
* [[Bit reservoir]]<br />
* [[Bitrate]]<br />
* [[Bit depth|Bits per sample]] / [[Bit depth]]<br />
* [[Blind test]]<br />
* [[Block switching]]<br />
* [[BSAC]] (Bit-sliced arithmetic coding)<br />
<br />
==C==<br />
<br />
* [[CBR]]/[[Constant bitrate]]/[[CBR|Constant bitrate coding]]<br />
* [[Channel coupling]]<br />
* [[Clipping]]<br />
* [[Codec]]<br />
* [[Container format]]<br />
* [[Critical band]]<br />
* [[C1/C2 errors]]<br />
<br />
==D==<br />
<br />
* [[dB]]/[[dB|Decibel]]<br />
* [[DC coefficient]]<br />
* [[DCT]]<br />
* [[DCT coefficient]]<br />
* [[Digital Radio Mondiale|DRM]] (Digital Radio Mondiale)<br />
* [[Digital Rights Management|DRM]] (Digital Rights Management)<br />
* [[Dither]] <br />
* [[DSP]]<br />
* [[DTS]]<br />
<br />
==F==<br />
<br />
* [[FFT]]<br />
* [[Filterbank]]<br />
* [[FIR filter]] (Finite Impulse Response Filters)<br />
* [[FLAC]]<br />
* [[Frequency]]<br />
* [[Frequency domain]]<br />
<br />
==H==<br />
<br />
* [[Harmonics]]<br />
* [[Highpass]]<br />
* [[Huffman coding]]<br />
* [[HDMI]] (High Definition Multimedia Interface)<br />
<br />
==I==<br />
<br />
* [[ID3]]<br />
* [[IIR filter]] (Infinite Impulse Response Filters)<br />
* [[Impulse]]<br />
* [[Intensity stereo]]<br />
* [[Inverse mix]]<br />
<br />
==J==<br />
<br />
* [[Joint stereo]]<br />
<br />
==L==<br />
<br />
* [[LFE]] (Low Frequency Extension)<br />
* [[LPAC]]<br />
* [[LPC]] (Linear Prediction Coding)<br />
* [[Long block]]<br />
* [[Lossless]]<br />
* [[Lossy]]<br />
* [[Lowpass]]<br />
* [[ATH|LTQ]] (Level of Threshold in Quiet)<br />
<br />
==M==<br />
<br />
* [[Masking]]<br />
* [[MDCT]]<br />
* [[Metadata]]<br />
* [[Mid-side stereo]]<br />
* [[Monkey's Audio]]<br />
* [[MP3]] (MPEG 1 Audio Layer 3)<br />
* [[MP3Pro]]<br />
* [[Musepack|MPC]]/[[Musepack|MP+]]/[[Musepack]]/[[Musepack|Mpeg Plus]]<br />
* [[Mpeg]] (Motion Picture Expert Group)<br />
* [[MPEG-4]]<br />
* [[Multichannel]]<br />
<br />
==N==<br />
<br />
* [[Neural network]]<br />
* [[Noise shaping]]<br />
* [[Noise normalization]] <br />
* [[Notch filter]]<br />
* [[Nyquist rate]]/[[Nyquist frequency]]/[[Nyquist sampling theorem]]<br />
<br />
==O==<br />
<br />
* [[Ogg]]<br />
* [[Ogg Vorbis]]<br />
* [[OptimFROG]]<br />
<br />
==P==<br />
<br />
* [[Parametric stereo]]/[[PS]]<br />
* [[PCM]]<br />
* [[Perceptual Noise Substitution]]<br />
* [[PNS]]/[[Perceptual Noise Substitution]]<br />
* [[Point stereo]] <br />
* [[Pre echo]]/[[Pre echo|Post echo]]<br />
* [[Psychoacoustic]]<br />
* [[Psychoacoustic#Psychoacoustic model|Psychoacoustic model]]<br />
* [[Pre-masking]]/[[Pre-masking|Post-masking]]<br />
<br />
==Q==<br />
<br />
* [[Quantize]]/[[Quantizer]]/[[Quantization]]<br />
* [[Quantization noise]]<br />
<br />
==R==<br />
<br />
* [[Range coding]]<br />
* [[Resampling]]<br />
* [[Rice coding]]<br />
* [[Ringing]]<br />
<br />
==S==<br />
<br />
* [[Sampling rate]]/[[Sampling rate|Sample rate]]/[[Sampling rate|Sampling frequency]]<br />
* [[SBR]] (Spectral Band Replication)<br />
* [[Scale factor]]/[[Scale factor|Scale factor band]]<br />
* [[Short block]]<br />
* [[Shorten]] ([[SHN]])<br />
* [[Sigma Delta Modulation]]<br />
* [[SNR]] (Signal-to-Noise Ratio)<br />
* [[Spectrogram]]<br />
* [[S/PDIF]] (Sony-Phillips Digital Interface)<br />
* [[Streaming]]<br />
* [[Subband]]<br />
* [[SZIP]]<br />
<br />
==T==<br />
<br />
* [[TTA]] '''T'''rue '''T'''ap '''A'''udio<br />
* [[Temporal accuracy]]<br />
* [[Temporal smearing]]<br />
* [[Temporal Noise Shaping]]<br />
* [[Time domain]]<br />
* [[Temporal Noise Shaping|TNS]]<br />
* [[Tonality]]/[[Tonality|Tonal signals]]/[[Tonality|Tonality estimation]]<br />
* [[Transcoding]]<br />
* [[Transform]]<br />
* [[Transient]]<br />
* [[Transient smearing]]<br />
* [[Transparency]]<br />
* [[Tremor]]<br />
<br />
==V==<br />
<br />
* [[VBR]] (Variable Bitrate/Variable Bitrate Coding)<br />
* [[Vector quantization]]<br />
<br />
==W==<br />
<br />
* [[WAV]]<br />
* [[Wavelet]]s<br />
* [[WavPack]]<br />
* [[Window function]] <br />
* [[WMA]]<br />
<br />
<br />
[[Category:Guides]]</div>Gigapodhttps://wiki.hydrogenaudio.org/index.php?title=Temporal_Noise_Shaping&diff=15613Temporal Noise Shaping2006-11-26T12:11:46Z<p>Gigapod: </p>
<hr />
<div>{{stub}}<br />
[[Category:Technical]]</div>Gigapodhttps://wiki.hydrogenaudio.org/index.php?title=Temporal_Noise_Shaping&diff=15612Temporal Noise Shaping2006-11-26T12:09:51Z<p>Gigapod: </p>
<hr />
<div>[[Stub]]<br />
[[Category:Technical]]</div>Gigapodhttps://wiki.hydrogenaudio.org/index.php?title=Glossary_Of_Audio_Terms&diff=15611Glossary Of Audio Terms2006-11-26T12:07:47Z<p>Gigapod: /* T */</p>
<hr />
<div>__NOTOC__<br />
<br />
==A==<br />
<br />
* [[AAC]]<br />
* [[ABR]]<br />
* [[ABX]]/[[ABX|ABX testing]]<br />
* [[AC3]]<br />
* [[Aliasing]]<br />
* [[AltPresets]]<br />
* [[Ambisonics]] <br />
* [[Amplitude]]<br />
* [[APE]] ([[Monkey's Audio]])<br />
* [[APE Tags]]<br />
* [[Artifact]]/[[Artifact|Distortion]]<br />
* [[ATH]] (Absolute Threshold of Hearing)<br />
* [[ASIO]] (Audio Stream Input Output)<br />
* [[ASPI]] (CD-ROM Installation)<br />
<br />
==B==<br />
<br />
* [[Bandpass filter]]<br />
* [[Bandstop filter]]/[[Bandstop filter|Band reject]]<br />
* [[Bandwidth]]<br />
* [[Bark]]<br />
* [[Bit reservoir]]<br />
* [[Bitrate]]<br />
* [[Bit depth|Bits per sample]] / [[Bit depth]]<br />
* [[Blind test]]<br />
* [[Block switching]]<br />
* [[BSAC]] (Bit-sliced arithmetic coding)<br />
<br />
==C==<br />
<br />
* [[CBR]]/[[Constant bitrate]]/[[CBR|Constant bitrate coding]]<br />
* [[Channel coupling]]<br />
* [[Clipping]]<br />
* [[Codec]]<br />
* [[Container format]]<br />
* [[Critical band]]<br />
* [[C1/C2 errors]]<br />
<br />
==D==<br />
<br />
* [[dB]]/[[dB|Decibel]]<br />
* [[DC coefficient]]<br />
* [[DCT]]<br />
* [[DCT coefficient]]<br />
* [[Digital Radio Mondiale|DRM]] (Digital Radio Mondiale)<br />
* [[Digital Rights Management|DRM]] (Digital Rights Management)<br />
* [[Dither]] <br />
* [[DSP]]<br />
* [[DTS]]<br />
<br />
==F==<br />
<br />
* [[FFT]]<br />
* [[Filterbank]]<br />
* [[FIR filter]] (Finite Impulse Response Filters)<br />
* [[FLAC]]<br />
* [[Frequency]]<br />
* [[Frequency domain]]<br />
<br />
==H==<br />
<br />
* [[Harmonics]]<br />
* [[Highpass]]<br />
* [[Huffman coding]]<br />
* [[HDMI]] (High Definition Multimedia Interface)<br />
<br />
==I==<br />
<br />
* [[ID3]]<br />
* [[IIR filter]] (Infinite Impulse Response Filters)<br />
* [[Impulse]]<br />
* [[Intensity stereo]]<br />
* [[Inverse mix]]<br />
<br />
==J==<br />
<br />
* [[Joint stereo]]<br />
<br />
==L==<br />
<br />
* [[LFE]] (Low Frequency Extension)<br />
* [[LPAC]]<br />
* [[LPC]] (Linear Prediction Coding)<br />
* [[Long block]]<br />
* [[Lossless]]<br />
* [[Lossy]]<br />
* [[Lowpass]]<br />
* [[ATH|LTQ]] (Level of Threshold in Quiet)<br />
<br />
==M==<br />
<br />
* [[Masking]]<br />
* [[MDCT]]<br />
* [[Metadata]]<br />
* [[Mid-side stereo]]<br />
* [[Monkey's Audio]]<br />
* [[MP3]] (MPEG 1 Audio Layer 3)<br />
* [[MP3Pro]]<br />
* [[Musepack|MPC]]/[[Musepack|MP+]]/[[Musepack]]/[[Musepack|Mpeg Plus]]<br />
* [[Mpeg]] (Motion Picture Expert Group)<br />
* [[MPEG-4]]<br />
* [[Multichannel]]<br />
