EP0689191B1 - Sprachverarbeitungsvorrichtung und Mobilfunkendgerät - Google Patents
Sprachverarbeitungsvorrichtung und Mobilfunkendgerät Download PDFInfo
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- EP0689191B1 EP0689191B1 EP95201578A EP95201578A EP0689191B1 EP 0689191 B1 EP0689191 B1 EP 0689191B1 EP 95201578 A EP95201578 A EP 95201578A EP 95201578 A EP95201578 A EP 95201578A EP 0689191 B1 EP0689191 B1 EP 0689191B1
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- speech
- values
- delay
- signal
- speech signal
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
- G10L21/00—Processing of the speech or voice signal to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
- G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
- G10L21/0208—Noise filtering
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
- G10L21/00—Processing of the speech or voice signal to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
- G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
- G10L21/0208—Noise filtering
- G10L21/0216—Noise filtering characterised by the method used for estimating noise
- G10L2021/02161—Number of inputs available containing the signal or the noise to be suppressed
- G10L2021/02166—Microphone arrays; Beamforming
Definitions
- the invention relates to a mobile radio terminal with a Speech processing device.
- Speech signals In the field of language processing are often to be processed Speech signals contain noise signal components, which helps to reduce the Speech quality and, in particular, deteriorated Speech intelligibility leads. This problem occurs, for example, with mobile radio terminals on that are used in motor vehicles and a Have handsfree. Voice signals from in the motor vehicle Arranged microphones of the speakerphone are received On the one hand, voice signal components that the respective user (language source) of the Mobile terminal are generated within the motor vehicle, and on the other hand Noise signal components resulting from other ambient noises and during a Driving essentially consist of engine and driving noises.
- the corresponding one Delay value is an integer multiple of a sampling interval Signals rounded. Problems of convergence occur in such a way that when they are reached very small error values strong oscillations of the rounded delay values occur.
- the invention has for its object the speech quality of the processed Improve speech signals and reduce convergence problems.
- the gradient estimates serve to estimate the respective gradient of the Performance of the error values or in other words the squared error values.
- the Control means determine the delay estimates such that the performance of the Error values is reduced.
- the convergence of the Delay estimates determined delay values significantly improved because the delay estimates versus the delay values due to the Rounding have a higher resolution. Oscillations of the delay values are essentially avoided.
- the resolution of the delay values is chosen lower than the resolution of the delay estimated values by the to keep the technical effort involved in delaying the speech signals as low as possible.
- the signal / noise ratio and the speech quality of one at the output of the Adding device applied sum signal are compared to the signal / Noise power ratio and the speech quality of the individual speech signals improved.
- the digital filter is a digital Hilbert transformer.
- a digital Hilbert transformer that has a phase shift of 90 degrees for causes all frequencies, has the transfer function of an amount Low pass filter, so that it is particularly important for the low and for a speech signal Frequencies the rounded delay values converge well.
- the Hilbert transformer can also be replaced by a differentiator, for example, which also causes a phase shift of 90 degrees.
- one Differentiators a linearly increasing transfer function, so that in particular the low frequencies of a speech signal are suppressed, so that convergence is not as good as with a Hilbert transformer.
- the speech processing device is for Processing of three voice signals provided.
- the signal / noise ratio and the speech quality of the applied at the output of the adder Improve the sum signal.
- the invention can also be designed in that for determining a Delay estimate for the further speech signal the use of a Linear combination of error values is provided.
- delay means are provided Delay of the first speech signal is provided with a fixed delay time.
- the speech processing device is shown in FIG a hands-free system is integrated.
- the speech processing device shown in Fig. 1 contains two microphones M1 and M2. These are used to convert acoustic to electrical Speech signals, which are composed of speech and noise signal components.
- the Speech signal components come from a single speech source (speaker), which in the Usually different distances to the two microphones Ml and M2 having. The speech signal components are thus highly correlated.
