EP2380169A1 - Procédé et moyen d'amélioration modulaire de la qualité d'un traitement de codage du signal - Google Patents

Procédé et moyen d'amélioration modulaire de la qualité d'un traitement de codage du signal

Info

Publication number
EP2380169A1
EP2380169A1 EP09807441A EP09807441A EP2380169A1 EP 2380169 A1 EP2380169 A1 EP 2380169A1 EP 09807441 A EP09807441 A EP 09807441A EP 09807441 A EP09807441 A EP 09807441A EP 2380169 A1 EP2380169 A1 EP 2380169A1
Authority
EP
European Patent Office
Prior art keywords
signal
reference signals
error signal
pulses
quality
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.)
Granted
Application number
EP09807441A
Other languages
German (de)
English (en)
Other versions
EP2380169B1 (fr
Inventor
Stefan Schandl
Panji Setiawan
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.)
Unify GmbH and Co KG
Original Assignee
Siemens Enterprise Communications GmbH and Co KG
Priority date (The priority date 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 date listed.)
Filing date
Publication date
Application filed by Siemens Enterprise Communications GmbH and Co KG filed Critical Siemens Enterprise Communications GmbH and Co KG
Publication of EP2380169A1 publication Critical patent/EP2380169A1/fr
Application granted granted Critical
Publication of EP2380169B1 publication Critical patent/EP2380169B1/fr
Active legal-status Critical Current
Anticipated expiration legal-status Critical

Links

Classifications

    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech 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
    • G10L19/04—Speech 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 using predictive techniques
    • G10L19/16—Vocoder architecture
    • G10L19/18—Vocoders using multiple modes
    • G10L19/24—Variable rate codecs, e.g. for generating different qualities using a scalable representation such as hierarchical encoding or layered encoding

Definitions

  • the invention relates to a method for the scalable improvement of the quality of a signal coding method
  • the audio signals to be transmitted are compressed by means of coding methods and decompressed after the transmission.
  • Such a coding method which is intended for transmission of a speech signal in a frequency range of 300 to 3400 Hz with a data rate of 8 kbit / s, is known, for example, from ITU-T Recommendation G.729.
  • the transmission of an extended frequency range from 50 Hz to
  • a so-called broadband voice codec provided for this purpose is, for example, in ITU-T Recommendation G.722. EV described.
  • the method uses subband adaptive differential pulse code modulation (SB-ADPCM) to encode audio signals.
  • SB-ADPCM subband adaptive differential pulse code modulation
  • the invention has for its object to provide a method for scalable improvement of the quality of a coding method according to the Subband Adaptive Differential Pulse Code principle.
  • the information about the reference signal with the minimum error signal is transmitted.
  • FIG. 2 shows the structure of a codec according to the invention
  • FIG. 3 shows the structure of a decoder according to the invention.
  • the reference signal according to FIG. 1 comprises a number of N discrete pulses ⁇ ⁇ n).
  • the Dirac pulses may have a predetermined number of amplitude values L.
  • FIG. 2 shows the structural design of an encoder according to the invention which, in addition to a conventional coder ADPCM according to the Subband Adaptive Differential Pulse Code principle according to IT-U Recommendation G.722, comprises means for generating reference signals which each have one for each stage of the extension Signal generator EHDSl, ... EHDSS for generating the reference signals c (n) and a control unit CB 1, ... CB S has.
  • the reference signals c (n) are compared with a digital error signal e H over a predetermined time interval, a so-called frame, which in a conventional coding method corresponds to IT-U recommen- tion G.722 was determined from an input signal to be coded and a prognosis signal.
  • the reference signal c (n) with the lowest error amount E n is now determined and the information about this signal is transmitted as additional information I ⁇ imin f • ⁇ iHSmin and used in the receiver for decoding the useful signal ,
  • the starting point is a sampling rate of 8 KHz and thus a duration of a sampling interval of 125 ⁇ sec.
  • the duration of a comparison section is 5 msec, the number of possible amplitude values L for the Dirac pulses is 2.
  • Fig. 3 shows the structure of a decoder according to the invention in the received signal from the I H, Im, IH 2 - the audio signal is obtained I HS - -.
  • the received signal includes the recovered with the invention, additional information Inimm / • • • I HS ⁇ H ⁇ depending on the number of selected channels in the expansion stages.

