Sub-band coding - Revision history

Kvng: Adding short description: "Lossy audio coding technique"

2025-03-01T16:10:09Z

Adding short description: "Lossy audio coding technique"

← Previous revision		Revision as of 16:10, 1 March 2025
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			{{Short description\|Lossy audio coding technique}}
	{{about\|the signal coding technique\|the Bluetooth audio codec\|SBC (codec)}}		{{about\|the signal coding technique\|the Bluetooth audio codec\|SBC (codec)}}

31.155.219.164: /* Basic principles */Fixed grammatical error

2025-02-03T00:23:36Z

Basic principles: Fixed grammatical error

← Previous revision		Revision as of 00:23, 3 February 2025
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	The basic idea of SBC is to enable a data reduction by discarding information about frequencies which are masked. The result differs from the original signal, but if the discarded information is chosen carefully, the difference will not be noticeable, or more importantly, objectionable.		The basic idea of SBC is to enable a data reduction by discarding information about frequencies which are masked. The result differs from the original signal, but if the discarded information is chosen carefully, the difference will not be noticeable, or more importantly, objectionable.

	First, a digital filter bank divides the input signal spectrum into some number (e.g., 32) of subbands. The psychoacoustic model looks at the energy in each of these subbands, as well as in the original signal, and computes masking thresholds using psychoacoustic information. Each of the subband samples is quantized and encoded so as to keep the quantization noise below the dynamically computed masking threshold. The final step is to format all these quantized samples into groups of data called frames, to facilitate eventual playback by a decoder.		First, a digital filter bank divides the input signal spectrum into some number (e.g., 32) of subbands. The psychoacoustic model looks at the energy in each of these subbands, as well as in the original signal, and computes masking thresholds using psychoacoustic information. Each of the subband samples are quantized and encoded so as to keep the quantization noise below the dynamically computed masking threshold. The final step is to format all these quantized samples into groups of data called frames, to facilitate eventual playback by a decoder.

	Decoding is much easier than encoding, since no psychoacoustic model is involved. The frames are unpacked, subband samples are decoded, and a frequency-time mapping reconstructs an output audio signal.		Decoding is much easier than encoding, since no psychoacoustic model is involved. The frames are unpacked, subband samples are decoded, and a frequency-time mapping reconstructs an output audio signal.

CoolieCoolster: Added non-breaking space to non-template file size, frequency, bitrate, and bandwidth values (via WP:JWB)

2024-04-25T22:41:13Z

Added non-breaking space to non-template file size, frequency, bitrate, and bandwidth values (via WP:JWB)

← Previous revision		Revision as of 22:41, 25 April 2024
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	The more bits used to represent each sample, the finer the granularity in the digital representation, and thus the smaller the quantization error. Such ''quantization errors'' may be thought of as a type of noise, because they are effectively the difference between the original source and its binary representation. With PCM, the audible effects of these errors can be mitigated with [[dither]] and by using enough bits to ensure that the noise is low enough to be masked either by the signal itself or by other sources of noise. A high quality signal is possible, but at the cost of a high [[bitrate]] (e.g., over 700 [[kbit/s]] for one channel of CD audio). In effect, many bits are wasted in encoding masked portions of the signal because PCM makes no assumptions about how the human ear hears.		The more bits used to represent each sample, the finer the granularity in the digital representation, and thus the smaller the quantization error. Such ''quantization errors'' may be thought of as a type of noise, because they are effectively the difference between the original source and its binary representation. With PCM, the audible effects of these errors can be mitigated with [[dither]] and by using enough bits to ensure that the noise is low enough to be masked either by the signal itself or by other sources of noise. A high quality signal is possible, but at the cost of a high [[bitrate]] (e.g., over 700 [[kbit/s]] for one channel of CD audio). In effect, many bits are wasted in encoding masked portions of the signal because PCM makes no assumptions about how the human ear hears.