<br />
==N==<br />
<br />
* [[Neural network]]<br />
* [[Noise shaping]]<br />
* [[Noise normalization]] <br />
* [[Notch filter]]<br />
* [[Nyquist rate]]/[[Nyquist frequency]]/[[Nyquist sampling theorem]]<br />
<br />
==O==<br />
<br />
* [[Ogg]]<br />
* [[Ogg Vorbis]]<br />
* [[OptimFROG]]<br />
<br />
==P==<br />
<br />
* [[Parametric stereo]]/[[PS]]<br />
* [[PCM]]<br />
* [[Point stereo]] <br />
* [[Pre echo]]/[[Pre echo|Post echo]]<br />
* [[Psychoacoustic]]<br />
* [[Psychoacoustic#Psychoacoustic model|Psychoacoustic model]]<br />
* [[Pre-masking]]/[[Pre-masking|Post-masking]]<br />
<br />
==Q==<br />
<br />
* [[Quantize]]/[[Quantizer]]/[[Quantization]]<br />
* [[Quantization noise]]<br />
<br />
==R==<br />
<br />
* [[Range coding]]<br />
* [[Resampling]]<br />
* [[Rice coding]]<br />
* [[Ringing]]<br />
<br />
==S==<br />
<br />
* [[Sampling rate]]/[[Sampling rate|Sample rate]]/[[Sampling rate|Sampling frequency]]<br />
* [[SBR]] (Spectral Band Replication)<br />
* [[Scale factor]]/[[Scale factor|Scale factor band]]<br />
* [[Short block]]<br />
* [[Shorten]] ([[SHN]])<br />
* [[Sigma Delta Modulation]]<br />
* [[SNR]] (Signal-to-Noise Ratio)<br />
* [[Spectrogram]]<br />
* [[S/PDIF]] (Sony-Phillips Digital Interface)<br />
* [[Streaming]]<br />
* [[Subband]]<br />
* [[SZIP]]<br />
<br />
==T==<br />
<br />
* [[TTA]] '''T'''rue '''T'''ap '''A'''udio<br />
* [[Temporal accuracy]]<br />
* [[Temporal smearing]]<br />
* [[Temporal Noise Shaping]]<br />
* [[Time domain]]<br />
* [[Temporal Noise Shaping|TNS]]<br />
* [[Tonality]]/[[Tonality|Tonal signals]]/[[Tonality|Tonality estimation]]<br />
* [[Transcoding]]<br />
* [[Transform]]<br />
* [[Transient]]<br />
* [[Transient smearing]]<br />
* [[Transparency]]<br />
* [[Tremor]]<br />
<br />
==V==<br />
<br />
* [[VBR]] (Variable Bitrate/Variable Bitrate Coding)<br />
* [[Vector quantization]]<br />
<br />
==W==<br />
<br />
* [[WAV]]<br />
* [[Wavelet]]s<br />
* [[WavPack]]<br />
* [[Window function]] <br />
* [[WMA]]<br />
<br />
<br />
[[Category:Guides]]</div>Gigapodhttps://wiki.hydrogenaudio.org/index.php?title=Transparency&diff=15610Transparency2006-11-26T11:57:16Z<p>Gigapod: </p>
<hr />
<div>In [[psychoacoustics]], '''transparency''' is the ideal result of [[lossy]] data compression. If a lossily compressed result is perceptually indistinguishible from the uncompressed input, then the compression can be declared to be transparent. In other words, transparency is the situation where [[artifact]]s are nonexistant or imperceptible.<br />
<br />
Transparency, like sound quality, is subjective. It depends most on the listener's familiarity with artifacts, and to a lesser extent, the compression method, [[bitrate]] used, input characteristics, listening conditions, and listening equipment. It also depends, of course, on the particular music piece or sound under examination. Some audio samples, when compressed with certain algorithms under certain conditions, are known to cause artifacts of a specific kind (see, for example, the samples from the [http://lame.sourceforge.net/quality.php Lame MP3 Encoder Quality and Listening Test Information web page]).<br />
<br />
Judging transparency can be difficult due to observation bias, in which subjective like/dislike of a certain compression methodology emotionally influences his/her judgment. This bias is commonly referred to as ''placebo,'' although this use is slightly different from the medical use of the term.<br />
<br />
By definition, there is no way to prove whether a certain compression methodology is transparent, since transparency is a perceptual, subjective phenomenon. To scientifically prove that a compression method is ''not'' transparent, one should either show that the particular compression method results in a measurable and easily perceived artifact, or use double-blind tests where the high anchor is the original uncompressed audio. For such tests the [[ABX]] method is normally used, and the audio samples should be chosen so as to make the detection of artifacts more obvious.<br />
<br />
Non-lossy compression algorithms are assumed to be transparent: a non-lossy compression methodology should never introduce any artifact.<br />
<br />
Transparency at the lowest possible bitrate also seems to be used as a measure of the quality or degree of sophistication and tuning of a lossy compression algorithm:<br />
* MP3-encoded files are generally considered artifact-free at bitrates at/above 192kbps.<br />
* Vorbis ogg files are supposedly artifact-free at bitrates at/above 160kbps.<br />
* AAC-encoded files, depending on the particular encoder implementation, are claimed to be artifact-free at lower bitrates than both Vorbis ogg and MP3.<br />
<br />
With the present competition between compression formats (open and proprietary) and encoder implementations (GPL'd, proprietary), any claims of transparency at any given bitrate should always be considered with some skepticism.<br />
<br />
[[Category:Listening Tests]]</div>Gigapodhttps://wiki.hydrogenaudio.org/index.php?title=Transparency&diff=15609Transparency2006-11-26T11:54:48Z<p>Gigapod: </p>
<hr />
<div>In [[psychoacoustics]], '''transparency''' is the ideal result of [[lossy]] data compression. If a lossily compressed result is perceptually indistinguishible from the uncompressed input, then the compression can be declared to be transparent. In other words, transparency is the situation where [[artifact]]s are nonexistant or imperceptible.<br />
<br />
Transparency, like sound quality, is subjective. It depends most on the listener's familiarity with artifacts, and to a lesser extent, the compression method, [[bitrate]] used, input characteristics, listening conditions, and listening equipment. It also depends, of course, on the particular music piece or sound under examination. Some audio samples, when compressed with certain algorithms under certain conditions, are known to cause artifacts of a specific kind (see, for example, the samples from the [http://lame.sourceforge.net/quality.php Lame MP3 Encoder Quality and Listening Test Information web page]).<br />
<br />
Judging transparency can be difficult due to observation bias, in which subjective like/dislike of a certain compression methodology emotionally influences his/her judgment. This bias is commonly referred to as ''placebo,'' although this use is slightly different from the medical use of the term.<br />
<br />
By definition, there is no way to prove whether a certain compression methodology is transparent, since transparency is a perceptual, subjective phenomenon. To scientifically prove that a compression method is ''not'' transparent, one should either show that the particular compression method causes a given measurable and easily perceived artifact, or use double-blind tests where the high-anchor is the original uncompressed audio. The [[ABX]] method is normally used, and the audio samples should be chosen so as to make the detection of artifacts more obvious.<br />
<br />
Non-lossy compression algorithms are assumed to be transparent: a non-lossy compression methodology should never introduce any artifact.<br />
<br />