- the Noise signal components of the two received by the microphones M1 and M2 Speech signals are not generated by the single speech source
- Ambient noise which is in the range of 10 at suitable microphone distances up to 60 cm can be assumed to be uncorrelated or only slightly correlated, when the microphones are in a so-called reverberated environment like for example, in the car or in an office.
- the noise signal components are caused in particular by engine and driving noises.
- the microphone signals generated by the microphones M1 and M2 are digitized by analog-digital converters 1 and 2.
- the resulting digitized and thus present as samples x1 (i) and x2 (i) microphone signals are evaluated by a control device 3, which is used to control and set a delay element 4.
- the sampled microphone signals x1 (i) and x2 (i) are referred to below as microphone or speech signals.
- the delay element 4 delays the microphone signal x1 with delay values T1 that can be set by the control device 3.
- An adding device 5 adds the microphone signal x1 (i) delayed by the delay element 4 and the microphone signal x2 (i) delayed by a delay element 16 with a constant time delay T max .
- the delay element 16 is provided in order to be able to set both a leading and a lagging of the microphone signal x1 (i) relative to the microphone signal x2 (i).
- a sum signal X (i) present at the output of the adding device 5 is a sampled speech signal, the signal / noise power ratio of which is increased compared to the signal / noise power ratios of the speech signals x1 (i) and x2 (i).
- the addition by the adder 5 increases the power of the voice signal components of the two voice signals x1 (i) and x2 (i) by approximately a factor of 4 and increases the power of the noise signal components only approximately caused by a factor of 2. This results in an improvement in the power-related signal / noise power ratio of approximately 3 dB.
- the speech signal estimates x1 int (i) are values that result from an interpolation of samples of the speech signal x1 (i). The determination of the speech signal estimates x1 int (i) will be explained later.
- i is a variable which can take integer values and with which, on the one hand, sampling times of the speech signals x1 (i) and x2 (i) and, on the other hand, program cycles of the programmable and control device 3 having control means 3 are indicated, with a new sample value per speech signal in each program cycle is processed.
- a digital filter 6 carries out a Hilbert transformation of the sample values x2 (i):
- the digital filter 6 supplying the values x2 H (i) of x2 (i) is an FIR filter of the order K, which has coefficients h (0), h (1), ..., h (K).
- K is sixteen, so that the digital filter 6 has seventeen coefficients.
- the amount of the digital filter 6 has the transfer function of a low pass. It continues to produce a 90 degree phase shift.
- the fixed phase shift of 90 degrees is the decisive property of the digital filter 6, the course of the amount of the transfer function is not decisive for the functioning of the speech processing device.
- the digital filter 6 can thus also be implemented with the aid of a differentiator, which would, however, lead to a suppression of low-frequency components of x2 (i) and thus to a reduced performance of the speech processing device.
- N indicates the number of samples of x2 used in the calculation. N is, for example, equal to 65.
- a function block 7 continuously forms from the samples of the speech signal x2 (i) Estimates SNR (i) of the associated signal-to-noise power ratio, which of a function block 8 can be evaluated.
- An evaluation of the Speech signal x1 (i) instead of the speech signal x2 (i) is possible without the Functionality of the speech processing device is restricted.
- the Operation of the function block 7 will be explained later with reference to FIGS. 6 to 8 explained in more detail.
- Function block 8 makes a threshold decision regarding the estimated values SNR (i). Only if the estimated values SNR (i) are above a predeterminable threshold, a buffer 9 with the newly determined Gradient estimate grad (i) overwritten.
- the memory content (degree (i)) of the buffer 9 is from a Functional unit 10 processed.
- the buffer 9 is not overwritten with the newly determined gradient estimated value grad (i) and it retains its old memory content at what is due to the open position of the switch 11 is symbolized.
- the predefinable threshold from which the opening and closing of the Switch 11 depends on the function block 8, is preferably between 0 and 10 dB.
