USRE44897E1 - Process of low sampling rate digital encoding of audio signals - Google Patents

Process of low sampling rate digital encoding of audio signals Download PDF

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Publication number
USRE44897E1
USRE44897E1 US13/897,221 US89722198A USRE44897E US RE44897 E1 USRE44897 E1 US RE44897E1 US 89722198 A US89722198 A US 89722198A US RE44897 E USRE44897 E US RE44897E
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Prior art keywords
scale factor
coding
frequency lines
scale
iec
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US13/897,221
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English (en)
Inventor
Oliver Kunz
Martin Dietz
Rainer Buchta
Jurgen Zeller
Karlheinz Brandenburg
Martin Sieler
Heinz Gerhauser
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Fraunhofer Gesellschaft zur Forderung der Angewandten Forschung eV
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Fraunhofer Gesellschaft zur Forderung der Angewandten Forschung eV
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    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04B—TRANSMISSION
    • H04B1/00—Details of transmission systems, not covered by a single one of groups H04B3/00 - H04B13/00; Details of transmission systems not characterised by the medium used for transmission
    • H04B1/66—Details of transmission systems, not covered by a single one of groups H04B3/00 - H04B13/00; Details of transmission systems not characterised by the medium used for transmission for reducing bandwidth of signals; for improving efficiency of transmission
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04B—TRANSMISSION
    • H04B1/00—Details of transmission systems, not covered by a single one of groups H04B3/00 - H04B13/00; Details of transmission systems not characterised by the medium used for transmission
    • H04B1/66—Details of transmission systems, not covered by a single one of groups H04B3/00 - H04B13/00; Details of transmission systems not characterised by the medium used for transmission for reducing bandwidth of signals; for improving efficiency of transmission
    • H04B1/665—Details of transmission systems, not covered by a single one of groups H04B3/00 - H04B13/00; Details of transmission systems not characterised by the medium used for transmission for reducing bandwidth of signals; for improving efficiency of transmission using psychoacoustic properties of the ear, e.g. masking effect

