WO1994007313A1 - Codec de signaux vocaux - Google Patents

Codec de signaux vocaux Download PDF

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Publication number
WO1994007313A1
WO1994007313A1 PCT/DE1993/000839 DE9300839W WO9407313A1 WO 1994007313 A1 WO1994007313 A1 WO 1994007313A1 DE 9300839 W DE9300839 W DE 9300839W WO 9407313 A1 WO9407313 A1 WO 9407313A1
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WO
WIPO (PCT)
Prior art keywords
mode
speech
channel
bit rate
encoder
Prior art date
Application number
PCT/DE1993/000839
Other languages
German (de)
English (en)
Inventor
Jörg-Martin Müller
Original Assignee
Ant Nachrichtentechnik Gmbh
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 Ant Nachrichtentechnik Gmbh filed Critical Ant Nachrichtentechnik Gmbh
Priority to AU49434/93A priority Critical patent/AU4943493A/en
Publication of WO1994007313A1 publication Critical patent/WO1994007313A1/fr

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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
    • G10L19/04—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
    • 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
    • G10L19/06—Determination or coding of the spectral characteristics, e.g. of the short-term prediction coefficients
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04B—TRANSMISSION
    • H04B14/00—Transmission systems not characterised by the medium used for transmission
    • H04B14/02—Transmission systems not characterised by the medium used for transmission characterised by the use of pulse modulation
    • H04B14/04—Transmission systems not characterised by the medium used for transmission characterised by the use of pulse modulation using pulse code modulation
    • H04B14/046—Systems or methods for reducing noise or bandwidth

