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
  • G10L13/00—Speech synthesis; Text to speech systems
• • 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
• • 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
  • G10L2019/0001—Codebooks
  • G10L2019/0004—Design or structure of the codebook
  • G10L2019/0005—Multi-stage vector quantisation
• • 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
  • G10L2019/0001—Codebooks
  • G10L2019/0011—Long term prediction filters, i.e. pitch estimation
• • 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
  • G10L2019/0001—Codebooks
  • G10L2019/0013—Codebook search algorithms

Definitions

• This invention relates generally to speech coders, and more particularly to digital speech coders that use vector excitation sources.
• Speech coders are known in the art. Some speech coders convert analog voice samples into digitized representations, and subsequently represent the spectral speech information through use of linear predictive coding. Other speech coders improve upon ordinary linear predictive coding techniques by providing an excitation signal that is related to the original voice signal. I have described, in previously issued U.S. Patent No. 4,817,157, a digital speech coder having an improved vector excitation source wherein a codebook of excitation vectors is accessed to select an excitation signal that best fits the available information, and hence provides a recovered speech signal that closely represents the original. In general, the resultant decoded speech signal will more closely represent the original unencoded speech signal if there is a significant number of candidate excitation vectors available for consideration as the excitation source. Increasing performance in this way, however, generally results in enlargement of the codebook size, and this will usually increase processing complexity and data rates.
• the coder when encoding a signal sample, such as a speech sample, the coder first determines a pitch period parameter for the speech sample. Relying in part upon this pitch period parameter, a particular coded excitation signal can be determined independent of the pitch filter coefficient, following which the pitch filter coefficient parameter can be optimized for that particular speech sample. This methodology allows candidate excitation signals to be considered without requiring a commensurate increase in processing complexity or data rates.
• the coded excitation signal is determined substantially independent from any pitch information.
• candidate excitation signals as provided by a codebook are processed to substantially remove components that are representable, at least in part, by a reference component that is related, at least in part, to the intermediate pitch vector. More particularly, the vector component related to the intermediate pitch vector is removed from the candidate excitation signal (a process known as orthogonalizing).
• the orthogonalized candidate excitation signals are then compared with the unencoded speech sample to identify the candidate excitation signal that best represents this particular speech sample.
• the pitch information including a pitch filter coefficient parameter, can be optimized later to best suit the selected excitation signal to thereby yield an overall optimized coded representation of the speech signal.
• a second codebook of candidate excitation signals wherein two excitation signals are used to represent the speech sample.
• the first excitation signal can be selected as described above, and the second excitation signal can be selected in a similar manner, wherein candidate second excitation signals are first orthogonalized with respect to both the intermediate pitch vector and the previously selected first excitation signal.
• Fig. 1 comprises a block diagrammatic depiction of the invention
• Fig. 2 comprises a simple vector diagram representing one aspect of the invention. Best Mode For Carrying Out The Invention:
• This invention can be embodied in a speech coder that makes use of an appropriate digital signal processor such as a Motorola DSP 56000 family device.
• the computational functions of such a DSP embodiment are represented in Fig. 1 as a block diagram equivalent circuit.
• a pitch period parameter (101 ) (determined in accordance with prior art technique) is provided to a pitch filter state (102) that comprises part of a pitch filter.
• the resultant signal (103) comprises an intermediate pitch vector that is provided to both a first multiplier (104) and two orthogonalizing processes (106 and 107) as described below in more detail.
• This first multiplier (104) functions to multiply the resultant signal by a pitch filter coefficient (108) to yield a pitch filter output (109). Selection of the pitch filter coefficient (108) will be described below in more detail.
• a first codebook (111) includes a set of basis vectors that can be linearly combined to form a plurality of resultant excitation signals.
• the number of possible resultant excitation signals can be, for example, between 64 and 2,048, with more of course being possible when appropriate to a particular application.
• the problem, when encoding a particular speech sample, is to select whichever of these excitation sources best represents the corresponding component of the original speech information.
• the excitation signals formulated by the first codebook (111 ) will be presented in seriatim fashion as candidate excitation sources.
• Each candidate excitation source will first be orthogonalized (106) with respect to the resultant signal. For example, referring momentarily to Fig. 2, if vector A were considered to represent the resultant signal and vector B were to represent a particular candidate excitation source, orthogonalization of the candidate excitation source signal would result in the vector denoted by reference character B'.
• the vector dimension space is a function of the number of samples comprising the vectors, which may be upwards of 40 samples or more.
• the candidate excitation vectors may be readily orthogonalized by orthogonalizing the basis vectors, wherein linear combinations of the orthogonadized basis vectors with one another will result in orthogonalized excitation vectors.
• the resulting candidate excitation source can be compared (112) with the unencoded signal (113) (or an appropriate representative signal based thereon) to determine the relative similarity or disparity between the two.
• the process is then repeated for each of the excitation sources of the first codebook (111 ).
• a determination can then be made as to which candidate excitation source most closely aligns with the unencoded signal (113).
• a gain factor (1 14) can also be used to modify each candidate excitation source signal, as well understood in the art.
• the excitation source selection and gain compensation can both be accomplished in a substantially simultaneous manner, as also well understood in the art.
• the orthogonalizing process (106) can thereafter be dispensed with and the exact excitation source signal selected (116) through an appropriate control mechanism (117). Thereafter, presuming a single codebook coder, the pitch information can be gated (117) and summed (118) together with the selected excitation source with the pitch filter coefficient (108) and excitation gain (114) optimized such that the combined excitation most closely aligns with the encoded signal (113).
• the pitch period parameter, pitch filter coefficient, and particular excitation source and gain are known, and appropriate representations thereof may be utilized thereafter as representative of the original speech sample.
• an additional codebook (121) can be utilized, which second codebook (121) again includes a plurality of basis vector derived candidate excitation sources.
• the use of such multiple codebooks is understood in the art.
• the candidate excitation sources from the second codebook (121) are orthogonalized (107) with respect to both the resultant signal (103) and the selected excitation source signal from the first codebook (111 ).
• the selection process can then continue as described above, with the orthogonalized candidate excitation source signals from the second codebook (121 ) being compared against a representative unencoded signal (113) to identify the closest fit.
• the pitch filter coefficient (108) and excitation gains (114 and 120) can then be optimized as described above. What is claimed is:

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

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• 1990
  • 1990-04-18 IL IL9411990A patent/IL94119A/en not_active IP Right Cessation
  • 1990-05-02 BR BR909007467A patent/BR9007467A/pt not_active IP Right Cessation
  • 1990-05-02 WO PCT/US1990/002469 patent/WO1991001545A1/en active IP Right Grant
  • 1990-05-02 EP EP90908908A patent/EP0484339B1/de not_active Expired - Lifetime
  • 1990-05-02 KR KR1019910701947A patent/KR950003557B1/ko not_active IP Right Cessation
  • 1990-05-02 AU AU57359/90A patent/AU638462B2/en not_active Expired
  • 1990-05-02 DE DE69032026T patent/DE69032026T2/de not_active Expired - Lifetime
  • 1990-05-02 CA CA002060310A patent/CA2060310C/en not_active Expired - Lifetime
  • 1990-06-19 CN CN90103020A patent/CN1023160C/zh not_active Expired - Lifetime
  • 1990-06-21 NZ NZ234180A patent/NZ234180A/en unknown

Non-Patent Citations (5)

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