Transparency at the lowest possible bitrate also seems to be used as a measure of the quality or degree of sophistication and tuning of a lossy compression algorithm:<br />
* MP3-encoded files are generally considered artifact-free at bitrates at/above 192kbps.<br />
* Vorbis ogg files are supposedly artifact-free at bitrates at/above 160kbps.<br />
* AAC-encoded files, depending on the particular encoder implementation, are claimed to be artifact-free at lower bitrates than both Vorbis ogg and MP3.<br />
<br />
With the present competition between compression formats (open and proprietary) and encoder implementations (GPL'd, proprietary), any claims of transparency at any given bitrate should always be considered with some skepticism.<br />
<br />
[[Category:Listening Tests]]</div>Gigapodhttps://wiki.hydrogenaudio.org/index.php?title=Transparency&diff=15608Transparency2006-11-26T11:49:17Z<p>Gigapod: </p>
<hr />
<div>In [[psychoacoustics]], '''transparency''' is the ideal result of [[lossy]] data compression. If a lossily compressed result is perceptually indistinguishible from the uncompressed input, then the compression can be declared to be transparent. In other words, transparency is the situation where [[artifact]]s are nonexistant or imperceptible.<br />
<br />
Transparency, like sound quality, is subjective. It depends most on the listener's familiarity with artifacts, and to a lesser extent, the compression method, [[bitrate]] used, input characteristics, listening conditions, and listening equipment. It also depends, of course, on the particular music piece or sound under examination. Some audio samples, when compressed with certain algorithms under certain conditions, are known to cause artifacts of a specific kind (see, for example, the samples from the [http://lame.sourceforge.net/quality.php Lame MP3 Encoder Quality and Listening Test Information web page]).<br />
<br />
Judging transparency can be difficult due to observation bias, in which subjective like/dislike of a certain compression methodology emotionally influences his/her judgment. This bias is commonly referred to as ''placebo,'' although this use is slightly different from the medical use of the term.<br />
<br />
By definition, there is no way to prove whether a certain compression methodology is transparent, since transparency is a perceptual, subjective phenomenon. To scientifically prove that a compression method is ''not'' transparent, one should either show that the particular compression method causes a given measurable and easily perceived artifact, or use double-blind tests where the high-anchor is the original uncompressed audio. The [[ABX]] method is normally used, and the audio samples should be chosen so as to make the detection of artifacts more obvious.<br />
<br />
Non-lossy compression algorithms are assumed to be transparent: a non-lossy compression methodology should never introduce any artifact.<br />
<br />
Transparency at the lowest possible bitrate also seems to be used as a measure of the quality or degree of sophistication and tuning of a lossy compression algorithm:<br />
* MP3-encoded files are generally considered artifact-free at bitrates at/above 192kbps.<br />
* Vorbis ogg files are supposedly artifact-free at bitrates at/above 160kbps.<br />
* AAC-encoded files, depending on the particular encoder implementation, are claimed to be artifact-free at lower bitrates than both Vorbis ogg and MP3.<br />
<br />
With the present competition between compression formats (open and proprietary) and encoder implementations (GPL'd, proprietary), any claims of transparency at any given bitrate should always be taken with some skepticism.<br />
<br />
[[Category:Listening Tests]]</div>Gigapodhttps://wiki.hydrogenaudio.org/index.php?title=Transparency&diff=15551Transparency2006-11-25T11:59:30Z<p>Gigapod: </p>
<hr />
<div>In [[psychoacoustics]], '''transparency''' is the ideal result of [[lossy]] data compression. If a lossily compressed result is perceptually indistinguishible from the uncompressed input, then the compression can be declared to be transparent. In other words, transparency is the situation where [[artifact]]s are nonexistant or imperceptible.<br />
<br />
Transparency, like sound quality, is subjective. It depends most on the listener's familiarity with artifacts, and to a lesser extent, the compression method, [[bitrate]] used, input characteristics, listening conditions, and listening equipment. It also depends, of course, on the particular music piece or sound under examination. Some audio samples, when compressed with certain algorithms under certain conditions, are known to cause artifacts of a specific kind (see, for example, the samples from the [http://lame.sourceforge.net/quality.php Lame MP3 Encoder Quality and Listening Test Information web page]).<br />
<br />
Judging transparency can be difficult due to observation bias, in which subjective like/dislike of a certain compression methodology emotionally influences his/her judgment. This bias is commonly referred to as ''placebo,'' although this use is slightly different from the medical use of the term.<br />
<br />
By definition, there is no way to prove whether a certain compression methodology is transparent, since transparency is a perceptual, subjective phenomenon. To scientifically prove that a compression method is ''not'' transparent, one should either show that the particular compression method causes a given measurable and easily perceived artifact, or use double-blind tests where the high-anchor is the original uncompressed audio. The [[ABX]] method is normally used, and the audio samples should be chosen so as to make the detection of artifacts more obvious.<br />
<br />
Non-lossy compression algorithms are assumed to be transparent: a non-lossy compression methodology should never introduce any artifact.<br />
<br />
Transparency at the lowest possible bitrate also seems to be used as a measure of the quality or degree of sophistication and tuning of a compression algorithm:<br />
* MP3-encoded files are generally considered artifact-free at bitrates at/above 192kbps.<br />
* Vorbis ogg files are supposedly artifact-free at bitrates at/above 160kbps.<br />
* AAC-encoded files, depending on the particular encoder implementation, are claimed to be artifact-free at lower bitrates than both Vorbis ogg and MP3.<br />
<br />
With the present competition between compression formats (open and proprietary) and encoder implementations (GPL'd, proprietary), any claims of transparency at any given bitrate should always be taken with a grain of salt.<br />
<br />
[[Category:Listening Tests]]</div>Gigapodhttps://wiki.hydrogenaudio.org/index.php?title=Transparency&diff=15550Transparency2006-11-25T11:58:33Z<p>Gigapod: </p>
<hr />
<div>In [[psychoacoustics]], '''transparency''' is the ideal result of [[lossy]] data compression. If a lossily compressed result is perceptually indistinguishible from the uncompressed input, then the compression can be declared to be transparent. In other words, transparency is the situation where [[artifact]]s are nonexistant or imperceptible.<br />
<br />
