EP0697123B1 - Verfahren zur aufbereitung von daten, insbesondere von codierten sprachsignalparametern - Google Patents

Verfahren zur aufbereitung von daten, insbesondere von codierten sprachsignalparametern Download PDF

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Publication number
EP0697123B1
EP0697123B1 EP94912471A EP94912471A EP0697123B1 EP 0697123 B1 EP0697123 B1 EP 0697123B1 EP 94912471 A EP94912471 A EP 94912471A EP 94912471 A EP94912471 A EP 94912471A EP 0697123 B1 EP0697123 B1 EP 0697123B1
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EP
European Patent Office
Prior art keywords
signal parameters
bits
bit
quantisation
total number
Prior art date
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Expired - Lifetime
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EP94912471A
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German (de)
English (en)
French (fr)
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EP0697123A1 (de
Inventor
Jörg-Martin Müller
Bertram Wächter
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Robert Bosch GmbH
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Robert Bosch GmbH
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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
    • G10L19/002—Dynamic bit allocation
    • 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
    • G10L19/02—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 spectral analysis, e.g. transform vocoders or subband vocoders
    • G10L19/032—Quantisation or dequantisation of spectral components
    • G10L19/038—Vector quantisation, e.g. TwinVQ audio
    • 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
    • G10L19/08—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters
    • G10L19/12—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters the excitation function being a code excitation, e.g. in code excited linear prediction [CELP] vocoders

Definitions

  • the invention relates to a method for processing Data, especially of coded speech signal parameters for Transmission purposes.
  • the voice signal When coding and decoding voice signals, especially for cellular applications, the voice signal sampled and divided into sections (time sections). For each section, prediction values for different types of signal parameters are formed. Such Signal parameters are e.g. Short-term parameters for Characterization of the formant structure (resonances of the Speech tract) and long-term parameters for characterization the pitch structure (pitch) of the speech signal (ANT Telecommunications Reports, Issue 5, Nov. 1988, pages 93 to 105). With speech coding using "Analysis by Synthesis "are the model and excitation parameters quantized, coded and transmitted to the recipient. For the bit rate will be further reduced Vector quantization used (see above; DE / EP 0 266 620 T1; EP 504 627 A2; EP 294 020 A2).
  • the object of the present invention is a method initially mentioned in such a way that further Reduction of the bit rate a satisfactory reconstruction the output data is possible. This task is accomplished by the Steps of claim 1 solved.
  • the other claims show advantageous configurations.
  • the method according to the invention is particularly noteworthy due to its robustness against transmission errors.
  • the method according to the invention enables the construction of Speech codecs whose speech quality is better than that of Speech codecs with reduction in quantization levels Multiples of 2. Because transmission errors in general occur frequently, there is none with reduced effort Worsening of an error correction.
  • speech signals from a speech signal source Q are sampled by means of an A / D converter and analyzed in an analysis unit A with regard to similar speech signal parameters.
  • the analysis unit delivers a set of voice signal parameters of the same type, for example a set of short-term parameters KP for the formant structure (excitation parameters), a set of long-term parameters LP for the pitch structure and a set of filter weighting parameters FP.
  • These parameter sets are used to obtain prediction values in predictors PRK, PRL, PRF in a conventional manner, for example in accordance with EP 364 647, which are subjected to vector quantization VQ.
  • the quantized signal parameters are combined in a frame formation unit RA, for example in such a way that a frame of the frame duration of, for example, 20 msec. consists of 4 frame sections of 5 msec each. Similar signal parameters are accommodated in each of these frame sections. Bits are now suppressed from at least two of these frame sections (the treatment of two frame sections in each case is described below, of course more than two frame sections can also be treated together) by means of a bit suppression unit BÜ.
  • the bit suppression according to the invention is not carried out individually for each frame section, but for the total number of bits from at least two types of similar frame sections combined, ie, for example, for the total number of bits of the short-term and long-term parameters in a frame of 20 msec.
  • the number n of bits to be suppressed is advantageously based on the relationship distributed over the frame sections, where m is the number of similar signal parameters and g is the total number of original bits. The bit difference from the total number g of the unreduced bits to the nearest higher power of two is thus suppressed.
  • bits are preferably used for bit suppression selected the most statistically unlikely Correspond to quantization levels. This requirement can for example, that little probable levels of quantization in a memory SP previously saved by the bit suppression unit BÜ controls. Because the probability of Quantization levels is generally conditional, i.e. to a selected signal parameter from a frame section there are signal parameters in the next frame section Occurs following the selected signal parameter is more likely than others to occur proceeding with the selection of the bit suppression according to FIG. 2, i.e. there are all bits in the structure shown suppressed, whose fields are crossed.
  • FIG. 1 A structure of 12 ⁇ 12 vectors is shown in FIG.
  • the frame section S1 has a quantization with 4 bits for amplitude values of the same type, as does the frame section S2. There are 7 bits for the vector. Bit suppression now takes place according to the following relationships: for S1 ⁇ 7, 0 ⁇ S2 ⁇ 10 applies and for S1> 7, 0 ⁇ S2 ⁇ 9.