- the intermediate memory 9 supplies the gradient estimated values grad (i) stored in it to the functional unit 10, to which sample values of the speech signal x1 (i) are also supplied and which is used both for supplying the speech signal estimated values x1 int (i) and for setting the delay element 4.
- ⁇ is a constant that has the value 0.95 in the exemplary embodiment.
- ⁇ is a constant factor or convergence parameter and is in the range 0 ⁇ ⁇ 1 10 * R x2x2 R x2x2 denotes an autocorrelation function of the speech signal x2 (i) at the zero position.
- a particularly advantageous value range of ⁇ in the present exemplary embodiment is 1.5 ⁇ ⁇ ⁇ 3.
- the delay estimated values T1 '(i) cannot do non-integer values be integer multiples of a sampling interval.
- a function block 14 rounds out the Delay estimated values T1 '(i) to integer delay values T1 (i), with which the delay device 4 is set. The rounding operation through Function block 14 is necessary because the values of the delay element 4 increase delaying speech signal x1 (i) only at the corresponding sampling times available.
- Function block 15 is thus able to use the speech signal estimate x1 int (i) in program cycle i to form or interpolate a value of speech signal x1 at time i + T1 (i), ie at a time between two sampling times.
- the described interpolation by function block 15 can be replaced by function block 15 performing low-pass filtering of the sample values x1 (i) for the interpolation of values between the sample times.
- the corresponding true time delay between the speech signal components which is determined by the different distances from the speaker to the microphones M1 and M2, would lie between these two delay values.
- such oscillations are avoided by using speech signal estimates x1 int (i) in the formation of the error values, by means of which the values of the speech signal x1 (i) are also available for delays by non-integer multiples of a sampling interval, i.e. also at points in time not equal to the sampling times i of the speech signal x1 (i).
- the function block 12 used to smooth the gradient estimates grad (i) causes an improved determination of the delay estimated values T1 '(i).
- the control device 3 adapts the delay estimated values T1 '(i) or the delay values T1 (i) so that the square or the power of the error values e 12 (i) is reduced from one program cycle to the next. The convergence of T1 '(i) or T1 (i) is thus ensured.
- FIG. 3 shows a speech processing device which works in principle like the speech processing device from FIG. 1 and now has three microphones M1, M2 and M3 for the delivery of microphone or speech signals.
- the microphone signals are fed to analog-to-digital converters 20, 21 and 22, which deliver digitized and thus sampled speech signals x1 (i), x2 (i) and x3 (i), which consist of speech and noise signal components.
- the speech signals x1 (i) and x3 (i) are supplied to adjustable delay elements 23 and 24.
- the speech signal x2 (i) is fed to a delay element 27 with a fixed delay time T max.
- the output values of the delay elements 23, 24 and 27 are added to the sum signal X (i) by an adding device 25.
- a control device 26 evaluates the sample values of the speech signals x1 (i), x2 (i) and x3 (i) and derives rounded integer delay values T1 (i) and T3 () from these sample values analogously to the mode of operation of the control device 3 from FIGS. i) ab, which correspond to the integer multiples of a sampling interval of the sampled speech signals x1 (i), x2 (i) and x3 (i) and with which the delay elements 23 and 24 are set, so that an expansion from two to three microphone to be processed or voice signals is enabled.
- FIG. 4 shows a first embodiment of the control device 26 from FIG. 3 shown.
- Two functional units 10 are provided, the structure of which is the same the structure of the functional unit 10 from FIG. 2 and for setting the Delay elements 23 and 24 with the rounded time delay values T1 (i) and T3 (i) serve.
- the upper functional unit 10 provides speech signal estimates x1 int (i).
- the lower functional unit 10 supplies speech signal estimates x3 int (i).
- Error values e 12 (i) and e 32 (i) are formed from a difference x1 int (i) - x2 (i) and from a difference x3 int (i) - x2 (i).
- a digital filter 6 which has already been described in more detail in the explanations relating to FIG. 2, and which serves to receive the sample values x2 (i) and to supply values x2 H (i) which are obtained by a Hilbert transformation of the Samples x2 (i) are generated.