Transparency, like sound quality, is subjective. It depends most on the listener's familiarity with artifacts, and to a lesser extent, the compression method, [[bitrate]] used, input characteristics, listening conditions, and listening equipment. It also depends, of course, on the particular music piece or sound under examination. Some audio samples, when compressed with certain algorithms under certain conditions, are known to cause artifacts of a specific kind (see, for example, the samples from the [http://lame.sourceforge.net/quality.php Lame MP3 Encoder Quality and Listening Test Information web page]).<br />
<br />
Judging transparency can be difficult due to observation bias, in which subjective like/dislike of a certain compression methodology emotionally influences his/her judgment. This bias is commonly referred to as ''placebo,'' although this use is slightly different from the medical use of the term.<br />
<br />
By definition, there is no way to prove whether a certain compression methodology is transparent, since transparency is a perceptual, subjective phenomenon. To scientifically prove that a compression method is ''not'' transparent, one should either show that the particular compression method causes a given measurable and easily perceived artifact, or use double-blind tests where the high-anchor is the original uncompressed audio. The [[ABX]] method is normally used, and the audio samples should be chosen so as to make the detection of artifacts more obvious.<br />
<br />
Non-lossy compression algorithms are assumed to be transparent: a non-lossy compression methodology should never introduce any artifact.<br />
<br />
Transparency at the lowest possible bitrate also seems to be used as a measure of the quality or degree of sophistication and tuning of a compression algorithm:<br />
* MP3-encoded files are generally considered artifact-free at bitrates at/above 192kbps.<br />
* Vorbis ogg files are supposedly artifact-free at bitrates at/above 160kbps.<br />
* AAC-encoded files, depending on the particular encoder implementation, are claimed to be artifact-free at lower bitrates than both Vorbis ogg and MP3.<br />
<br />
With the present competition between compression formats (open and proprietary) and encoder implementations (GPL'd, proprietary), any claims of transparency at any given bitrate should always be taken with a grain of salt.<br />
{{stub}}<br />
<br />
<br />
[[Category:Listening Tests]]</div>Gigapodhttps://wiki.hydrogenaudio.org/index.php?title=Transparency&diff=15549Transparency2006-11-25T11:16:35Z<p>Gigapod: </p>
<hr />
<div>In [[psychoacoustics]], '''transparency''' is the ideal result of [[lossy]] data compression. If a lossily compressed result is perceptually indistinguishible from the uncompressed input, then the compression can be declared to be transparent. In other words, transparency is the situation where [[artifact]]s are nonexistant or imperceptible.<br />
<br />
Transparency, like sound quality, is subjective. It depends most on the listener's familiarity with artifacts, and to a lesser extent, the compression method, [[bitrate]] used, input characteristics, listening conditions, and listening equipment. It also depends, of course, on the particular music piece or sound under examination. Some audio samples, when compressed with certain algorithms under certain conditions, are known to cause artifacts of a specific kind (see, for example, the samples from the [http://lame.sourceforge.net/quality.php Lame MP3 Encoder Quality and Listening Test Information web page]).<br />
<br />
Judging transparency can be difficult due to observation bias, in which subjective like/dislike of a certain compression methodology emotionally influences his/her judgment. This bias is commonly referred to as ''placebo,'' although this use is slightly different from the medical use of the term.<br />
<br />
By definition, there is no way to prove whether a certain compression methodology is transparent, since transparency is a perceptual, subjective phenomenon. To scientifically prove that a compression method is ''not'' transparent, one should either show that the particular compression method causes a given measurable and easily perceived artifact, or use double-blind tests where the high-anchor is the original uncompressed audio. The [[ABX]] method is normally used, and the audio samples should be chosen so as to make the detection of artifacts more obvious.<br />
<br />
Non-lossy compression algorithms are assumed to be transparent: a non-lossy compression methodology should never introduce any artifact.<br />
<br />
Transparency at the lowest possible bitrate also seems to be used as a measure of the quality or degree of sophistication and tuning of a compression algorithm:<br />
* MP3-encoded files are generally considered artifact-free at bitrates at/above 192kbps.<br />
* Vorbis ogg files are supposedly artifact-free at bitrates at/above 160kbps.<br />
* AAC-encoded files, depending on the particular encoder implementation, are claimed to be artifact-free at lower bitrates than both Vorbis ogg and MP3.<br />
<br />
With a certain amount of competition between compression formats (open and proprietary) and encoder implementations (GPL'd, proprietary), any claims of transparency at any given bitrate should always be taken with a grain of salt.<br />
{{stub}}<br />
<br />
<br />
[[Category:Listening Tests]]</div>Gigapodhttps://wiki.hydrogenaudio.org/index.php?title=Transparency&diff=15548Transparency2006-11-25T11:12:54Z<p>Gigapod: </p>
<hr />
<div>In [[psychoacoustics]], '''transparency''' is the ideal result of [[lossy]] data compression. If a lossily compressed result is perceptually indistinguishible from the uncompressed input, then the compression can be declared to be transparent. In other words, transparency is the situation where [[artifact]]s are nonexistant or imperceptible.<br />
<br />
Transparency, like sound quality, is subjective. It depends most on the listener's familiarity with artifacts, and to a lesser extent, the compression method, [[bitrate]] used, input characteristics, listening conditions, and listening equipment. It also depends, of course, on the particular music piece or sound under examination. Some sounds, when compressed with certain algorithms under certain conditions, are known to cause artifacts of a specific kind.<br />
<br />
Judging transparency can be difficult due to observation bias, in which subjective like/dislike of a certain compression methodology emotionally influences his/her judgment. This bias is commonly referred to as ''placebo,'' although this use is slightly different from the medical use of the term.<br />
<br />
By definition, there is no way to prove whether a certain compression methodology is transparent, since transparency is a perceptual, subjective phenomenon. To scientifically prove that a compression method is ''not'' transparent, one should either show that the particular compression method causes a given measurable and easily perceived artifact, or use double-blind tests where the high-anchor is the original uncompressed audio. The [[ABX]] method is normally used, and the audio samples should be chosen so as to make the detection of artifacts more obvious (see, for example, the [http://lame.sourceforge.net/quality.php Lame MP3 Encoder Quality and Listening Test Information web page]).<br />
