US5799272A
Expired / Lapsed
US
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Switched multiple sequence excitation model for low bit rate speech compression
Assignee: ESS Technology Inc
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Filing: 1996-07-01
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Granted: 1998-08-25
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Expires: 2016-07-01
Abstract
Abstract An apparatus for compressing a speech signal into a compressed speech signal that is represented by a plurality of parameters. A time-varying digital filter is used to model the vocal tract. A number of LPC coefficients specify the transfer function of the filter updated on frame basis. An excitation signal is input to the filter analyzed on sub frame basis. This excitation signal includes either an adaptive vector quantiser code or a first pulse sequence, followed by a second pulse sequence. Selection logic is used to determine whether the adaptive vector quantiser code or the first pulse sequence better represents the speech signal. Based thereon, a switch selects either the adaptive vector quantiser code or the first pulse sequence. Thus, the parameters which are transmitted through a channel to a destination decoder include the LPC filter coefficients, either the adaptive vector quantiser code or the first pulse sequence, the second pulse sequence, and one bit indicating the state of the switch.
Inventors
- 👤 Qinglin Zhu
Bibliographic Details
Application Number
—
Jurisdiction
US
Data Source
google_patents
Last Synchronized
2026-11-01 07:39
CPC Classifications
G; G10; G10L; G10L19/00; G10L19/04; G10L19/16; G10L19/18
Claims Analysis
Total Claims: 14
Claim 1
Independent Claim
298 words
An apparatus for compressing a speech signal into a compressed speech signal that is represented by a plurality of parameters, comprising: a time-varying digital filter for modeling a vocal tract, wherein a plurality of coefficients per frame specify a transfer function of the filter; an excitation circuit coupled to the filter for generating an excitation signal as an input to the filter, wherein the excitation circuit generates an adaptive vector quantiser code, a first pulse sequence, and a second pulse sequence for a plurality of subframes, each of the first pulse sequence and the second pulse sequence having delta pulses with varying amplitudes and a time pattern constrained to be equally spaced with a prechosen value so that the first pulse sequence and the second pulse sequence are characterized by the phase and amplitudes of the delta pulses and wherein the second pulse sequence is non-switchable; selection logic coupled to the excitation circuit for determining whether the adaptive vector quantiser code or the first pulse sequence better corresponds to the speech signal by using a normalized cross-correlation function; a switch coupled to the excitation circuit for selecting between a first excitation mode characterized by the adaptive vector quantiser code and a second excitation mode characterized by a first pulse sequence according to the selection logic; a combination circuit coupled to the switch for combining either the selected adaptive vector quantiser code plus the second pulse sequence or the first pulse sequence plus the second pulse sequence, wherein the parameters which are transmitted through a channel to a destination decoder include the plurality of filter coefficients, either the adaptive vector quantiser code or the first pulse sequence, the second pulse sequence, and a bit indicating the state of the switch in order to produce a switched multiple-sequence excitation modeling.
Claim 2
Independent Claim
298 words
An apparatus for compressing a speech signal into a compressed speech signal that is represented by a plurality of parameters, comprising: a time-varying digital filter for modeling a vocal tract, wherein a plurality of coefficients per frame specify a transfer function of the filter; an excitation circuit coupled to the filter for generating an excitation signal as an input to the filter, wherein the excitation circuit generates an adaptive vector quantiser code, a first pulse sequence, and a second pulse sequence for a plurality of subframes, each of the first pulse sequence and the second pulse sequence having delta pulses with varying amplitudes and a time pattern constrained to be equally spaced with a prechosen value so that the first pulse sequence and the second pulse sequence are characterized by the phase and amplitudes of the delta pulses and wherein the second pulse sequence is non-switchable; selection logic coupled to the excitation circuit for determining whether the adaptive vector quantiser code or the first pulse sequence better corresponds to the speech signal by using a normalized cross-correlation function; a switch coupled to the excitation circuit for selecting between a first excitation mode characterized by the adaptive vector quantiser code and a second excitation mode characterized by a first pulse sequence according to the selection logic; a combination circuit coupled to the switch for combining either the selected adaptive vector quantiser code plus the second pulse sequence or the first pulse sequence plus the second pulse sequence, wherein the parameters which are transmitted through a channel to a destination decoder include the plurality of filter coefficients, either the adaptive vector quantiser code or the first pulse sequence, the second pulse sequence, and a bit indicating the state of the switch in order to produce a switched multiple-sequence excitation modeling.
Claim 3
Dependent Claim
48 words
The apparatus of claim 1, wherein the first pulse sequence is comprised of a plurality of bits specifying a phase of a first pulse and amplitudes corresponding to the first pulse and any following pulses, wherein the pulses of the pulse sequence are equally spaced apart in time.
Claim 4
Dependent Claim
73 words
The apparatus of claim 1, wherein the number of bits allocated per parameter on a frame basis includes: 35 bits to represent ten linear predictive code filter coefficients; 44 bits to represent either the adaptive vector quantiser code or the first pulse sequence, whichever is selected; 40 bits to represent the second sequence; and 4 bits to indicate the state of the switch, to result in a bit rate of approximately 4 kbps.