- the values x2 H (i) are multiplied on the one hand by the error values e 12 (i) and on the other hand by the error values e 32 (i).
- the first product x2 H (i) * e 12 (i) is fed to the upper, the second product x2 H (i) * e 32 (i) to the lower functional unit 10.
- the arrangement of the function blocks 7 and 8, the buffer 9 and the switch 11 is carried out analogously to FIG. 2 and is not shown in FIG. 4 for reasons of clarity.
- FIG. 5 shows a version of the control device 26 that is expanded compared to FIG. 4.
- three digital filters 6 are now arranged. These form the values x1 H (i), x2 H (i) and x3 H (i) from the speech signal samples x1 (i), x2 (i) and x3 (i) by Hilbert transformation.
- error values e 13 (i) are formed from the difference x1 int (i) -x2 (i), which into a first product 0.3 * e 13 (i) * x3 H (i) die.
- a second product results from 0.7 * e 12 (i) * x2 h (i).
- the two products correspond to weighted gradient estimates of the squares of the error values e 13 (i) and e 12 (i).
- the sum of the first and second product and thus a linear combination of the weighted gradient estimated values is fed to the upper functional unit 10.
- error values e 31 (i) and e 32 (i) are formed in the lower half of the block diagram shown in FIG. 5.
- the error values e 31 (i) result from the difference x3 int (i) -x1 (i).
- the error values e 32 (i) are formed by the difference x3 int (i) -x2 (i).
- a third product 0.3 * e 31 (i) * x1 H (i) and a fourth product 0.7 * e 32 (i) * x2 H (i) are added and the resulting sum is fed to the lower functional unit 10 .
- the one 4 or 5 contains control device can be compared to the Improved speech processing device with two microphones according to FIG. 1 Generate sum signal X (i).
- the signal / noise ratio and thus the Speech quality of the sum signal X (i) of the speech processing device according to FIG. 3 is compared to that of the speech processing device according to FIG. 1 generated sum signal X (i) further increased.
- 5 points to the control device of FIG. 4 when used in the 3 has an increased stability.
- the delay estimates T1 '(i) and T3' (i) e.g. B. floating point numbers
- T1 (i) and T3 (i) Values are rounded to an integer multiple of a sampling interval correspond (here: whole numbers), but to values that are multiples of one Correspond to a fraction of a sampling interval.
- a rounding of the Delay estimates are advantageous to multiples of a value that is a quarter or half a sampling interval.
- the Resolution of the delay values increased which can thus be adjusted more precisely, so that also the speech quality of the sum signals X (i) is further increased because Differences in transit time from the speech source producing the speech signal components the microphones M1, M2 and M3 can be compensated more precisely.
- an interpolation or low pass filtering of Speech signal samples are provided to generate speech signal values that are between two speech signal samples.
- the interpolation or Low-pass filtering can be integrated in the delay means 4, 23 and 24 in particular become.
- the function block 7 from a sampled speech signal x (i), which consists of noise and speech signal components, the associated estimated values SNR (i) of the signal / noise power ratio, that is Ratio of the power of the speech signal components to the power of the noise signal components, determined.
- the sample values x2 (i) correspond to the sample values x (i).
- the function block 7 is shown in FIG. 6 on the basis of a block diagram.
- a function block 30 serves to form power values P x (i) of the sample values x (i) by squaring the sample values. Function block 30 also smoothes these power values P x (i).
- the resulting smoothed power values P x, s (i) are supplied to both function block 31 and function block 32.
- Function block 31 continuously determines estimated values P n (i) for estimating the power of the noise signal component of the sampled values x (i), ie the power of the noise signal components of the sampled values x (i) is determined.
- the function block 32 continuously determines estimated values SNR (i) of the signal / noise power ratio of the sampled values x (i).
- FIG. 7 shows a flow chart which explains the function of the function block 7 in more detail.