<br />
Non-lossy compression algorithms are assumed to be transparent: a non-lossy compression methodology should never introduce any artifact.<br />
<br />
Transparency at the lowest possible bitrate also seems to be used as a measure of the quality or degree of sophistication and tuning of a compression algorithm:<br />
* MP3-encoded files are generally considered artifact-free at bitrates at/above 192kbps.<br />
* Vorbis ogg files are supposedly artifact-free at bitrates at/above 160kbps.<br />
* AAC-encoded files, depending on the particular encoder implementation, are claimed to be artifact-free at lower bitrates than both Vorbis ogg and MP3.<br />
<br />
With a certain amount of competition between compression formats (open and proprietary) and encoder implementations (GPL'd, proprietary), any claims of transparency at any given bitrate should always be taken with a grain of salt.<br />
{{stub}}<br />
<br />
<br />
[[Category:Listening Tests]]</div>Gigapodhttps://wiki.hydrogenaudio.org/index.php?title=Transparency&diff=15547Transparency2006-11-25T09:43:14Z<p>Gigapod: </p>
<hr />
<div>In [[psychoacoustics]], '''transparency''' is the ideal result of [[lossy]] data compression. If a lossily compressed result is perceptually indistinguishible from the uncompressed input, then the compression can be declared to be transparent. In other words, transparency is the situation where [[artifact]]s are nonexistant or imperceptible.<br />
<br />
Transparency, like sound quality, is subjective. It depends most on the listener's familiarity with artifacts, and to a lesser extent, the compression method, [[bitrate]] used, input characteristics, listening conditions, and listening equipment. It also depends, of course, on the particular music piece or sound under examination. Some sounds, when compressed with certain algorithms under certain conditions, are known to cause artifacts of a specific kind.<br />
<br />
Judging transparency can be difficult due to observation bias, in which subjective like/dislike of a certain compression methodology emotionally influences his/her judgment. This bias is commonly referred to as ''placebo,'' although this use is slightly different from the medical use of the term.<br />
<br />
By definition, there is no way to prove whether a certain compression methodology is transparent, since transparency is a perceptual, subjective phenomenon. To scientifically prove that a compression method is ''not'' transparent, one should either show that the particular compression method causes a given measurable and easily perceived artifact, or use double-blind tests where the high-anchor is the original uncompressed audio. The [[ABX]] method is normally used, and the audio samples should be chosen so as to make the detection of artifacts more obvious.<br />
<br />
Non-lossy compression algorithms are assumed to be transparent: a non-lossy compression methodology should never introduce any artifact.<br />
<br />
Transparency at the lowest possible bitrate also seems to be used as a measure of the quality or degree of sophistication and tuning of a compression algorithm:<br />
* MP3-encoded files are generally considered artifact-free at bitrates at/above 192kbps.<br />
* Vorbis ogg files are supposedly artifact-free at bitrates at/above 160kbps.<br />
* AAC-encoded files, depending on the particular encoder implementation, are claimed to be artifact-free at lower bitrates than both Vorbis ogg and MP3.<br />
<br />
With a certain amount of competition between compression formats (open and proprietary) and encoder implementations (GPL'd, proprietary), any claims of transparency at any given bitrate should always be taken with a grain of salt.<br />
{{stub}}<br />
<br />
<br />
[[Category:Listening Tests]]</div>Gigapodhttps://wiki.hydrogenaudio.org/index.php?title=Resampling&diff=15508Resampling2006-11-23T09:22:40Z<p>Gigapod: /* References */</p>
<hr />
<div>==Resampling or Sample Rate Conversion==<br />
<br />
Digital audio is always sampled, which means that any digital audio file is created with a fixed sample rate (and resolution). Redbook Audio CD's sample rate is 44.1kHz (16-bits resolution). Audio on DVDs is sampled at 96kHz (24-bit resolution). Resampling or Sample Rate Conversion is required when one wants to convert a digital audio file (i.e. an analog audio signal that has already been digitized) from a given sample rate into a different sample rate (resolution can stay the same or change). Upsampling (aka interpolation) is the process of converting from a lower to higher sample rate (e.g. from 44.1kHz to 48kHz); downsampling (aka decimation) is the process of converting from a higher to a lower sample rate (e.g. from 96kHz to 48kHz).<br />
<br />
Resampling can also be used to describe resolution changes (changes in bit depth, e.g. from 16-bit to 24-bit, or from 24-bit to 16-bit).<br />
<br />
Low-quality resampling algorithms, whether upsampling or downsampling, can introduce artifacts which are clearly audible in the resampled audio file. A typical low-quality, but extremely fast resampling algorithm will just be based on linear interpolation. High-quality resampling algorithms use more CPU time, since they require tranlation to the frequency domain. Modern PC processors (~2GHz clock) can easily deal with very high-quality resampling in real time. Sound cards that do resampling in real time require a good DSP.<br />
<br />
Resampling is very often required and is in fact part of the audio mastering process for CDs, since professional audio equipment uses 96kHz or 192kHz for masters, whereas the RedBook Audio CD spec uses a 44.1kHz sample rate. Different media are recorded at different sample rates (CD at 44.1kHz, DAT at 48kHz, DVD audio at 96kHz, etc). Digitally mixing different sources sampled at different rates will require resampling to a common rate and resolution.<br />
<br />
Many PC audio cards (most notably the 10k1 and 10k2 based Creative Labs ones) and AC97 codecs can only input, output or internally process audio data at 48kHz and forcefully resample any digital audio data at one stage or another. Sometimes the audio software or the drivers will add a resampling step (e.g. ALSA when software mixing, see this thread: [http://www.hydrogenaudio.org/forums/index.php?showtopic=47591 ALSA sample rate conversion, ALSA uses a poor-quality linear interpolator]).<br />
<br />
==References==<br />
* Digital Audio Resampling Home Page: http://ccrma-www.stanford.edu/~jos/resample/<br />
* PeakPro 5 Sample Rate Converter Comparison with Other Audio Applications, Bias Inc., December 2005: http://www.bias-inc.com/products/peakPro5/resampling/peakResamplingWhitePaper.pdf<br />
* Sample Rate Conversion Comparisons (96kHz to 44.1kHz): http://src.infinitewave.ca/<br />
* iZotope 64-bit SRC Precise Sample Rate Conversion: http://www.izotope.com/tech/src/<br />
* Sox Sampling rate conversion. W. G. Unruh 2006: http://axion.physics.ubc.ca/soundcard/resample.html<br />
