Hydrogenaudio Knowledgebase - User contributions [en]

Advanced SCSI Programming Interface

2005-05-14T22:32:31Z

Chigung:

ASPI : Advanced SCSI Programming Interface.

This is an established system of accessing Drives (with its origons in 1970's mainframe technology) which today is commonly supported by CDROM reading/ripping softwares. The performance of some Software/OS version combinations can be improved by installing the missing (or an alternative) ASPI driver, which is also sometimes called the ASPI 'Layer'.
Windows 2000 and XP come with ASPI preinstalled and software can come with its own version.
There is normaly little to gain from changing ASPI unless prompted by a specific problem.

Advanced SCSI Programming Interface

2005-05-14T22:30:06Z

Chigung:

ASPI stands for Advanced SCSI Programming Interface.
It is an established system of accessing Drives (with its origons in 1970's mainframe technology) which today is commonly supported by CDROM reading/ripping softwares. The performance of some software/OS version combinations can be improved by installing the missing (or an alternative) ASPI driver, also sometimes called the ASPI 'layer'
Windows 2000 and XP come with ASPI preinstalled and software can come with its own version.
There is normaly little to gain from changing ASPI unless prompted by a specific problem.

Glossary Of Audio Terms

2005-05-14T22:08:58Z

Chigung: /* A */

==A==

* [[AAC]]
* [[ABR]]
* [[ABX]]/[[ABX|ABX testing]]
* [[AC3]]
* [[Aliasing]]
* [[AltPresets]]
* [[Amplitude]]
* [[APE]] ([[Monkey's Audio]])
* [[APE Tags]]
* [[Artifact]]/[[Artifact|Distortion]]
* [[ATH]] (Absolute Threshold of Hearing)
* [[ASPI]] (CD-ROM Installation)

==B==

* [[Bandpass filter]]
* [[Bandstop filter]]/[[Bandstop filter|Band reject]]
* [[Bandwidth]]
* [[Bark]]
* [[Bit reservoir]]
* [[Bitrate]]
* [[Bit depth|Bits per sample]] / [[Bit depth]]
* [[Blind test]]
* [[Block switching]]

==C==

* [[CBR]]/[[Constant bitrate]]/[[CBR|Constant bitrate coding]]
* [[Channel coupling]]
* [[Clipping]]
* [[Codec]]
* [[Container format]]
* [[Critical band]]

==D==

* [[dB]]/[[dB|Decibel]]
* [[DC coefficient]]
* [[DCT]]
* [[DCT coefficient]]
* [[Digital Radio Mondiale|DRM]] (Digital Radio Mondiale)
* [[Digital Rights Management|DRM]] (Digital Rights Management)
* [[DSP]]
* [[DTS]]

==F==

* [[FFT]]
* [[Filterbank]]
* [[FIR filter]] (Finite Impulse Response Filters)
* [[FLAC]]
* [[Frequency]]
* [[Frequency domain]]

==H==

* [[Harmonics]]
* [[Highpass]]
* [[Huffman coding]]

==I==

* [[ID3]]
* [[IIR filter]] (Infinite Impulse Response Filters)
* [[Impulse]]
* [[Intensity stereo]]
* [[Inverse mix]]

==J==

* [[Joint stereo]]

==L==

* [[LFE]] (Low Frequency Extension)
* [[LPAC]]
* [[Long block]]
* [[Lossless]]
* [[Lossy]]
* [[Lowpass]]
* [[ATH|LTQ]] (Level of Threshold in Quiet)

==M==

* [[Masking]]
* [[MDCT]]
* [[Metadata]]
* [[Mid-side stereo]]
* [[Monkey's Audio]]
* [[MP3]] (MPEG 1 Audio Layer 3)
* [[MP3Pro]]
* [[Musepack|MPC]]/[[Musepack|MP+]]/[[Musepack]]/[[Musepack|Mpeg Plus]]
* [[Mpeg]] (Motion Picture Expert Group)
* [[MPEG-4]]
* [[Multichannel]]