- the flow chart shows how estimated values SNR (i) of the corresponding signal / noise power ratio are formed from the sampled values x (i) of the speech signal x by a computer program.
- a counter variable Z is set to 0 and a variable P Mmin is set to a value P max at the beginning of the program described by FIG.
- P max is chosen so large that the smoothed power values P x, s (i) are always smaller than P max .
- P max can, for example, be set to the maximum representable numerical value of a computer used to implement the program.
- a new sample value x (i) is read in in block 34.
- a short-term power value P x (i) of a group of N successive sample values x (i) is determined using formula (9). N here is 128, for example.
- Equation (10) The value ⁇ from equation (10) is between 0.95 and 0.98.
- the determination of smoothed power values P x, s (i) can also only be carried out using equation (10), in which case however the value ⁇ should be increased approximately to the value 0.99 and P x (i) by x 2 (i) is to be replaced.
- a branch 37 queries whether the smoothed power value P x, s (i) that has just been determined is less than P Mmin . If this question is answered in the affirmative, ie P x, s (i) is less than P Mmin , block 38 sets P Mmin to the value of P x, s (i). If the question of branch 37 is answered in the negative, block 38 is skipped. This means that the minimum of M smoothed power values P x, s is in P Mmin after M program cycles . Then the branch 39 is used to query whether the counter variable Z has a value greater than or equal to a value M. In this way it is determined whether M smoothed power values have already been processed.
- SNR (i) [P x, p (i) - min ⁇ c * P n (i), P x, p (i) ⁇ ] / [c * P n (i)] a current estimate SNR (i) of the signal / noise power ratio of the speech signal x (i) is determined.
- the product c * P n (i) is used to estimate the current power of the noise signal component
- the difference P x, s (i) -c * P n (i) is used to estimate the current power of the voice signal component of the voice signal x ( i).
- the current power of the speech signal is estimated by the smoothed power value P x, s (i).
- the weighting with a scaling factor c prevents P n (i) from estimating the noise signal power with a value that is too small.
- the scaling factor c is typically in the range from 1.3 to 2.
- the minimum formation in block 41 or equation (12) ensures that the non-logarithmic signal / noise power ratio SNR (i) is also positive if, in exceptional cases, c * P n (i) is greater than P x, s (i). Then the power of the noise signal component of the voice signal is set equal to the power of the voice signal estimated by P x, s (i).
- the power of the speech signal component of the speech signal estimated by P x, s (i) -P x, s (i) is then equal to zero, as is the non-logarithmic signal / noise power ratio.
- the program continues with the reading in of a new speech signal sample value x (i) by block 34.
- branch 39 If the query of branch 39 is answered in the affirmative, ie M smoothed sample values P x, s (i) have been processed, in block 42 by updated the components of a vector minvec of dimension W. Subsequently, branch 43 queries whether the components minvec 1 to minvec w increase with increasing vector index, ie whether: minvec j + 1 > minvec j for 1 ⁇ j ⁇ W-1
- P n (i) is set equal to P Mmin in block 45, so that an adaptation of the estimation of the noise signal component is accelerated takes place since P n (i) is determined at the minimum of the last (M ⁇ L) values. Then in block 46 the counter variable Z is reset to 0 and P Mmin again receives the value P max .
- M successive smoothed P x, s (i) samples x (i) of the speech signal x are combined into a subgroup.
- the minimum of the smoothed power values P x, s (i) is determined by the operations carried out with branch 37 and block 38.
- the W minima determined last are stored in the components of the vector minvec. If the last W minima are not monotonically increasing (see branch 43), then a preliminary estimate P n (i) of the power of the noise signal component is determined from the minimum of the minima of the last W subgroups, ie from the minimum of a group, according to block 44.
- the minimum of the last subgroup with M smoothed power values P x is determined by block 45 to estimate the current estimated value P n (i) of the power of the noise signal component . s (i) used. This shortens the time period with which monotonically increasing smoothed power values P x, s (i) also cause a change in the estimated values SNR (i).