* An Analysis of Sample Rate Conversion in Sox, Andreas Wilde, 19. Dec. 2003: http://www.leute.server.de/wilde/resample.html<br />
* Lyons, Richard G. Understanding Digital Signal Processing. Indiana: Prentice Hall, March 2004: Edition: 3rd <code><nowiki>ISBN 0-13-108989-7</nowiki></code> <br />
* Secret Rabbit Code (aka libsamplerate): http://www.mega-nerd.com/SRC/index.html<br />
<br />
[[Category:Signal Processing]]</div>Gigapodhttps://wiki.hydrogenaudio.org/index.php?title=Guide_aac_kaudiocreator&diff=15290Guide aac kaudiocreator2006-11-11T19:40:19Z<p>Gigapod: </p>
<hr />
<div>In KAudioCreator (the standard KDE CD ripper), you can configure the encoder part to call FAAC, the Free Advanced Audio Coder.<br />
<br />
In the Settings menu, choose Configure KAudioCreator, and open the Encoder pane. Click the Add button, and point to the faac binary. Then click on the Configure button and configure the name of the encoder as FAAC, the Command Line as "faac -o %o -q 150 -c 22000 -w --artist %{artist} --title %{title} --year %{year} --album %{albumtitle} --track %{number} %f"<br />
and the Extension as m4a.<br />
<br />
Adjust the -q (quality) and -c (cutoff frequency) parameters to suit your usual requirements.<br />
<br />
Now you are all set to rip CDs directly to AAC files.</div>Gigapodhttps://wiki.hydrogenaudio.org/index.php?title=Guide_aac_kaudiocreator&diff=15289Guide aac kaudiocreator2006-11-11T19:39:28Z<p>Gigapod: Configuring KAudioCreator to use FAAC</p>
<hr />
<div>In KAudioCreator (the standard KDE CD ripper), you can configure the encoder part to call FAAC, the Free Advanced Audio Coder.<br />
<br />
In the Settings menu, choose Configure KAudioCreator, and open the Encoder pane. Click the Add button, and point to the faac binary. Then click on the Configure button and configure the name of the encoder as FAAC, the Coomand Line as "faac -o %o -q 150 -c 22000 -w --artist %{artist} --title %{title} --year %{year} --album %{albumtitle} --track %{number} %f<br />
and the Extension as m4a.<br />
<br />
Adjust the -q (quality) and -c (cutoff frequency) parameters to suit your usual requirements.<br />
<br />
Now you are all set to rip CDs directly to AAC files.</div>Gigapodhttps://wiki.hydrogenaudio.org/index.php?title=AAC_Guide&diff=15288AAC Guide2006-11-11T19:30:43Z<p>Gigapod: /* Ripping from CD directly to AAC */</p>
<hr />
<div>The purpose of this guide is helping you create AAC/MP4 files the most easy and quick way possible.<br />
<br />
==Ripping from CD directly to AAC==<br />
<br />
# Using [[Guide aac itunes ripping|Apple iTunes]]<br />
# Using [[Guide aac cdex|CDex]] + Psytel AACenc or NAACenc<br />
# Using [[Guide aac eac|EAC]] + Psytel AACenc or NAACenc<br />
# Using [[Guide aac kaudiocreator|KAudioCreator]] + FAAC<br />
<br />
==Encoding to AAC==<br />
<br />
# Using [[Guide aac itunes|Apple iTunes]]<br />
# Using [[Nero_AAC]] (or NAACenc)<br />
# Using [[FAAC]]<br />
# Using [[Guide aac psytel|Psytel AACenc and Fastenc]]<br />
# Using [[Guide aac squeeze|Sorenson Squeeze]]<br />
<br />
==Transcoding from other formats to AAC==<br />
''(of course, conversion from lossy formats is never recommended. The idea of this guide is to help people convert from lossless formats. i.e [[ALAC]])''<br />
<br />
# Using [[Guide aac foobar|Foobar2000]] + foo_neroaac<br />
# Using [[Guide aac nero plugins|Ahead Nero]] + Mausau's plugins<br />
<br />
==Wrapping AAC streams in MP4==<br />
<br />
# Using [[Guide aac mp4ui|MP4ui]]<br />
# Using [[Guide aac mp4creator|MPEG4ip MP4creator]]<br />
<br />
==Appendix==<br />
<br />
# [[Playback aac|Playing back AAC files]]<br />
<br />
<br />
[[Category:Guides]]</div>Gigapodhttps://wiki.hydrogenaudio.org/index.php?title=Resampling&diff=15220Resampling2006-11-08T18:58:42Z<p>Gigapod: </p>
<hr />
<div>==Resampling or Sample Rate Conversion==<br />
<br />
Digital audio is always sampled, which means that any digital audio file is created with a fixed sample rate (and resolution). Redbook Audio CD's sample rate is 44.1kHz (16-bits resolution). Audio on DVDs is sampled at 96kHz (24-bit resolution). Resampling or Sample Rate Conversion is required when one wants to convert a digital audio file (i.e. an analog audio signal that has already been digitized) from a given sample rate into a different sample rate (resolution can stay the same or change). Upsampling (aka interpolation) is the process of converting from a lower to higher sample rate (e.g. from 44.1kHz to 48kHz); downsampling (aka decimation) is the process of converting from a higher to a lower sample rate (e.g. from 96kHz to 48kHz).<br />
<br />
Resampling can also be used to describe resolution changes (changes in bit depth, e.g. from 16-bit to 24-bit, or from 24-bit to 16-bit).<br />
<br />
Low-quality resampling algorithms, whether upsampling or downsampling, can introduce artifacts which are clearly audible in the resampled audio file. A typical low-quality, but extremely fast resampling algorithm will just be based on linear interpolation. High-quality resampling algorithms use more CPU time, since they require tranlation to the frequency domain. Modern PC processors (~2GHz clock) can easily deal with very high-quality resampling in real time. Sound cards that do resampling in real time require a good DSP.<br />
<br />
Resampling is very often required and is in fact part of the audio mastering process for CDs, since professional audio equipment uses 96kHz or 192kHz for masters, whereas the RedBook Audio CD spec uses a 44.1kHz sample rate. Different media are recorded at different sample rates (CD at 44.1kHz, DAT at 48kHz, DVD audio at 96kHz, etc). Digitally mixing different sources sampled at different rates will require resampling to a common rate and resolution.<br />
<br />
Many PC audio cards (most notably the 10k1 and 10k2 based Creative Labs ones) and AC97 codecs can only input, output or internally process audio data at 48kHz and forcefully resample any digital audio data at one stage or another. Sometimes the audio software or the drivers will add a resampling step (e.g. ALSA when software mixing, see this thread: [http://www.hydrogenaudio.org/forums/index.php?showtopic=47591 ALSA sample rate conversion, ALSA uses a poor-quality linear interpolator]).<br />
<br />
==References==<br />
* PeakPro 5 Sample Rate Converter Comparison with Other Audio Applications, Bias Inc., December 2005: http://www.bias-inc.com/products/peakPro5/resampling/peakResamplingWhitePaper.pdf<br />
* Sample Rate Conversion Comparisons (96kHz to 44.1kHz): http://src.infinitewave.ca/<br />
* iZotope 64-bit SRC Precise Sample Rate Conversion: http://www.izotope.com/tech/src/<br />
* Sox Sampling rate conversion. W. G. Unruh 2006: http://axion.physics.ubc.ca/soundcard/resample.html<br />
* An Analysis of Sample Rate Conversion in Sox, Andreas Wilde, 19. Dec. 2003: http://www.leute.server.de/wilde/resample.html<br />
* Understanding Digital Signal Processing, Second Edition, Richard G. Lyons, 2004<br />
* Secret Rabbit Code (aka libsamplerate): http://www.mega-nerd.com/SRC/index.html</div>Gigapodhttps://wiki.hydrogenaudio.org/index.php?title=Resampling&diff=15219Resampling2006-11-08T18:54:59Z<p>Gigapod: </p>
<hr />