Definitions

  • the present invention refers to a method for coding an audio signal which has been digitized at a low sampling rate.
  • the invention refers to a coding method which is only slightly modified relative to the Standard ISO/IEC 13878-3 (MPEG2 layer 3) and which enables audio signals which are digitized at a lower sampling rate than the sampling rate according to the Standard ISO/IEC 13818-3 to be transmitted at a low bit rate.
  • the existing Standard ISO/IEC 13818-3 published in May 15, 1995 defines with layer 3 a coding method for signals with sampling frequencies between 24 kHz and 16 kHz and makes possible bit rates of down to 8 kbit/s. In particular at this very low bit rate, which is very attractive for a transmission in computer networks e.g., the use of still smaller sampling frequencies would be desirable.
  • the cited Standard ISO/IEC 13818-3 does not provide these, however.
  • the present invention provides a method for coding an audio signal digitized at a low sampling rate to obtain time domain audio samples, comprising the steps of producing a frequency domain representation of the time domain audio samples, the frequency domain representation including a total number of successive frequency lines; subdividing the total number of successive frequency lines into a plurality of scale factor bands each scale factor band having a number of successive frequency lines wherein a scale factor is assigned to each scale factor band, the assigned scale factor being used for coding the frequency lines in the respective scale factor band; forming a plurality of regions, each region including a plurality of successive scale factor bands wherein the scale factors assigned to the plurality of scale factor bands in a region are each coded with the same number of bits, which is determined according to the largest scale factor of the region, and wherein a region including the scale factor bands having frequency lines that correspond to the higher frequency range frequency lines among the frequency lines in all regions is the highest region; and setting to a value of zero the scale factors that are assigned to the scale factor bands in at least the highest region to obtain zero-valued scale
  • the present invention provides coding of audio signals which have been digitized at a sampling rate which is lower than the sampling rate according to the Standard ISO-MPEG2 layer 3.
  • the successive frequency lines of the digitized audio signal which are assigned to a scale factor band are coded with the same scale factor, this being transmitted together with the coded scale factor band (see table B.8 of ISO/IEC 13818-3).
  • the present invention is so conceived that at least the frequency lines of the highest region of scale factor bands are coded with the scale factor 0, so that for at least the highest region no scale factor is coded and transmitted.
  • the bits which are saved through the missing scale factor or scale factors are used for the finer quantization, compared to the Standard ISO/IEC 13818-3, of the frequency lines in the rest of the spectrum.
  • the grouping of the frequency lines into scale factor bands is modified relative to the cited Standard ISO/IEC 13818-3 in such a way that the scale factor bandwidths within the highest region are reduced relative to the scale factor bandwidths of the highest region according to the Standard ISO-MPEG2 layer 3.
  • FIG. 1 illustrates a general block diagram of the inventive method
  • FIG. 2a illustrates a digitized audio signal having audio samples
  • FIG. 2b illustrates a frequency domain representation of the audio samples in FIG. 2a , the frequency domain representation having 576 successive frequency lines as in ISO/IEC 13818-3 (MPEG2 layer 3);
  • FIG. 2c illustrates the subdivision of the frequency domain representation into 22 scale factor bands (sfb 0 to sfb 21 ) as in ISO/IEC 13818-3 (MPEG2 layer 3);
  • FIG. 2d illustrates the four regions (region 0 to 3) as in ISO/IEC 13818-3, region 3 being the highest region;
  • FIG. 2e illustrates the four regions as in FIG. 2d with zero valued scale factors in the highest region in accordance with the present invention.
  • FIG. 2f illustrates the coded frequency lines for the 0 th to 2 nd regions for which bits are required for coding the scale factors themselves, and for the 3 rd region for which no bits are required for the scale factors.
  • a frame header which is a modification of the standard MPEG frame header is used so as to signal the non-standard sampling rate and the non-standard coding for a decoder.
  • the hexadecimal sync word “FFF” hexadecimal for twelve ones
  • the sync word “FFE” eleven ones and a zero
  • a decoder recognizes a sync word which has been modified in this way, this is an indication that the bit stream contains a signal which, compared to the Standard MPEG2, has been digitized at preferably half the sampling rate (12 kHz, 11.025 kHz or 8 kHz) (10 in FIG. 1 , FIG. 2a ).
  • the structure of the bit stream is unchanged with respect to the Standard ISO/IEC 13818-3.
  • a frequency domain representation of the thus obtained audio samples is produced (12 in FIG. 1 , FIG. 2b ).
  • the grouping (14 in FIG. 1 , FIG. 2c ) of the frequency lines into scale factor bands is also modified.
  • a construction in the bit stream format of layer 3 is exploited here, with the aid of which the number of bits available for the coding of the spectrum can be increased.
  • the table of the scale factor bandwidths is changed for the sampling rate 8 kHz in such a way that the highest region in the spectrum contains only very few lines, which because of bandwidth restrictions are as a rule not used or only little used anyway.
  • the scale factors of the scale factor bands of the highest region are set to zero without exception (18 in FIG. 1 , FIG. 2e ), so that no coding of the scale factors is necessary (22 in FIG. 1 ).
  • the spectrum is thus divided up for this embodiment into 17 scale factor bands without however having to renounce the bit stream syntax of the Standard MPEG2.
  • the tables for the scale factor bandwidths are reproduced below.
  • the first table for the widths of the scale factor bands for 16 kHz sampling rate corresponds to the Standard ISO-MPEG2 layer 3.
  • the second table has been modified according to the teaching of the present invention for the 8 kHz sampling rate.
  • the last scale factor bandwidths in the method according to the present invention have been markedly reduced compared to the known method according to Standard.
  • the last 100 frequency lines are coded without scale factor (20 in FIG. 1 , FIG. 2f ).
  • the desired effect can thus be achieved provided the utilized bandwidth is smaller than 3300 Hz, which at a bit rate of 8 kbit/s, for which the 8 kHz sampling rate is intended, certainly makes sense.

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  • Engineering & Computer Science (AREA)
  • Computer Networks & Wireless Communication (AREA)
  • Signal Processing (AREA)
  • Compression, Expansion, Code Conversion, And Decoders (AREA)
  • Transmission Systems Not Characterized By The Medium Used For Transmission (AREA)
  • Reduction Or Emphasis Of Bandwidth Of Signals (AREA)
  • Circuits Of Receivers In General (AREA)
US13/897,221 1996-04-04 1997-02-19 Process of low sampling rate digital encoding of audio signals Expired - Lifetime USRE44897E1 (en)

Priority Applications (1)

Application Number Priority Date Filing Date Title
US13/897,221 USRE44897E1 (en) 1996-04-04 1997-02-19 Process of low sampling rate digital encoding of audio signals

Applications Claiming Priority (5)

Application Number Priority Date Filing Date Title
DE19613643 1996-04-04
DE19613643A DE19613643A1 (de) 1996-04-04 1996-04-04 Verfahren zum Codieren eines mit einer niedrigen Abtastrate digitalisierten Audiosignals
US7739597A 1997-02-19 1997-02-19
US13/897,221 USRE44897E1 (en) 1996-04-04 1997-02-19 Process of low sampling rate digital encoding of audio signals
PCT/EP1997/000792 WO1997038497A1 (de) 1996-04-04 1997-02-19 Verfahren zum codieren eines mit einer niedrigen abtastrate digitalisierten audiosignals

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USRE44897E1 true USRE44897E1 (en) 2014-05-13

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US13/897,221 Expired - Lifetime USRE44897E1 (en) 1996-04-04 1997-02-19 Process of low sampling rate digital encoding of audio signals
US09/077,395 Expired - Lifetime US6185539B1 (en) 1996-04-04 1997-02-19 Process of low sampling rate digital encoding of audio signals