==N==

* [[Neural network]]
* [[Noise shaping]]
* [[Noise like signals]]
* [[Notch filter]]
* [[Nyquist rate]]/[[Nyquist frequency]]/[[Nyquist sampling theorem]]

==O==

* [[Ogg]]
* [[Ogg Vorbis]]
* [[OptimFROG]]

==P==

* [[Parametric stereo]]/[[PS]]
* [[PCM]]
* [[Point stereo]]
* [[Pre echo]]/[[Pre echo|Post echo]]
* [[Psychoacoustic]]
* [[Psychoacoustic model]]

==Q==

* [[Quantize]]/[[Quantizer]]/[[Quantization]]
* [[Quantization noise]]

==R==

* [[Range coding]]
* [[Rice coding]]
* [[Ringing]]

==S==

* [[Sampling rate]]/[[Sampling rate|Sample rate]]/[[Sampling rate|Sampling frequency]]
* [[SBR]] (Spectral Band Replication)
* [[Scale factor]]/[[Scale factor|Scale factor band]]
* [[Short block]]
* [[Shorten]] ([[SHN]])
* [[Sigma Delta Modulation]]
* [[Spectrogram]]
* [[Streaming]]
* [[Subband]]
* [[SZIP]]

==T==

* [[TTA]] '''T'''rue '''T'''ap '''A'''udio
* [[Temporal accuracy]]
* [[Temporal smearing]]
* [[Time domain]]
* [[Tonality]]/[[Tonality|Tonal signals]]/[[Tonality|Tonality estimation]]
* [[Transcoding]]
* [[Transform]]
* [[Transient]]
* [[Transient smearing]]
* [[Transparency]]
* [[Tremor]]

==V==

* [[VBR]] (Variable Bitrate/Variable Bitrate Coding)
* [[Vector quantization]]

==W==

* [[WAV]]
* [[Wavelet]]
* [[WavPack]]
* [[WMA]]

Short block

2005-05-10T03:42:57Z

Chigung:

See [[Long block]] and [[Block switching]]

Long block

2005-05-10T03:41:37Z

Chigung:

See also [[Block switching]]

Audio encoding systems will commonly summarise the sound energy in one small stretch of time at a time. The duration of a single summarisation is also called a [[window]] and once a window of sound has been quantised, selected and packed, it can be considered as an individual block of audio information to be finaly stored in a frame.

Available mathematics used to summarise one block at a time can struggle with evenly representing details throughout a window's duration. There may be a tendency to mirror features at the start of the window onto the other end of the window, or the ends of the window might be the most efficient place to have details occur, it will depend on the particular mathematical transformation used.
With such variance in transformation accuracy throughout each block, varying the size of blocks to best fit over the actual details present in the audio stream increases the overall accuracy of the encoding.

[[Lame]] has only two block sizes available, [[Long block]] and [[Short block]]. In Lame's case Short blocks require more data to describe less time than Long blocks so they are only selected when they significantly improve the alignment of transformation windows to audio details.

Long block

2005-05-10T03:41:08Z

Chigung: i added this before i noticed blockswitching, may move it...

See also [[Block Switching]]

Audio encoding systems will commonly summarise the sound energy in one small stretch of time at a time. The duration of a single summarisation is also called a [[window]] and once a window of sound has been quantised, selected and packed, it can be considered as an individual block of audio information to be finaly stored in a frame.

Available mathematics used to summarise one block at a time can struggle with evenly representing details throughout a window's duration. There may be a tendency to mirror features at the start of the window onto the other end of the window, or the ends of the window might be the most efficient place to have details occur, it will depend on the particular mathematical transformation used.
With such variance in transformation accuracy throughout each block, varying the size of blocks to best fit over the actual details present in the audio stream increases the overall accuracy of the encoding.

[[Lame]] has only two block sizes available, [[Long block]] and [[Short block]]. In Lame's case Short blocks require more data to describe less time than Long blocks so they are only selected when they significantly improve the alignment of transformation windows to audio details.