<div>==Resampling or Sample Rate Conversion==<br />
<br />
Digital audio is always sampled, which means that any digital audio file is created with a fixed sample rate (and resolution). Redbook Audio CD's sample rate is 44.1kHz (16-bits resolution). Audio on DVDs is sampled at 96kHz (24-bit resolution). Resampling or Sample Rate Conversion is required when one wants to convert a digital audio file (i.e. an analog audio signal that has already been digitized) from a given sample rate into a different sample rate (resolution can stay the same or change). Upsampling (aka interpolation) is the process of converting from a lower to higher sample rate (e.g. from 44.1kHz to 48kHz); downsampling (aka decimation) is the process of converting from a higher to a lower sample rate (e.g. from 96kHz to 48kHz).<br />
<br />
Low-quality resampling algorithms, whether upsampling or downsampling, can introduce artifacts which are clearly audible in the resampled audio file. A typical low-quality, but extremely fast resampling algorithm will just be based on linear interpolation. High-quality resampling algorithms use more CPU time, since they require tranlation to the frequency domain. Modern PC processors (~2GHz clock) can easily deal with very high-quality resampling in real time. Sound cards that do resampling in real time require a good DSP.<br />
<br />
Resampling is very often required and is in fact part of the audio mastering process for CDs, since professional audio equipment uses 96kHz or 192kHz for masters, whereas the RedBook Audio CD spec uses a 44.1kHz sample rate. Different media are recorded at different sample rates (CD at 44.1kHz, DAT at 48kHz, DVD audio at 96kHz, etc). Digitally mixing different sources sampled at different rates will require resampling to a common rate.<br />
<br />
Many PC audio cards (most notably the 10k1 and 10k2 based Creative Labs ones) and AC97 codecs can only input, output or internally process audio data at 48kHz and forcefully resample any digital audio data at one stage or another. Sometimes the audio software or the drivers will add a resampling step (e.g. ALSA when software mixing, see this thread: [http://www.hydrogenaudio.org/forums/index.php?showtopic=47591 ALSA sample rate conversion, ALSA uses a poor-quality linear interpolator]).<br />
<br />
==References==<br />
* PeakPro 5 Sample Rate Converter Comparison with Other Audio Applications, Bias Inc., December 2005: http://www.bias-inc.com/products/peakPro5/resampling/peakResamplingWhitePaper.pdf<br />
* Sample Rate Conversion Comparisons (96kHz to 44.1kHz): http://src.infinitewave.ca/<br />
* iZotope 64-bit SRC Precise Sample Rate Conversion: http://www.izotope.com/tech/src/<br />
* Sox Sampling rate conversion. W. G. Unruh 2006: http://axion.physics.ubc.ca/soundcard/resample.html<br />
* An Analysis of Sample Rate Conversion in Sox, Andreas Wilde, 19. Dec. 2003: http://www.leute.server.de/wilde/resample.html<br />
* Understanding Digital Signal Processing, Second Edition, Richard G. Lyons, 2004<br />
* Secret Rabbit Code (aka libsamplerate): http://www.mega-nerd.com/SRC/index.html</div>Gigapodhttps://wiki.hydrogenaudio.org/index.php?title=Resampling&diff=15218Resampling2006-11-08T18:54:33Z<p>Gigapod: </p>
<hr />
<div>=Resampling or Sample Rate Conversion=<br />
<br />
Digital audio is always sampled, which means that any digital audio file is created with a fixed sample rate (and resolution). Redbook Audio CD's sample rate is 44.1kHz (16-bits resolution). Audio on DVDs is sampled at 96kHz (24-bit resolution). Resampling or Sample Rate Conversion is required when one wants to convert a digital audio file (i.e. an analog audio signal that has already been digitized) from a given sample rate into a different sample rate (resolution can stay the same or change). Upsampling (aka interpolation) is the process of converting from a lower to higher sample rate (e.g. from 44.1kHz to 48kHz); downsampling (aka decimation) is the process of converting from a higher to a lower sample rate (e.g. from 96kHz to 48kHz).<br />
<br />
Low-quality resampling algorithms, whether upsampling or downsampling, can introduce artifacts which are clearly audible in the resampled audio file. A typical low-quality, but extremely fast resampling algorithm will just be based on linear interpolation. High-quality resampling algorithms use more CPU time, since they require tranlation to the frequency domain. Modern PC processors (~2GHz clock) can easily deal with very high-quality resampling in real time. Sound cards that do resampling in real time require a good DSP.<br />
<br />
Resampling is very often required and is in fact part of the audio mastering process for CDs, since professional audio equipment uses 96kHz or 192kHz for masters, whereas the RedBook Audio CD spec uses a 44.1kHz sample rate. Different media are recorded at different sample rates (CD at 44.1kHz, DAT at 48kHz, DVD audio at 96kHz, etc). Digitally mixing different sources sampled at different rates will require resampling to a common rate.<br />
<br />
Many PC audio cards (most notably the 10k1 and 10k2 based Creative Labs ones) and AC97 codecs can only input, output or internally process audio data at 48kHz and forcefully resample any digital audio data at one stage or another. Sometimes the audio software or the drivers will add a resampling step (e.g. ALSA when software mixing, see this thread: [http://www.hydrogenaudio.org/forums/index.php?showtopic=47591 ALSA sample rate conversion, ALSA uses a poor-quality linear interpolator]).<br />
<br />
=References=<br />
* PeakPro 5 Sample Rate Converter Comparison with Other Audio Applications, Bias Inc., December 2005: http://www.bias-inc.com/products/peakPro5/resampling/peakResamplingWhitePaper.pdf<br />
* Sample Rate Conversion Comparisons (96kHz to 44.1kHz): http://src.infinitewave.ca/<br />
* iZotope 64-bit SRC Precise Sample Rate Conversion: http://www.izotope.com/tech/src/<br />
* Sox Sampling rate conversion. W. G. Unruh 2006: http://axion.physics.ubc.ca/soundcard/resample.html<br />
* An Analysis of Sample Rate Conversion in Sox, Andreas Wilde, 19. Dec. 2003: http://www.leute.server.de/wilde/resample.html<br />
* Understanding Digital Signal Processing, Second Edition, Richard G. Lyons, 2004<br />
* Secret Rabbit Code (aka libsamplerate): http://www.mega-nerd.com/SRC/index.html</div>Gigapodhttps://wiki.hydrogenaudio.org/index.php?title=Resampling&diff=15217Resampling2006-11-08T18:51:07Z<p>Gigapod: What resampling is and some links</p>
<hr />
<div>Resampling or Sample Rate Conversion<br />
<br />
Digital audio is always sampled, which means that any digital audio file is created with a fixed sample rate (and resolution). Redbook Audio CD's sample rate is 44.1kHz (16-bits resolution). Audio on DVDs is sampled at 96kHz (24-bit resolution). Resampling or Sample Rate Conversion is required when one wants to convert a digital audio file (i.e. an analog audio signal that has already been digitized) from a given sample rate into a different sample rate (resolution can stay the same or change). Upsampling (aka interpolation) is the process of converting from a lower to higher sample rate (e.g. from 44.1kHz to 48kHz); downsampling (aka decimation) is the process of converting from a higher to a lower sample rate (e.g. from 96kHz to 48kHz).<br />