	Coding techniques reduce bitrate by exploiting known characteristics of the auditory system. A classic method is nonlinear PCM, such as the [[μ-law algorithm]]. Small signals are digitized with finer granularity than are large ones; the effect is to add noise that is proportional to the signal strength. Sun's [[Au file format]] for sound is a popular example of mu-law encoding. Using 8-bit mu-law encoding would cut the per-channel bitrate of CD audio down to about 350 kbit/s, half the standard rate. Because this simple method only minimally exploits masking effects, it produces results that are often audibly inferior compared to the original.		Coding techniques reduce bitrate by exploiting known characteristics of the auditory system. A classic method is nonlinear PCM, such as the [[μ-law algorithm]]. Small signals are digitized with finer granularity than are large ones; the effect is to add noise that is proportional to the signal strength. Sun's [[Au file format]] for sound is a popular example of mu-law encoding. Using 8-bit mu-law encoding would cut the per-channel bitrate of CD audio down to about 350 kbit/s, half the standard rate. Because this simple method only minimally exploits masking effects, it produces results that are often audibly inferior compared to the original.

	==Basic principles==		==Basic principles==

Kvng: mv {{FOLDOC}} attribution to talk

2021-11-30T21:27:19Z

mv {{FOLDOC}} attribution to talk

← Previous revision		Revision as of 21:27, 30 November 2021
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	Sub-band coding is used in the [[G.722]] codec which uses sub-band adaptive differential pulse code modulation (SB-[[ADPCM]]) within a bit rate of 64 kbit/s. In the SB-ADPCM technique, the frequency band is split into two sub-bands (higher and lower) and the signals in each sub-band are encoded using ADPCM.		Sub-band coding is used in the [[G.722]] codec which uses sub-band adaptive differential pulse code modulation (SB-[[ADPCM]]) within a bit rate of 64 kbit/s. In the SB-ADPCM technique, the frequency band is split into two sub-bands (higher and lower) and the signals in each sub-band are encoded using ADPCM.

	==References==
	{{FOLDOC}}

	==External links==		==External links==

Kvng: Reverted good faith edits by Mmncit (talk): Changes inconsistent with title. we can change the title but probably should discuss that on the talk page first. (TW)

2019-07-27T13:53:57Z

Reverted good faith edits by Mmncit (talk): Changes inconsistent with title. we can change the title but probably should discuss that on the talk page first. (TW)

← Previous revision		Revision as of 13:53, 27 July 2019
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	{{no footnotes\|date = October 2011}}		{{no footnotes\|date = October 2011}}

	[[File:SubBandCoding.svg\|thumb\|500px\|~~Subband~~ coding and decoding signal flow diagram]]		[[File:SubBandCoding.svg\|thumb\|500px\|Sub-band coding and decoding signal flow diagram]]

	In [[signal processing]], '''~~subband~~ coding''' ('''SBC''') is any form of [[transform coding]] that breaks a signal into a number of different [[frequency band]]s, typically by using a [[fast Fourier transform]], and encodes each one independently. This decomposition is often the first step in data compression for audio and video signals.		In [[signal processing]], '''sub-band coding''' ('''SBC''') is any form of [[transform coding]] that breaks a signal into a number of different [[frequency band]]s, typically by using a [[fast Fourier transform]], and encodes each one independently. This decomposition is often the first step in data compression for audio and video signals.

	SBC is the core technique used in many popular [[lossy audio compression]] algorithms including [[MP3]].		SBC is the core technique used in many popular [[lossy audio compression]] algorithms including [[MP3]].
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	First, a digital filter bank divides the input signal spectrum into some number (e.g., 32) of subbands. The psychoacoustic model looks at the energy in each of these subbands, as well as in the original signal, and computes masking thresholds using psychoacoustic information. Each of the subband samples is quantized and encoded so as to keep the quantization noise below the dynamically computed masking threshold. The final step is to format all these quantized samples into groups of data called frames, to facilitate eventual playback by a decoder.		First, a digital filter bank divides the input signal spectrum into some number (e.g., 32) of subbands. The psychoacoustic model looks at the energy in each of these subbands, as well as in the original signal, and computes masking thresholds using psychoacoustic information. Each of the subband samples is quantized and encoded so as to keep the quantization noise below the dynamically computed masking threshold. The final step is to format all these quantized samples into groups of data called frames, to facilitate eventual playback by a decoder.

	Decoding is much easier than encoding since no psychoacoustic model is involved. The frames are unpacked, subband samples are decoded, and a frequency-time mapping reconstructs an output audio signal.		Decoding is much easier than encoding, since no psychoacoustic model is involved. The frames are unpacked, subband samples are decoded, and a frequency-time mapping reconstructs an output audio signal.