Definitions

  • the invention relates to a method for coding speech signals according to the preamble of claim 1.
  • speech coding methods are known, for example from German Patent 38 34 871.
  • a common feature of all speech coding methods is a prediction analysis of the input signal (linear prediction coder, LPC).
  • the voice signal at the input of the encoder is subdivided within a certain period of 20-30 ms, for example.
  • Each speech frame is subjected to a linear prediction analysis in the encoder, which removes linear dependencies in the speech signal.
  • the linear prediction is carried out with the help of FIR filters (Finite Impulse Response).
  • FIR filters Finite Impulse Response
  • the coefficients of these filters are determined anew in every frame, ie these are adaptive filters.
  • CELP Code Excited Linear Prediction
  • SELP Stochastically Excited Linear Prediction
  • the filter coefficients of the short-term predictor are determined once per speech frame, while the coefficients of the long-term predictor are typically determined four times per speech frame.
  • the so-called residual the error signal of the LPC analysis
  • the residue is generated by a Gaussian-distributed random sequence, with a code book containing the random vectors searched and the vector is selected that generates the smallest error in the synthesized speech signal. Then only the addresses of the selected vectors in the code book are to be transmitted.
  • Good voice quality is generally required for voice transmission, both with error-free and with disturbed channels.
  • a redundancy R is added to the bits from the speech encoder in digital voice transmissions, this is called channel coding, in order to be able to correct transmission errors on the receiving side. Since the channel capacity is a predefined and unchangeable system size, transmission errors can no longer be corrected in the case of certain channel disturbances, which is why the quality or intelligibility of the received speech signal suffers as a result.
  • the present invention was based on the object of specifying a speech coding method of the type mentioned at the outset, which is capable of increasing the quality or intelligibility of the speech transmitted over a channel both in the case of interference-free and disturbed channel, i.e. to increase both the voice quality with error-free transmission and the robustness of the voice transmission system.
  • the invention is based on the knowledge that a speech signal can be divided into three classes:
  • this smaller bit rate is used to encode the voiced speech sections, this can be done statistically for about 45 to 50% of the total speech transmission, whereby the speech quality of the speech codec is not deteriorated if the channel is free of errors, but the quality is significantly increased if the channel is disturbed .
  • the method according to the invention it is additionally proposed to force mode 1 independently of the statistics of the speech input signal if the channel interference exceeds a certain level and the intelligibility would thus be greatly reduced.
  • the method according to the invention significantly increases the robustness and thus the quality or intelligibility in the case of a disturbed or severely disturbed channel. If a measure of the level of the interference is available as a signal, the reception quality can be controlled in the transmitter using this. This is possible, for example, if a return channel is available, via which a corresponding signal is transmitted back from the receiver to the transmitter as a measure of the quality of the received signal.
  • Figure 1 and Figure 2 show block diagrams for speech and channel coders or decoders with variable bit rate.
  • Figure 3 demonstrates a radio transmission system with return channel
  • Figure 4 is the structure of a variable language coder
  • FIG. 5 is a flow chart of a
  • the channel coding on the transmitting and receiving sides is adapted to the bit rate of the speech codec.
  • the mode in which the language encoder works is signaled with a so-called mode bit. This mode bit must be reconstructed on the receiving side in the channel decoder.
  • Figures 1 and 2 give an overview of the transmitting and receiving part.
  • the bit rate of the encoder part is controlled by two blocks. On the one hand, this is the voiced / unvoiced decision maker SH / SL, who statistically evaluates the speech input signal s (n).
  • the language coder SE is informed whether a language frame is voiced or unvoiced.
  • the encoder is switched to mode 1, in which differential coding of the pitch analysis parameters is used.
  • This differential coding of the pitch analysis parameters can also be enforced independently of the statistics of the input signal by appropriate setting of the parameters relevant for this in the block external control AS.
  • the percentage of speech frames transmitted with mode 1, i.e. differential coding can be increased and an optimal setting between speech quality and robustness of the channel can be achieved become.
  • the method according to the invention uses two channel encoders KEO and KE1, which encode the encoded speech parameters generated by the speech encoder and the mode bit in mode 0 with the bit rate BO and in mode 1 with the bit rate B1, where B0 is greater than B1.
  • the receiver according to Figure 2 contains a module for mode determination, which switches the channel signal to be decoded in mode 0 to the channel decoder KDO and in mode 1 to the channel decoder KD1.
  • the output signals of the two channel decoders are decoded by the subsequent speech decoder SD into the output speech signal s (n).
  • FIG. 3 shows a radio transmission system with a return channel, the modules according to FIGS. 1 and 2 being contained in simplified form.
  • the reception quality is determined at the modulator output of the receiver and transmitted to the transmitter.
  • the received quality signals act directly on an external control AS, through which the language encoder SE can be switched to the mode with differential coding. If the reception quality is poor, the percentage of delta-coded speech frames (mode 1) can be increased. Although the voice quality deteriorates slightly, the robustness against transmission errors and thus the quality of the receiver is improved. If the reception quality improves, the proportion of Mode 1 speech frames is reduced to the normal proportion, and the speech quality is correspondingly better. It is thus possible to dynamically adapt the speech codec to the channel conditions in a simple manner.
  • FIG. 1 shows a radio transmission system with a return channel, the modules according to FIGS. 1 and 2 being contained in simplified form.
  • the reception quality is determined at the modulator output of the receiver and transmitted to the transmitter.
  • the received quality signals act directly on an external control AS,
  • the blocks excitation analysis and LPC analysis are carried out as in known CELP methods (see reference 1).
  • the long-term prediction parameters are determined using the also known closed-loop method (reference 2).
  • the parameters of the LPC analysis are determined, for example, once per speech frame (for example 20 ms) and the long-term prediction analysis N su b Ma - L ( Z - B - every 5 ms) per frame.
  • the speech subframe On The speech section for which the long-term prediction parameters are determined is referred to as the speech subframe.
  • the long-term predictor can be represented as an adaptive code book.
  • the code book consists of 256 signals, for example
  • N s L (n) n 0 ... -1
  • the error energy between the prediction signal and the speech signal s (n) serves as a measure of the quality of the prediction
  • variable bit rate speech codec Only the function blocks that are relevant for the variable bit rate speech codec are described below.
  • This decision maker is an "open loop" pitch analysis which is carried out in three steps:
  • T G is a threshold that is set in the "External control" module or dynamically when using a return channel.
  • P optimal delta pitch period, which was calculated on the condition that a delta coding for the pitch period of the last speech subframe is possible with a predetermined number of bits.
  • ⁇ * optimal delta scaling factor for the adaptive codebook vector, which was calculated on the condition that a delta coding to the value of the scaling factor of the last speech subframe is possible with a predetermined number of bits
  • E ⁇ error energy of the "closed loop" pitch predictor if the pitch period and the scaling factor are differentially coded with the corresponding values of the last speech subframe with a predetermined number of bits.
  • P SH Pitch period from the "Voiced / Unvoiced” decision maker. In the voiced case, this value defines the delta environment for the "closed loop” pitch in the first language subframe.
  • T G defines by means of which a differential coding can be forced. These parameters are either fixed or can be varied in time by evaluating the return channel information.
  • the difference coding is applied to the pitch period and to the scaling factor by the difference between the current and the last calculated
  • Parameters are coded and transmitted.
  • the bit rate for transmitting the long-term prediction parameters is 2.4 kbit / sec in mode 0 and 1.8 kbit / sec in mode 1.
  • no differential coding can be carried out for the first parameter of a speech subframe, which is why no bit rate can be saved at this point.