Short block

2005-05-10T03:36:33Z

Chigung:

See [[Long block]]

Short block

2005-05-10T03:35:23Z

Chigung:

See [[Long Block]]

Long block

2005-05-10T03:34:24Z

Chigung:

Audio encoding systems will commonly summarise the sound energy in one small stretch of time at a time. The duration of a single summarisation is also called a [[window]] and once a window of sound has been quantised, selected and packed, it can be considered as an individual block of audio information to be finaly stored in a frame.

Available mathematics used to summarise one block at a time can struggle with evenly representing details throughout a window's duration. There may be a tendency to mirror features at the start of the window onto the other end of the window, or the ends of the window might be the most efficient place to have details occur, it will depend on the particular mathematical transformation used.
With such variance in transformation accuracy throughout each block, varying the size of blocks to best fit over the actual details present in the audio stream increases the overall accuracy of the encoding.

[[Lame]] has only two block sizes available, [[Long block]] and [[Short block]]. In Lame's case Short blocks require more data to describe less time than Long blocks so they are only selected when they significantly improve the alignment of transformation windows to audio details.

Long block

2005-05-10T03:33:43Z

Chigung:

Audio encoding systems will commonly summarise the sound energy in one small stretch of time at a time. The duration of a single summarisation is also called a [[window]] and once a window of sound has been quantised, selected and packed, it can be considered as an individual block of audio information to be finaly stored in a frame.
Available mathematics used to summarise one block at a time can struggle with evenly representing details throughout a window's duration. There may be a tendency to mirror features at the start of the window onto the other end of the window, or the ends of the window might be the most efficient place to have details occur, it will depend on the particular mathematical transformation used.
With such variance in transformation accuracy throughout each block, varying the size of blocks to best fit over the actual details present in the audio stream increases the overall accuracy of the encoding.
[[Lame]] has only two block sizes available, [[Long block]] and [[Short block]]. In Lame's case Short blocks require more data to describe less time than Long blocks so they are only selected when they significantly improve the alignment of transformation windows to audio details.

Bit reservoir

2005-05-09T02:24:48Z

Chigung: vbr does still use bit reservior - try comparing --nores encodes for proof. Also observe max bitreservoir of 511 with encspot (rather than320)

=In MP3:=

[[CBR]] (and also to some degree [[ABR]]) uses a constant defined [[bitrate]]. Because that [[bitrate]] is taken into consideration at every frame, there will be certain moments of such complexity that they can't be properly encoded within the limitations of the chosen [[bitrate]]; they need a higher [[bitrate]] than the defined one. Therefore, the [[MP3]] spec defines a bit reservoir.

Example: a certain moment in a song needs 130 kbit to be accurately encoded (as defined by the psychoacoustic model and the settings of the encoder) the [[CBR]] bitrate is set to 160 kbps. 30 bits are not used (160 - 130 = 30). those bits can be saved for following frames. To limit the streams complexity, the maximum reservoir size is 511 bits however this also limits the ability to cope with sustained complex passages of sound.

With [[VBR]], the encoder can choose the needed framesize for each moment, again as defined by the [[psymodel]] and the quality settings. So [[VBR]] (e.g. in [[LAME]]) doesn't use bit reservoir nearly as much, but still may do to collect bits that would otherwise be wasted to fill an available framesize (eg 160 - 130 = 30 spare bits).

Harmonics

2005-05-09T01:33:39Z

Chigung: see http://www.harmony-central.com/Guitar/harmonics.html for odd order harmonics

Harmonics are vibrations at frequencies that are multiples of the fundamental. They are characterized as even-order and odd- order harmonics. For instance, the "second-order harmonic" is the fundamental [[frequency]] multiplied by two, and is an even-order harmonic. Each even-order harmonic is one octave or x octaves higher than the fundamental; they are therefore musically equivalent to the fundamental. Odd-order harmonics create a series of notes that are musically related to the fundamental [[frequency]] -unparallel but resonant with the fundamental, they inform musical scales and give rise to Chords. Harmonics are also called "overtones" or "partials".