Landscapes

  • Engineering & Computer Science (AREA)
  • Quality & Reliability (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)
  • Compression, Expansion, Code Conversion, And Decoders (AREA)

Abstract

La présente invention concerne un procédé permettant l'amélioration modulaire de la qualité d'un traitement de codage du signal selon la recommandation UIT G.722. Ce procédé comporte les étapes suivantes: - au cours d'un traitement itératif comportant un nombre d'étapes répétitives fonction de la quantité d'extensions, on prend un signal d'erreur (E) déterminé sur la base, d'une part d'un signal d'entrée à coder, et d'autre part d'un signal de pronostic, et on le compare à une quantité M*LN de signaux de référence différents, puis on utilise le résultat de cette comparaison pour déterminer le signal de référence correspondant au minimum du signal d'erreur sur la base d'un critère d'erreur prédéfini; - les signaux de référence sont tous constitués d'impulsions Dirac équidistantes δ(n) décrites par la formule (I) dans laquelle "off = [0.. M-1]" est l'intervalle entre la première impulsion et l'instant zéro, "αP ∈ {α0,α1,..,αL-1}" est la valeur d'amplitude, "M" est l'intervalle entre chaque impulsion, "N" est le nombre d'impulsions, et "L" est le nombre de niveaux de mesure différents; - et enfin, on transmet l'information concernant le signal de référence correspondant au minimum du signal d'erreur.
EP09807441.2A 2008-12-19 2009-12-10 Procédé et moyen d'amélioration modulaire de la qualité d'un traitement de codage du signal Active EP2380169B1 (fr)

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
ATA1982/2008A AT509439B1 (de) 2008-12-19 2008-12-19 Verfahren und mittel zur skalierbaren verbesserung der qualität eines signalcodierverfahrens
PCT/EP2009/008853 WO2010069513A1 (fr) 2008-12-19 2009-12-10 Procédé et moyen d'amélioration modulaire de la qualité d'un traitement de codage du signal

Publications (2)

Publication Number Publication Date
EP2380169A1 true EP2380169A1 (fr) 2011-10-26
EP2380169B1 EP2380169B1 (fr) 2015-12-09

Family

ID=41812891

Family Applications (1)

Application Number Title Priority Date Filing Date
EP09807441.2A Active EP2380169B1 (fr) 2008-12-19 2009-12-10 Procédé et moyen d'amélioration modulaire de la qualité d'un traitement de codage du signal

Country Status (6)

Country Link
US (1) US8774312B2 (fr)
EP (1) EP2380169B1 (fr)
CN (1) CN102257565B (fr)
AT (1) AT509439B1 (fr)
BR (1) BRPI0922993A2 (fr)
WO (1) WO2010069513A1 (fr)

Family Cites Families (5)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
FR2481026B1 (fr) * 1980-04-21 1984-06-15 France Etat
JP2598159B2 (ja) * 1990-08-28 1997-04-09 三菱電機株式会社 音声信号処理装置
US5956674A (en) * 1995-12-01 1999-09-21 Digital Theater Systems, Inc. Multi-channel predictive subband audio coder using psychoacoustic adaptive bit allocation in frequency, time and over the multiple channels
KR100711989B1 (ko) * 2002-03-12 2007-05-02 노키아 코포레이션 효율적으로 개선된 스케일러블 오디오 부호화
KR100467326B1 (ko) 2002-12-09 2005-01-24 학교법인연세대학교 추가 비트 할당 기법을 이용한 음성 부호화 및 복호화를위한 송수신기

Non-Patent Citations (1)

* Cited by examiner, † Cited by third party
Title
See references of WO2010069513A1 *

Also Published As

Publication number Publication date
WO2010069513A1 (fr) 2010-06-24
EP2380169B1 (fr) 2015-12-09
AT509439B1 (de) 2013-05-15
BRPI0922993A2 (pt) 2016-01-26
AT509439A1 (de) 2011-08-15
US20120014474A1 (en) 2012-01-19
CN102257565A (zh) 2011-11-23
US8774312B2 (en) 2014-07-08
CN102257565B (zh) 2013-05-29

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