EP1238481B1 - Verfahren zur dekodierung von digitalen audiodaten - Google Patents

Verfahren zur dekodierung von digitalen audiodaten Download PDF

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Publication number
EP1238481B1
EP1238481B1 EP00981165A EP00981165A EP1238481B1 EP 1238481 B1 EP1238481 B1 EP 1238481B1 EP 00981165 A EP00981165 A EP 00981165A EP 00981165 A EP00981165 A EP 00981165A EP 1238481 B1 EP1238481 B1 EP 1238481B1
Authority
EP
European Patent Office
Prior art keywords
audio data
digital audio
frame
error
reference values
Prior art date
Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
Expired - Lifetime
Application number
EP00981165A
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German (de)
English (en)
French (fr)
Other versions
EP1238481A2 (de
Inventor
Claus Kupferschmidt
Torsten Mlasko
Marc Klein Middelink
Current Assignee (The listed assignees may be inaccurate. Google has not performed a legal analysis and makes no representation or warranty as to the accuracy of the list.)
Robert Bosch GmbH
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Robert Bosch GmbH
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Publication date
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Publication of EP1238481A2 publication Critical patent/EP1238481A2/de
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    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/005Correction of errors induced by the transmission channel, if related to the coding algorithm
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04HBROADCAST COMMUNICATION
    • H04H2201/00Aspects of broadcast communication
    • H04H2201/10Aspects of broadcast communication characterised by the type of broadcast system
    • H04H2201/20Aspects of broadcast communication characterised by the type of broadcast system digital audio broadcasting [DAB]