Claim 5
Dependent Claim
19 words
The apparatus of claim 1, wherein the combiner circuit combines additional pulse sequences as a function of channel loading.
Claim 6
Dependent Claim
15 words
The apparatus of claim 1, wherein the pulse sequence is comprised of equally spaced pulses.
Claim 7
Dependent Claim
70 words
The apparatus of claim 5, wherein an optimal phase of a first pulse of the pulse sequence is determined according to a minimum mean-square error: ##EQU10## Where: S w (n) perceptually weighted original speech h(n) impulse response of the filter gli i=1, . . . ,4 pulse amplitudes in MS0 phase initial phase in MS0 or MS1 , here from 0 to 15 or from 0 to 30 for example!.
Claim 8
Independent Claim
267 words
A method for compressing a speech signal into a compressed speech signal that is represented by a plurality of parameters, comprising the steps of; modeling a vocal tract by using a time-varying digital filter, wherein a plurality of coefficients per frame specify a transfer function of the filter; generating an excitation signal, an adaptive vector quantiser code, a first pulse sequence, and a second pulse sequence for a plurality of subframes, each of the first pulse sequence and the second pulse sequence having delta pulses with varying amplitudes and a time pattern constrained to be equally spaced with a prechosen value so that the first pulse sequence and the second pulse sequence are characterized by the phase and amplitudes of the delta pulses and wherein the second pulse sequence is non-switchable; inputting the excitation signal is to the filter; determining whether the adaptive vector quantiser code or the first pulse sequence better corresponds to the speech signal by using a normalized cross-correlation function; selecting between a first excitation mode characterized by the adaptive vector quantiser code and a second excitation mode characterized by a first pulse sequence according to the selection logic; combining either the selected adaptive vector quantiser code plus the second pulse sequence or the first pulse sequence plus the second pulse sequence, wherein the parameters which are transmitted through a channel to a destination decoder include the plurality of filter coefficients, either the adaptive vector quantiser code or the first pulse sequence, the second pulse sequence, and a bit indicating the state of the switch in order to produce a switched multiple-sequence excitation modeling.
Claim 9
Independent Claim
267 words
A method for compressing a speech signal into a compressed speech signal that is represented by a plurality of parameters, comprising the steps of; modeling a vocal tract by using a time-varying digital filter, wherein a plurality of coefficients per frame specify a transfer function of the filter; generating an excitation signal, an adaptive vector quantiser code, a first pulse sequence, and a second pulse sequence for a plurality of subframes, each of the first pulse sequence and the second pulse sequence having delta pulses with varying amplitudes and a time pattern constrained to be equally spaced with a prechosen value so that the first pulse sequence and the second pulse sequence are characterized by the phase and amplitudes of the delta pulses and wherein the second pulse sequence is non-switchable; inputting the excitation signal is to the filter; determining whether the adaptive vector quantiser code or the first pulse sequence better corresponds to the speech signal by using a normalized cross-correlation function; selecting between a first excitation mode characterized by the adaptive vector quantiser code and a second excitation mode characterized by a first pulse sequence according to the selection logic; combining either the selected adaptive vector quantiser code plus the second pulse sequence or the first pulse sequence plus the second pulse sequence, wherein the parameters which are transmitted through a channel to a destination decoder include the plurality of filter coefficients, either the adaptive vector quantiser code or the first pulse sequence, the second pulse sequence, and a bit indicating the state of the switch in order to produce a switched multiple-sequence excitation modeling.
Claim 10
Dependent Claim
49 words
The method of claim 7, wherein the first pulse sequence is comprised of a plurality of bits specifying a phase of a first pulse and amplitudes corresponding to the first pulse and any following pulses and wherein the pulses of the pulse sequence are equally spaced apart in time.
Claim 11
Dependent Claim
73 words
The method of claim 7, wherein the number of bits allocated per parameter on a frame basis includes: 35 bits to represent ten linear predictive code filter coefficients; 44 bits to represent either the adaptive vector quantiser code or the first pulse sequence, whichever is selected; 40 bits to represent the second sequence; and 4 bits to indicate the state of the switch, to result in a bit rate of approximately 4 kbps.
Claim 12
Dependent Claim
20 words
The method of claim 7 further comprising the step of combining additional pulse sequences as a function of channel loading.
Claim 13
Dependent Claim
15 words
The method of claim 7, wherein the pulse sequence is comprised of equally spaced pulses.
Claim 14
Dependent Claim
57 words
The method of claim 7, wherein an optimal phase of a first pulse of the pulse sequence is determined according to a minimum mean-square error: ##EQU11## Where: S w (n) perceptually weighted original speech h(n) impulse response of the filter gli i=1, . . .,4 pulse amplitudes in MS 0 phase initial phase in MS0 or MS1.
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