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US09/077,395 Expired - Lifetime US6185539B1 (en) 1996-04-04 1997-02-19 Process of low sampling rate digital encoding of audio signals

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US (2) USRE44897E1 (de)
EP (1) EP0832521B3 (de)
JP (1) JP3103382B2 (de)
KR (1) KR100266054B1 (de)
AT (1) ATE174167T1 (de)
AU (1) AU703390B2 (de)
CA (1) CA2241453C (de)
DE (2) DE19613643A1 (de)
DK (1) DK0832521T3 (de)
ES (1) ES2125739T7 (de)
NO (1) NO315298B1 (de)
RU (1) RU2134016C1 (de)
UA (1) UA48195C2 (de)
WO (1) WO1997038497A1 (de)

Families Citing this family (6)

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JP3515903B2 (ja) * 1998-06-16 2004-04-05 松下電器産業株式会社 オーディオ符号化のための動的ビット割り当て方法及び装置
JP3813025B2 (ja) * 1998-10-29 2006-08-23 株式会社リコー デジタル音響信号符号化装置、デジタル音響信号符号化方法及びデジタル音響信号符号化プログラムを記録した媒体
US8290782B2 (en) * 2008-07-24 2012-10-16 Dts, Inc. Compression of audio scale-factors by two-dimensional transformation
DE102009028919A1 (de) 2009-08-27 2011-03-03 Robert Bosch Gmbh Verfahren und Steuergerät zur Direktabtastung einer Mehrzahl von Radiobändern
JP5316896B2 (ja) * 2010-03-17 2013-10-16 ソニー株式会社 符号化装置および符号化方法、復号装置および復号方法、並びにプログラム
US8552900B1 (en) * 2012-04-20 2013-10-08 Texas Instruments Incorporated System and method of clocking low sample rate analog to digital converters while minimizing linearity errors

Citations (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
EP0457391A1 (de) * 1990-05-14 1991-11-21 Koninklijke Philips Electronics N.V. Codierverfahren und Codieranordnung mit einem Unterbandcoder und Sender mit der Codieranordnung
EP0525774A2 (de) * 1991-07-31 1993-02-03 Matsushita Electric Industrial Co., Ltd. Verfahren und Einrichtung zur Kodierung eines digitalen Audiosignals
EP0612159A2 (de) * 1993-02-19 1994-08-24 Matsushita Electric Industrial Co., Ltd. Verfahren zur Verbesserung eines Grobquantisierers in einem ATRAC-Codierer

Family Cites Families (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
DE4236989C2 (de) * 1992-11-02 1994-11-17 Fraunhofer Ges Forschung Verfahren zur Übertragung und/oder Speicherung digitaler Signale mehrerer Kanäle

Patent Citations (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
EP0457391A1 (de) * 1990-05-14 1991-11-21 Koninklijke Philips Electronics N.V. Codierverfahren und Codieranordnung mit einem Unterbandcoder und Sender mit der Codieranordnung
EP0525774A2 (de) * 1991-07-31 1993-02-03 Matsushita Electric Industrial Co., Ltd. Verfahren und Einrichtung zur Kodierung eines digitalen Audiosignals
EP0612159A2 (de) * 1993-02-19 1994-08-24 Matsushita Electric Industrial Co., Ltd. Verfahren zur Verbesserung eines Grobquantisierers in einem ATRAC-Codierer

Non-Patent Citations (3)

* Cited by examiner, † Cited by third party
Title
ISO/IEL "Standard 11172-3" ISO pp. 73 and 741. *
ISO/IEL "STandard 13818-3 amendment" ISO pages all. *
ISO/IEL "Standard 13818-3" ISO pages all. *

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Publication number Publication date
ATE174167T1 (de) 1998-12-15
EP0832521B1 (de) 1998-12-02
ES2125739T7 (es) 2014-04-08
JPH11507198A (ja) 1999-06-22
NO982920D0 (no) 1998-06-23
KR100266054B1 (ko) 2000-09-15
ES2125739T3 (es) 1999-03-01
CA2241453C (en) 2002-01-29
DK0832521T3 (da) 1999-08-16
AU2091697A (en) 1997-10-29
NO982920L (no) 1998-07-23
UA48195C2 (uk) 2002-08-15
US6185539B1 (en) 2001-02-06
AU703390B2 (en) 1999-03-25
JP3103382B2 (ja) 2000-10-30
WO1997038497A1 (de) 1997-10-16
DE59700044D1 (de) 1999-01-14
DE19613643A1 (de) 1997-10-09
EP0832521A1 (de) 1998-04-01
EP0832521B3 (de) 2014-01-15
KR19990067625A (ko) 1999-08-25
NO315298B1 (no) 2003-08-11
RU2134016C1 (ru) 1999-07-27
CA2241453A1 (en) 1997-10-16

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