	==Applications==		==Applications==
	Beginning in the late 1980s, a standardization body, the [[Moving Picture Experts Group]] (MPEG), developed standards for coding of both audio and video. Subband coding resides at the heart of the popular MP3 format (more properly known as [[MPEG-1 Audio Layer III]]), for example.		Beginning in the late 1980s, a standardization body, the [[Moving Picture Experts Group]] (MPEG), developed standards for coding of both audio and video. Subband coding resides at the heart of the popular MP3 format (more properly known as [[MPEG-1 Audio Layer III]]), for example.

	~~Subband~~ coding is used in the [[G.722]] codec which uses ~~subband~~ adaptive differential pulse code modulation (SB-[[ADPCM]]) within a bit rate of 64 kbit/s. In the SB-ADPCM technique, the frequency band is split into two ~~subbands~~ (higher and lower) and the signals in each ~~subband~~ are encoded using ADPCM.		Sub-band coding is used in the [[G.722]] codec which uses sub-band adaptive differential pulse code modulation (SB-[[ADPCM]]) within a bit rate of 64 kbit/s. In the SB-ADPCM technique, the frequency band is split into two sub-bands (higher and lower) and the signals in each sub-band are encoded using ADPCM.

	==References==		==References==
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	==External links==		==External links==
	* [https://web.archive.org/web/20070613152917/http://www.otolith.com/otolith/olt/sbc.html ~~Subband~~ Coding Tutorial]		* [https://web.archive.org/web/20070613152917/http://www.otolith.com/otolith/olt/sbc.html Sub-Band Coding Tutorial]

	{{Compression Methods}}		{{Compression Methods}}

Mmncit: update sub-band to subband

2019-07-24T22:27:02Z

update sub-band to subband

← Previous revision		Revision as of 22:27, 24 July 2019
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	{{no footnotes\|date = October 2011}}		{{no footnotes\|date = October 2011}}

	[[File:SubBandCoding.svg\|thumb\|500px\|~~Sub-band~~ coding and decoding signal flow diagram]]		[[File:SubBandCoding.svg\|thumb\|500px\|Subband coding and decoding signal flow diagram]]

	In [[signal processing]], '''subband coding''' ('''SBC''') is any form of [[transform coding]] that breaks a signal into a number of different [[frequency band]]s, typically by using a [[fast Fourier transform]], and encodes each one independently. This decomposition is often the first step in data compression for audio and video signals.		In [[signal processing]], '''subband coding''' ('''SBC''') is any form of [[transform coding]] that breaks a signal into a number of different [[frequency band]]s, typically by using a [[fast Fourier transform]], and encodes each one independently. This decomposition is often the first step in data compression for audio and video signals.
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	First, a digital filter bank divides the input signal spectrum into some number (e.g., 32) of subbands. The psychoacoustic model looks at the energy in each of these subbands, as well as in the original signal, and computes masking thresholds using psychoacoustic information. Each of the subband samples is quantized and encoded so as to keep the quantization noise below the dynamically computed masking threshold. The final step is to format all these quantized samples into groups of data called frames, to facilitate eventual playback by a decoder.		First, a digital filter bank divides the input signal spectrum into some number (e.g., 32) of subbands. The psychoacoustic model looks at the energy in each of these subbands, as well as in the original signal, and computes masking thresholds using psychoacoustic information. Each of the subband samples is quantized and encoded so as to keep the quantization noise below the dynamically computed masking threshold. The final step is to format all these quantized samples into groups of data called frames, to facilitate eventual playback by a decoder.

	Decoding is much easier than encoding, since no psychoacoustic model is involved. The frames are unpacked, subband samples are decoded, and a frequency-time mapping reconstructs an output audio signal.		Decoding is much easier than encoding since no psychoacoustic model is involved. The frames are unpacked, subband samples are decoded, and a frequency-time mapping reconstructs an output audio signal.