Landscapes

  • Engineering & Computer Science (AREA)
  • Physics & Mathematics (AREA)
  • Computational Linguistics (AREA)
  • Signal Processing (AREA)
  • Health & Medical Sciences (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • Human Computer Interaction (AREA)
  • Acoustics & Sound (AREA)
  • Multimedia (AREA)
  • Spectroscopy & Molecular Physics (AREA)
  • Compression, Expansion, Code Conversion, And Decoders (AREA)
  • Transmission Systems Not Characterized By The Medium Used For Transmission (AREA)
EP94912471A 1993-05-07 1994-04-20 Verfahren zur aufbereitung von daten, insbesondere von codierten sprachsignalparametern Expired - Lifetime EP0697123B1 (de)

Applications Claiming Priority (3)

Application Number Priority Date Filing Date Title
DE4315319 1993-05-07
DE4315319A DE4315319C2 (de) 1993-05-07 1993-05-07 Verfahren zur Aufbereitung von Daten, insbesondere von codierten Sprachsignalparametern
PCT/DE1994/000433 WO1994027284A1 (de) 1993-05-07 1994-04-20 Verfahren zur aufbereitung von daten, insbesondere von codierten sprachsignalparametern

Publications (2)

Publication Number Publication Date
EP0697123A1 EP0697123A1 (de) 1996-02-21
EP0697123B1 true EP0697123B1 (de) 1999-07-14

Family

ID=6487542

Family Applications (1)

Application Number Title Priority Date Filing Date
EP94912471A Expired - Lifetime EP0697123B1 (de) 1993-05-07 1994-04-20 Verfahren zur aufbereitung von daten, insbesondere von codierten sprachsignalparametern

Country Status (9)

Country Link
US (1) US5794183A (hu)
EP (1) EP0697123B1 (hu)
AU (1) AU679980B2 (hu)
DE (2) DE4315319C2 (hu)
DK (1) DK0697123T3 (hu)
ES (1) ES2136193T3 (hu)
FI (1) FI116598B (hu)
HU (1) HU215620B (hu)
WO (1) WO1994027284A1 (hu)

Families Citing this family (5)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US7729918B2 (en) * 2001-03-14 2010-06-01 At&T Intellectual Property Ii, Lp Trainable sentence planning system
US7046636B1 (en) 2001-11-26 2006-05-16 Cisco Technology, Inc. System and method for adaptively improving voice quality throughout a communication session
US20070286351A1 (en) * 2006-05-23 2007-12-13 Cisco Technology, Inc. Method and System for Adaptive Media Quality Monitoring
US8248953B2 (en) 2007-07-25 2012-08-21 Cisco Technology, Inc. Detecting and isolating domain specific faults
US7948910B2 (en) * 2008-03-06 2011-05-24 Cisco Technology, Inc. Monitoring quality of a packet flow in packet-based communication networks

Family Cites Families (10)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
DE266620C (hu) *
IT1195350B (it) * 1986-10-21 1988-10-12 Cselt Centro Studi Lab Telecom Procedimento e dispositivo per la codifica e decodifica del segnale vocale mediante estrazione di para metri e tecniche di quantizzazione vettoriale
US4969192A (en) * 1987-04-06 1990-11-06 Voicecraft, Inc. Vector adaptive predictive coder for speech and audio
US4817157A (en) * 1988-01-07 1989-03-28 Motorola, Inc. Digital speech coder having improved vector excitation source
DE3853161T2 (de) * 1988-10-19 1995-08-17 Ibm Vektorquantisierungscodierer.
JPH02272500A (ja) * 1989-04-13 1990-11-07 Fujitsu Ltd コード駆動音声符号化方式
DE69029120T2 (de) * 1989-04-25 1997-04-30 Toshiba Kawasaki Kk Stimmenkodierer
US5091945A (en) * 1989-09-28 1992-02-25 At&T Bell Laboratories Source dependent channel coding with error protection
JP3151874B2 (ja) * 1991-02-26 2001-04-03 日本電気株式会社 音声パラメータ符号化方式および装置
US5233660A (en) * 1991-09-10 1993-08-03 At&T Bell Laboratories Method and apparatus for low-delay celp speech coding and decoding

Also Published As

Publication number Publication date
DE4315319A1 (de) 1994-11-10
FI116598B (fi) 2005-12-30
AU6502494A (en) 1994-12-12
HUT73532A (en) 1996-08-28
FI955323A0 (fi) 1995-11-06
HU9503181D0 (en) 1995-12-28
DE59408494D1 (de) 1999-08-19
WO1994027284A1 (de) 1994-11-24
DK0697123T3 (da) 1999-12-13
HU215620B (hu) 1999-01-28
EP0697123A1 (de) 1996-02-21
DE4315319C2 (de) 2002-11-14
US5794183A (en) 1998-08-11
FI955323A (fi) 1995-11-06
ES2136193T3 (es) 1999-11-16
AU679980B2 (en) 1997-07-17

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