Definitions

  • the invention is based on a method for decoding digital audio data according to the preamble of the independent claim.
  • DAB Digital Audio Broadcasting
  • the entire frequency spectrum of the digital audio signals to be transmitted is divided into frequency ranges. These frequency ranges are called subbands in English.
  • a maximum of three scale factors are defined as reference values.
  • 36 samples are generated one after the other per channel in stereo transmissions.
  • the 36 samples are divided into time separated groups of 12 samples each.
  • a maximum of one scale factor is defined per group. If two or all three scale factors of a frequency range are the same or at least have very similar values, then only one scale factor is transmitted for these scale factors.
  • the receiver then performs error detection and correction procedures on the source decode after such procedures have been performed on a previous channel decode.
  • error detection and correction methods during source decoding affect both the DAB frame and the scale factors.
  • the digital audio data is denormalized by the scale factors, and decoding of the audio data takes place.
  • US 5,706,396 A describes a comparison of the transition of scale factors in decoding.
  • a Schellwertu a comparison with an expected transition or a polarity comparison is performed from DE 44 09 960 A is the error detection in digital audio data, in particular at scale factors by means of a checksum, so CRC to determine and depending on these error detection
  • CRC an error detection by CRC is described.
  • the inventive method results from the combination of features of claim 1 and has the advantage that by means of a plausibility check an error is detected, then to initiate error correction or concealment procedures.
  • the method is simple and uses the property of audio data that there are no large jumps over time. Therefore, advantageously, comparison of temporally successive reference values depending on the audio data results in a meaningful result as to whether an error exists or not.
  • the method according to the invention is advantageously simple and can be implemented in any audio decoder.
  • the method according to the invention is applicable to further audio decoding methods (standards). These standards include MPEG-1, MPEG-2 and MPEG-4.
  • the standards may or may not have their own error description.
  • a close correlation between the reference values which are scale factors in DAB, is used to determine whether an error exists.
  • Audio data entails that temporally adjacent data are closely correlated with each other. This is a property of language and music.
  • the feature is determined by means of a difference or averaging, whereby a meaningful, manageable and simple decision is made whether an error exists or not.
  • the method according to the invention is thus independent of a signal type, since the calculation method which is optimal for a respective signal can be used.
  • the signaling of the decision as to whether an error exists is made by means of a bit sequence, preferably a flag, whereby a simple evaluation of this decision is possible.
  • substitute values when no data is transmitted in a frequency range, so-called substitute values, known as default in English, are entered as reference values and then these substitute values are identified as such, so that the error detection according to the invention is not performed here otherwise an error would be erroneously assumed.
  • suitable substitute values can be determined so that error detection can be performed for all frequency values.
  • such substitute values are advantageously determined which lead to a feature which does not indicate an error, that is to say an adaptive determination of the substitute values. This simplifies the procedure, since the special case of the substitute value does not have to be intercepted.
  • FIG. 1 shows an MPEG1 Layer II frame
  • Figure 2 is a block diagram of the method according to the invention.
  • DAB Digital Video Broadcasting
  • DRM Digital Radio Mondial
  • a feature is generated which is suitable for additional error protection in the source decoding, in order to determine in a further stage whether an error exists.
  • the method according to the invention therefore uses the already existing methods. This concerns the error detection and correction of reference values in the source decoding. If errors are present here, the reference values identified as defective are replaced by previous reference values which have been stored. The reference values are thus monitored for errors by two methods.
  • the method according to the invention can also act as sole error detection in the decoding of the digital audio data because it is independent of other error detection methods and of the frame structure.
  • FIG. 1 shows an MPEG-1 Layer II frame.
  • the MPEG-1 Layer II frame begins with a frame header 1 followed by a field 2 for frame error detection.
  • a checksum referred to in English as Cyclic Redundancy Check used. If a faulty frame has been detected from the checksum, then an appropriate frame will replace the faulty frame, for example the previous frame may be used for this, or the faulty frame is muted. Alternatively, a prediction can also be made. In this case, an incorrectly received and corrected frame is calculated from correctly received or corrected frames.
  • the checksum is designed so that it can not detect all possible errors due to transmission efficiency reasons. In such a case, the checksum fails. With a checksum, however, several overlapping errors can mutually correct one another, so that in such a case, erroneously no error is detected by means of the checksum.
  • Characteristic of the checksum is the test of a sum of bits, with a consideration of the content of the audio data, as is the case in the inventive method, is omitted.
  • the audio signals are quantized.
  • a non-linear quantization is carried out, based on a psychoacoustic quantization curve.
  • Noise near the frequency that is related to the frequency of the sound spectrum is no longer perceived by the ear. This is called the listening threshold.
  • the different frequency ranges are also finely and finely quantized, the fineness of the quantization being determined by the fact that the quantization noise is still below the listening threshold. From this different quantization per frequency range results that different numbers of bits per frequency range are assigned. For example, the bit allocation per frequency range varies between 3 and 16 bits.
  • time-sequential reference values for a frequency range have the same or at least very similar size, since the power approximately matches. Therefore, it is not necessary to transmit a plurality of reference values for the frequency range when a reference value represents a plurality of time-separated groups of samples.
  • This field 4 now describes which reference values are to be used for which groups of samples for denormalization.
  • field 5 the reference values are then stored themselves.
  • field 6 the actual audio data, which are denormalized with the reference values, are stored.
  • Field 7 contains additional data that includes program-accompanying information and above all the checksum for the reference values of the following frame.
  • FIG. 2 shows a block diagram of the method according to the invention.
  • At an input 8 are the audio data.
  • an error detection of the reference values of the past frame is performed.
  • a feature is extracted from the current frame in which the reference values of the past frame and the current frame are subtracted from each other. If the sum is above a predetermined threshold value, then the difference is so great that there is no correlation between the two reference values, which actually can not occur with audio data. Therefore, this case is detected as an error.
  • averaging can be used instead of a mere subtraction, for example to calculate a standard deviation. If the standard deviation is above a predetermined threshold, this is recognized as an error.
  • a decider which compares the difference of the successive reference values with the predetermined threshold and makes a corresponding output, i. if there is an error, if a bit is set to 1, if there is no error, this bit remains at 0. This bit is also referred to as flag.
  • the error detection from the block 9 for the reference values and the error detection is linked together by the feature analysis from the block 11, the method being such that the block 11 uses the result of the previous frame, therefore in block 9 also Error detection performed for the reference value of the past frame.
  • the link 12 is designed such that the decision as to whether a fault exists is determined by means of a logical or link, i. Errors are signaled by a 1, no error by a 0, so that both, the error detection by checksum and the feature analysis, are not allowed to display an error if no error is to be detected.
  • error correction or fogging methods include frame repeats and a prediction.
  • a replacement value a default
  • the difference between a default and another reference value can lead to an indication of an error.
  • This replacement value must be characteristic, whereby it usually does not occur in the audio data, so that in this case the difference formation is omitted and here alone the error detection for the reference values is performed by checksum. That is, the flag for the error detection of the reference values remains here at 0.
  • the substitute value can also be designed so that the feature formed with the substitute value is always below the threshold value for error detection. This adjusts the substitute value to the reference values. In principle, then simply the corresponding reference value can be taken, so that a difference image will result in zero.
  • the decision is signaled whether an error exists or not. If there is an error, stored reference values from a past frame that has been transmitted correctly are taken instead of the erroneous reference value, if there is no error, all reference values from this frame are used.