<br />
Low-quality resampling algorithms, whether upsampling or downsampling, can introduce artifacts which are clearly audible in the resampled audio file. A typical low-quality, but extremely fast resampling algorithm will just be based on linear interpolation. High-quality resampling algorithms use more CPU time, since they require tranlation to the frequency domain. Modern PC processors (~2GHz clock) can easily deal with very high-quality resampling in real time. Sound cards that do resampling in real time require a good DSP.<br />
<br />
Resampling is very often required and is in fact part of the audio mastering process for CDs, since professional audio equipment uses 96kHz or 192kHz for masters, whereas the RedBook Audio CD spec uses a 44.1kHz sample rate. Different media are recorded at different sample rates (CD at 44.1kHz, DAT at 48kHz, DVD audio at 96kHz, etc). Digitally mixing different sources sampled at different rates will require resampling to a common rate.<br />
<br />
Many PC audio cards (most notably the 10k1 and 10k2 based Creative Labs ones) and AC97 codecs can only input, output or internally process audio data at 48kHz and forcefully resample any digital audio data at one stage or another. Sometimes the audio software or the drivers will add a resampling step (e.g. ALSA when software mixing, see this thread: ALSA sample rate conversion, ALSA uses a poor-quality linear interpolator - http://www.hydrogenaudio.org/forums/index.php?showtopic=47591).<br />
<br />
References:<br />
- PeakPro 5 Sample Rate Converter Comparison with Other Audio Applications, Bias Inc., December 2005: http://www.bias-inc.com/products/peakPro5/resampling/peakResamplingWhitePaper.pdf<br />
- Sample Rate Conversion Comparisons (96kHz to 44.1kHz): http://src.infinitewave.ca/<br />
- iZotope 64-bit SRC Precise Sample Rate Conversion: http://www.izotope.com/tech/src/<br />
- Sox Sampling rate conversion. W. G. Unruh 2006: http://axion.physics.ubc.ca/soundcard/resample.html<br />
- An Analysis of Sample Rate Conversion in Sox, Andreas Wilde, 19. Dec. 2003: http://www.leute.server.de/wilde/resample.html<br />
- Understanding Digital Signal Processing, Second Edition, Richard G. Lyons, 2004<br />
- Secret Rabbit Code (aka libsamplerate): http://www.mega-nerd.com/SRC/index.html</div>Gigapodhttps://wiki.hydrogenaudio.org/index.php?title=Glossary_Of_Audio_Terms&diff=15216Glossary Of Audio Terms2006-11-08T18:48:47Z<p>Gigapod: /* R */</p>
<hr />
<div>__NOTOC__<br />
<br />
==A==<br />
<br />
* [[AAC]]<br />
* [[ABR]]<br />
* [[ABX]]/[[ABX|ABX testing]]<br />
* [[AC3]]<br />
* [[Aliasing]]<br />
* [[AltPresets]]<br />
* [[Ambisonics]] <br />
* [[Amplitude]]<br />
* [[APE]] ([[Monkey's Audio]])<br />
* [[APE Tags]]<br />
* [[Artifact]]/[[Artifact|Distortion]]<br />
* [[ATH]] (Absolute Threshold of Hearing)<br />
* [[ASIO]] (Audio Stream Input Output)<br />
* [[ASPI]] (CD-ROM Installation)<br />
<br />
==B==<br />
<br />
* [[Bandpass filter]]<br />
* [[Bandstop filter]]/[[Bandstop filter|Band reject]]<br />
* [[Bandwidth]]<br />
* [[Bark]]<br />
* [[Bit reservoir]]<br />
* [[Bitrate]]<br />
* [[Bit depth|Bits per sample]] / [[Bit depth]]<br />
* [[Blind test]]<br />
* [[Block switching]]<br />
* [[BSAC]] (Bit-sliced arithmetic coding)<br />
<br />
==C==<br />
<br />
* [[CBR]]/[[Constant bitrate]]/[[CBR|Constant bitrate coding]]<br />
* [[Channel coupling]]<br />
* [[Clipping]]<br />
* [[Codec]]<br />
* [[Container format]]<br />
* [[Critical band]]<br />
* [[C1/C2 errors]]<br />
<br />
==D==<br />
<br />
* [[dB]]/[[dB|Decibel]]<br />
* [[DC coefficient]]<br />
* [[DCT]]<br />
* [[DCT coefficient]]<br />
* [[Digital Radio Mondiale|DRM]] (Digital Radio Mondiale)<br />
* [[Digital Rights Management|DRM]] (Digital Rights Management)<br />
* [[Dither]] <br />
* [[DSP]]<br />
* [[DTS]]<br />
<br />
==F==<br />
<br />
* [[FFT]]<br />
* [[Filterbank]]<br />
* [[FIR filter]] (Finite Impulse Response Filters)<br />
* [[FLAC]]<br />
* [[Frequency]]<br />
* [[Frequency domain]]<br />
<br />
==H==<br />
<br />
* [[Harmonics]]<br />
* [[Highpass]]<br />
* [[Huffman coding]]<br />
* [[HDMI]] (High Definition Multimedia Interface)<br />
<br />
==I==<br />
<br />
* [[ID3]]<br />
* [[IIR filter]] (Infinite Impulse Response Filters)<br />
* [[Impulse]]<br />
* [[Intensity stereo]]<br />
* [[Inverse mix]]<br />
<br />
==J==<br />
<br />
* [[Joint stereo]]<br />
<br />
==L==<br />
<br />
* [[LFE]] (Low Frequency Extension)<br />
* [[LPAC]]<br />
* [[LPC]] (Linear Prediction Coding)<br />
* [[Long block]]<br />
* [[Lossless]]<br />
* [[Lossy]]<br />
* [[Lowpass]]<br />
* [[ATH|LTQ]] (Level of Threshold in Quiet)<br />
<br />
==M==<br />
<br />
* [[Masking]]<br />
* [[MDCT]]<br />
* [[Metadata]]<br />
* [[Mid-side stereo]]<br />
* [[Monkey's Audio]]<br />
* [[MP3]] (MPEG 1 Audio Layer 3)<br />
* [[MP3Pro]]<br />
* [[Musepack|MPC]]/[[Musepack|MP+]]/[[Musepack]]/[[Musepack|Mpeg Plus]]<br />
* [[Mpeg]] (Motion Picture Expert Group)<br />
* [[MPEG-4]]<br />
* [[Multichannel]]<br />
<br />
==N==<br />
<br />
* [[Neural network]]<br />
* [[Noise shaping]]<br />
* [[Noise normalization]] <br />
* [[Notch filter]]<br />
* [[Nyquist rate]]/[[Nyquist frequency]]/[[Nyquist sampling theorem]]<br />
<br />
==O==<br />
<br />
* [[Ogg]]<br />
* [[Ogg Vorbis]]<br />
* [[OptimFROG]]<br />
<br />
==P==<br />
<br />
* [[Parametric stereo]]/[[PS]]<br />
* [[PCM]]<br />
* [[Point stereo]] <br />
* [[Pre echo]]/[[Pre echo|Post echo]]<br />
* [[Psychoacoustic]]<br />
* [[Psychoacoustic#Psychoacoustic model|Psychoacoustic model]]<br />
* [[Pre-masking]]/[[Pre-masking|Post-masking]]<br />
<br />
==Q==<br />
<br />
* [[Quantize]]/[[Quantizer]]/[[Quantization]]<br />
* [[Quantization noise]]<br />
<br />
==R==<br />
<br />
* [[Range coding]]<br />
* [[Resampling]]<br />
* [[Rice coding]]<br />
* [[Ringing]]<br />
<br />
==S==<br />
<br />
* [[Sampling rate]]/[[Sampling rate|Sample rate]]/[[Sampling rate|Sampling frequency]]<br />
* [[SBR]] (Spectral Band Replication)<br />
* [[Scale factor]]/[[Scale factor|Scale factor band]]<br />
* [[Short block]]<br />
* [[Shorten]] ([[SHN]])<br />
* [[Sigma Delta Modulation]]<br />
* [[SNR]] (Signal-to-Noise Ratio)<br />
* [[Spectrogram]]<br />
* [[S/PDIF]] (Sony-Phillips Digital Interface)<br />
* [[Streaming]]<br />
* [[Subband]]<br />
* [[SZIP]]<br />
<br />
==T==<br />
<br />
* [[TTA]] '''T'''rue '''T'''ap '''A'''udio<br />
* [[Temporal accuracy]]<br />
* [[Temporal smearing]]<br />
* [[Time domain]]<br />
* [[Temporary noise shaping|TNS]]<br />
* [[Tonality]]/[[Tonality|Tonal signals]]/[[Tonality|Tonality estimation]]<br />
* [[Transcoding]]<br />
* [[Transform]]<br />
* [[Transient]]<br />
* [[Transient smearing]]<br />
* [[Transparency]]<br />
* [[Tremor]]<br />
<br />
==V==<br />
<br />
* [[VBR]] (Variable Bitrate/Variable Bitrate Coding)<br />
* [[Vector quantization]]<br />
<br />
==W==<br />
<br />
* [[WAV]]<br />
* [[Wavelet]]s<br />
* [[WavPack]]<br />
* [[Window function]] <br />
* [[WMA]]<br />
<br />
<br />
[[Category:Guides]]</div>Gigapod