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  • Engineering & Computer Science (AREA)
  • Signal Processing (AREA)
  • Physics & Mathematics (AREA)
  • Computational Linguistics (AREA)
  • Health & Medical Sciences (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • Human Computer Interaction (AREA)
  • Acoustics & Sound (AREA)
  • Multimedia (AREA)
  • Spectroscopy & Molecular Physics (AREA)
  • Computer Networks & Wireless Communication (AREA)
  • Compression, Expansion, Code Conversion, And Decoders (AREA)

Abstract

L'invention concerne un procédé de codage de signaux vocaux à transmettre d'un émetteur à un récepteur, en passant par un canal à capacité de transmission (BBR) limitée, avec un codeur de signaux vocaux et un codeur de canaux, qui se caractérise en ce que le codeur de signaux vocaux et le codeur de canaux comportent chacun deux modes différents. Dans le premier mode, le signal vocal est codé par le codeur de signaux vocaux avec un débit binaire (B1) moins important que dans le second mode (B0). Dans le premier mode, la différence des débits binaires (B0-B1) par rapport au débit binaire (B0) du second mode est mise à la disposition du codeur de canaux. Cette différence de débits binaires supplémentaire est utilisée par le codeur de canaux pour transmettre d'autres informations de redondance. Ce procédé permet d'améliorer la qualité de transmission de la voix et peut être utilisé par exemple dans les radiotéléphones mobiles.
PCT/DE1993/000839 1992-09-24 1993-09-11 Codec de signaux vocaux WO1994007313A1 (fr)

Priority Applications (1)

Application Number Priority Date Filing Date Title
AU49434/93A AU4943493A (en) 1992-09-24 1993-09-11 Speech codec

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
DEP4231918.8 1992-09-24
DE19924231918 DE4231918C1 (de) 1992-09-24 1992-09-24 Verfahren für die Codierung von Sprachsignalen

Publications (1)

Publication Number Publication Date
WO1994007313A1 true WO1994007313A1 (fr) 1994-03-31

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AU (1) AU4943493A (fr)
DE (1) DE4231918C1 (fr)
WO (1) WO1994007313A1 (fr)

Cited By (5)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO1996019880A1 (fr) * 1994-12-19 1996-06-27 Nokia Telecommunications Oy Procede de transmission de donnees, systeme de transmission de donnees et systeme de radio cellulaire
EP0803989A1 (fr) * 1996-04-26 1997-10-29 Deutsche Thomson-Brandt Gmbh Procédé et appareil pour le codage d'un signal audio-nimérique
WO1997041549A1 (fr) * 1996-04-26 1997-11-06 Telefonaktiebolaget Lm Ericsson Procede de commande de mode de codage et appareil de determination de mode de decodage
US6009399A (en) * 1996-04-26 1999-12-28 Deutsche Thomson-Brandt Gmbh Method and apparatus for encoding digital signals employing bit allocation using combinations of different threshold models to achieve desired bit rates
US6134220A (en) * 1994-04-13 2000-10-17 Alcatel Cit Method of adapting the air interface in a mobile radio system and corresponding base transceiver station, mobile station and transmission mode