	==Applications==		==Applications==
	Beginning in the late 1980s, a standardization body, the [[Moving Picture Experts Group]] (MPEG), developed standards for coding of both audio and video. Subband coding resides at the heart of the popular MP3 format (more properly known as [[MPEG-1 Audio Layer III]]), for example.		Beginning in the late 1980s, a standardization body, the [[Moving Picture Experts Group]] (MPEG), developed standards for coding of both audio and video. Subband coding resides at the heart of the popular MP3 format (more properly known as [[MPEG-1 Audio Layer III]]), for example.

	~~Sub-band~~ coding is used in the [[G.722]] codec which uses ~~sub-band~~ adaptive differential pulse code modulation (SB-[[ADPCM]]) within a bit rate of 64 kbit/s. In the SB-ADPCM technique, the frequency band is split into two ~~sub-bands~~ (higher and lower) and the signals in each ~~sub-band~~ are encoded using ADPCM.		Subband coding is used in the [[G.722]] codec which uses subband adaptive differential pulse code modulation (SB-[[ADPCM]]) within a bit rate of 64 kbit/s. In the SB-ADPCM technique, the frequency band is split into two subbands (higher and lower) and the signals in each subband are encoded using ADPCM.

	==References==		==References==
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	==External links==		==External links==
	* [https://web.archive.org/web/20070613152917/http://www.otolith.com/otolith/olt/sbc.html ~~Sub-Band~~ Coding Tutorial]		* [https://web.archive.org/web/20070613152917/http://www.otolith.com/otolith/olt/sbc.html Subband Coding Tutorial]

	{{Compression Methods}}		{{Compression Methods}}

Mmncit: Undid revision 907733770 by Mmncit (talk)

2019-07-24T22:25:10Z

Undid revision 907733770 by Mmncit (talk)

Show changes

Mmncit at 22:24, 24 July 2019

2019-07-24T22:24:30Z

Show changes

Mmncit at 22:22, 24 July 2019

2019-07-24T22:22:48Z

← Previous revision		Revision as of 22:22, 24 July 2019
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	[[File:SubBandCoding.svg\|thumb\|500px\|Sub-band coding and decoding signal flow diagram]]		[[File:SubBandCoding.svg\|thumb\|500px\|Sub-band coding and decoding signal flow diagram]]

	In [[signal processing]], '''~~sub-band~~ coding''' ('''SBC''') is any form of [[transform coding]] that breaks a signal into a number of different [[frequency band]]s, typically by using a [[fast Fourier transform]], and encodes each one independently. This decomposition is often the first step in data compression for audio and video signals.		In [[signal processing]], '''subband coding''' ('''SBC''') is any form of [[transform coding]] that breaks a signal into a number of different [[frequency band]]s, typically by using a [[fast Fourier transform]], and encodes each one independently. This decomposition is often the first step in data compression for audio and video signals.

	SBC is the core technique used in many popular [[lossy audio compression]] algorithms including [[MP3]].		SBC is the core technique used in many popular [[lossy audio compression]] algorithms including [[MP3]].

Kvng: /* Applications */ avoid unnec redirect

2019-03-08T00:32:01Z

Applications: avoid unnec redirect

← Previous revision		Revision as of 00:32, 8 March 2019
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	==Applications==		==Applications==
	Beginning in the late 1980s, a standardization body, the [[~~Motion~~ Picture Experts Group]] (MPEG), developed standards for coding of both audio and video. Subband coding resides at the heart of the popular MP3 format (more properly known as [[MPEG-1 Audio Layer III]]), for example.		Beginning in the late 1980s, a standardization body, the [[Moving Picture Experts Group]] (MPEG), developed standards for coding of both audio and video. Subband coding resides at the heart of the popular MP3 format (more properly known as [[MPEG-1 Audio Layer III]]), for example.

	Sub-band coding is used in the [[G.722]] codec which uses sub-band adaptive differential pulse code modulation (SB-[[ADPCM]]) within a bit rate of 64 kbit/s. In the SB-ADPCM technique, the frequency band is split into two sub-bands (higher and lower) and the signals in each sub-band are encoded using ADPCM.		Sub-band coding is used in the [[G.722]] codec which uses sub-band adaptive differential pulse code modulation (SB-[[ADPCM]]) within a bit rate of 64 kbit/s. In the SB-ADPCM technique, the frequency band is split into two sub-bands (higher and lower) and the signals in each sub-band are encoded using ADPCM.