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  • Engineering & Computer Science (AREA)
  • Computational Linguistics (AREA)
  • Signal Processing (AREA)
  • Health & Medical Sciences (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • Human Computer Interaction (AREA)
  • Physics & Mathematics (AREA)
  • Acoustics & Sound (AREA)
  • Multimedia (AREA)
  • Detection And Prevention Of Errors In Transmission (AREA)
  • Circuits Of Receivers In General (AREA)
  • Transmission Systems Not Characterized By The Medium Used For Transmission (AREA)
  • Compression, Expansion, Code Conversion, And Decoders (AREA)
EP00981165A 1999-12-08 2000-11-07 Verfahren zur dekodierung von digitalen audiodaten Expired - Lifetime EP1238481B1 (de)

Applications Claiming Priority (3)

Application Number Priority Date Filing Date Title
DE19959038 1999-12-08
DE19959038A DE19959038A1 (de) 1999-12-08 1999-12-08 Verfahren zur Dekodierung von digitalen Audiodaten
PCT/DE2000/003896 WO2001043320A2 (de) 1999-12-08 2000-11-07 Verfahren zur dekodierung von digitalen audiodaten

Publications (2)

Publication Number Publication Date
EP1238481A2 EP1238481A2 (de) 2002-09-11
EP1238481B1 true EP1238481B1 (de) 2007-04-11

Family

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EP00981165A Expired - Lifetime EP1238481B1 (de) 1999-12-08 2000-11-07 Verfahren zur dekodierung von digitalen audiodaten

Country Status (5)

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US (1) US7080006B1 (ja)
EP (1) EP1238481B1 (ja)
JP (1) JP2004500599A (ja)
DE (2) DE19959038A1 (ja)
WO (1) WO2001043320A2 (ja)

Families Citing this family (7)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
JP2004522198A (ja) * 2001-05-08 2004-07-22 コーニンクレッカ フィリップス エレクトロニクス エヌ ヴィ 音声符号化方法
US7428684B2 (en) 2002-04-29 2008-09-23 Fraunhofer-Gesellschaft Zur Forderung Der Angewandten Forschung E.V. Device and method for concealing an error
DE10219133B4 (de) * 2002-04-29 2007-02-22 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. Vorrichtung und Verfahren zum Verschleiern eines Fehlers
US7580476B2 (en) * 2003-06-26 2009-08-25 Northrop Grumman Corporation Communication system and method for improving efficiency and linearity
JP4539180B2 (ja) * 2004-06-07 2010-09-08 ソニー株式会社 音響復号装置及び音響復号方法
JP4698688B2 (ja) * 2007-02-27 2011-06-08 シャープ株式会社 送受信方法、送受信装置及びプログラム
CN100524462C (zh) * 2007-09-15 2009-08-05 华为技术有限公司 对高带信号进行帧错误隐藏的方法及装置

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US4831624A (en) * 1987-06-04 1989-05-16 Motorola, Inc. Error detection method for sub-band coding
US5091945A (en) * 1989-09-28 1992-02-25 At&T Bell Laboratories Source dependent channel coding with error protection
DE4202140A1 (de) 1992-01-27 1993-07-29 Thomson Brandt Gmbh Verfahren zur uebertragung digitaler audio-signale
US5617333A (en) * 1993-11-29 1997-04-01 Kokusai Electric Co., Ltd. Method and apparatus for transmission of image data
DE4409960A1 (de) * 1994-03-23 1995-09-28 Inst Rundfunktechnik Gmbh Verfahren zur Verminderung der subjektiven Störempfindung bei störungsbehaftetem Empfang bei Verwendung von digital übertragenen Tonsignalen
JP3046213B2 (ja) * 1995-02-02 2000-05-29 三菱電機株式会社 サブバンド・オーディオ信号合成装置
JP2757818B2 (ja) * 1995-04-20 1998-05-25 日本電気株式会社 補助データ処理回路
EP0976216B1 (de) * 1997-02-27 2002-11-27 Siemens Aktiengesellschaft Verfahren und anordnung zur rahmenfehlerdetektion zwecks fehlerverdeckung insbesondere bei gsm übertragungen
DE19735675C2 (de) * 1997-04-23 2002-12-12 Fraunhofer Ges Forschung Verfahren zum Verschleiern von Fehlern in einem Audiodatenstrom
US6208959B1 (en) * 1997-12-15 2001-03-27 Telefonaktibolaget Lm Ericsson (Publ) Mapping of digital data symbols onto one or more formant frequencies for transmission over a coded voice channel
DE19921122C1 (de) * 1999-05-07 2001-01-25 Fraunhofer Ges Forschung Verfahren und Vorrichtung zum Verschleiern eines Fehlers in einem codierten Audiosignal und Verfahren und Vorrichtung zum Decodieren eines codierten Audiosignals
US6208699B1 (en) * 1999-09-01 2001-03-27 Qualcomm Incorporated Method and apparatus for detecting zero rate frames in a communications system
US6728323B1 (en) * 2000-07-10 2004-04-27 Ericsson Inc. Baseband processors, mobile terminals, base stations and methods and systems for decoding a punctured coded received signal using estimates of punctured bits

Also Published As

Publication number Publication date
US7080006B1 (en) 2006-07-18
WO2001043320A2 (de) 2001-06-14
WO2001043320A3 (de) 2002-02-14
JP2004500599A (ja) 2004-01-08
DE50014248D1 (de) 2007-05-24
EP1238481A2 (de) 2002-09-11
DE19959038A1 (de) 2001-06-28

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