Families Citing this family (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US6134521A (en) * 1994-02-17 2000-10-17 Motorola, Inc. Method and apparatus for mitigating audio degradation in a communication system

Citations (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO1989012292A1 (fr) * 1988-06-08 1989-12-14 Fujitsu Limited Appareil codeur/decodeur
US5060269A (en) * 1989-05-18 1991-10-22 General Electric Company Hybrid switched multi-pulse/stochastic speech coding technique

Family Cites Families (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
DE3834871C1 (en) * 1988-10-13 1989-12-14 Ant Nachrichtentechnik Gmbh, 7150 Backnang, De Method for encoding speech

Patent Citations (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO1989012292A1 (fr) * 1988-06-08 1989-12-14 Fujitsu Limited Appareil codeur/decodeur
US5060269A (en) * 1989-05-18 1991-10-22 General Electric Company Hybrid switched multi-pulse/stochastic speech coding technique

Non-Patent Citations (1)

* Cited by examiner, † Cited by third party
Title
TANIGUCHI: "Combined Source and Channel Coding Based on Multimode Coding", ICASSP 90, SPEECH PROCESSING 1, vol. 1, April 1990 (1990-04-01), NY,USA, pages 477 - 480, XP000146509 *

Cited By (15)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US6456598B1 (en) 1994-04-13 2002-09-24 Alcatel Cit Method of adapting the air interface in a mobile radio system and corresponding base transceiver station, mobile station and transmission mode
US6134220A (en) * 1994-04-13 2000-10-17 Alcatel Cit Method of adapting the air interface in a mobile radio system and corresponding base transceiver station, mobile station and transmission mode
AU698404B2 (en) * 1994-12-19 1998-10-29 Nokia Telecommunications Oy Data transmission method, data transmission system, and cellular radio system
WO1996019880A1 (fr) * 1994-12-19 1996-06-27 Nokia Telecommunications Oy Procede de transmission de donnees, systeme de transmission de donnees et systeme de radio cellulaire
US6092222A (en) * 1994-12-19 2000-07-18 Nokia Telecommunications Oy Data transmission method, data transmission system, and cellular radio system
WO1997041662A1 (fr) * 1996-04-26 1997-11-06 Telefonaktiebolaget Lm Ericsson (Publ) Procede et appareil de commande de mode de codage de source/voie
US5982766A (en) * 1996-04-26 1999-11-09 Telefonaktiebolaget Lm Ericsson Power control method and system in a TDMA radio communication system
US6009399A (en) * 1996-04-26 1999-12-28 Deutsche Thomson-Brandt Gmbh Method and apparatus for encoding digital signals employing bit allocation using combinations of different threshold models to achieve desired bit rates
AU720308B2 (en) * 1996-04-26 2000-05-25 Telefonaktiebolaget Lm Ericsson (Publ) Encoding mode control method and decoding mode determining apparatus
WO1997041663A1 (fr) * 1996-04-26 1997-11-06 Telefonaktiebolaget Lm Ericsson (Publ) Appareil et procede de commande adaptative de mode de codage dans un systeme de radiocommunication amrt
WO1997041549A1 (fr) * 1996-04-26 1997-11-06 Telefonaktiebolaget Lm Ericsson Procede de commande de mode de codage et appareil de determination de mode de decodage
US6163577A (en) * 1996-04-26 2000-12-19 Telefonaktiebolaget Lm Ericsson (Publ) Source/channel encoding mode control method and apparatus
US6195337B1 (en) 1996-04-26 2001-02-27 Telefonaktiebolaget Lm Ericsson (Publ) Encoding mode control method and decoding mode determining apparatus
EP0803989A1 (fr) * 1996-04-26 1997-10-29 Deutsche Thomson-Brandt Gmbh Procédé et appareil pour le codage d'un signal audio-nimérique
MY119786A (en) * 1996-04-26 2005-07-29 Ericsson Telefon Ab L M Power control method and system in a tdma radio communication system.

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Publication number Publication date
AU4943493A (en) 1994-04-12
DE4231918C1 (de) 1993-12-02

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