- the value P n (i) is determined from the minimum of the last W subgroup minima or the last L smoothed power values P x, s (i), which is used to estimate the noise signal power.
- the described speech processing device thus has an estimation device which is suitable for the continuous formation of estimated values SNR (i) of the signal / noise power ratio of noisy speech signals x (i). In particular, no speech pauses are required to estimate the noise signal power.
- the estimation device described uses the special time profile of smoothed power values of the speech signal x (i), which is characterized by peaks and intermediate areas with smaller smoothed power values P x, s (i), their temporal expansion from the respective speech source, ie the respective speaker , depends. The areas between the peaks are used to estimate the power of the noise signal component.
- the groups with L smoothed power values P x, s (i) must follow one another without gaps, ie they must either adjoin or overlap.
- each group must contain so many smoothed power values P x, s (i) that at least all values belonging to any peak can be recorded. Since the most extended peaks can be estimated by the most extended phonemes of a speech signal, ie the vowels, the number L describing the group size can be derived from this. For a sampling rate of the speech signal of 8 kHz, a useful value of L is in the range between 3000 and 8000. An advantageous value for W is 4. With such a dimensioning, there is a good compromise between the computational effort and the speed of reaction of the function block 7.
- FIG. 9 shows a use of the speech processing device from FIG. 3 in a mobile terminal 50 is shown.
- the language processing means 20 to 26 are summarized in a function block 51 which consists of those of the Microphones M1, M2 and M3 generated the microphone or speech signals Sum signal values X (i) forms.
- the microphones M1, M2 and M3 have advantages a distance of 10 to 60 cm, so that in a so-called "reverberated" Environment (e.g. car, office) the interference signal components from the microphones M1, M2 and M3 delivered speech signals are largely uncorrelated. this is also valid when using only two microphones as in Fig. 1.
- processing function block 52 summarizes all other means of the mobile radio terminal 50 for receiving, processing and transmitting signals together, which are used for communication with a base station, not shown, wherein sending and receiving signals via one to function block 52 coupled antenna 54 takes place.
- function block 52 coupled speaker 53 provided.
- Acoustic communication User (speaker, listener) with the mobile terminal 50 takes place via the Microphones M1 to M3 and the speaker 53, the parts of one in the mobile terminal 50 integrated handsfree are.
- the application of such Mobile radio terminal 50 is particularly advantageous in motor vehicles, since there the Hands-free calling via the mobile terminal, in particular through motor or Driving noise (noise) is disturbed.
Description
- zur Bildung von Gradientenschätzwerten durch Multiplikation von Fehlerwerten für zwei Sprachsignale mit den Ausgangswerten eines Digitalfilters, das eine Phasenverschiebung von 90 Grad bewirkt und zur Filterung eines der zwei Sprachsignale dient,
- zur rekursiven Ermittlung von Verzögerungsschätzwerten aus den Gradientenschätzwerten, wobei aus den Verzögerungsschätzwerten durch Rundung die Verzögerungswerte gebildet werden, die zur Einstellung der Verzögerungsmittel dienen und
- zur Bildung jeweils wenigstens eines Fehlerwertes für einen bestimmten Abtastzeitpunkt aus der Differenz zwischen einem Sprachsignalschätzwert, der zur Abschätzung des weiteren Sprachsignals zu einem gegenüber dem bestimmten Abtastzeitpunkt um den Verzögerungsschätzwert verschobenen Zeitpunkt dient und durch Interpolation von Abtastwerten des weiteren Sprachsignals gebildet wird, und dem Abtastwert eines anderen der zu verarbeitenden Sprachsignale zu dem bestimmten Abtastzeitpunkt
- Fig. 1
- eine Sprachverarbeitungsvorrichtung für zwei Sprachsignale,
- Fig. 2
- eine Steuervorrichtung zur Einstellung eines Zeitversatzes zwischen den beiden Sprachsignalen nach Fig. 1,
- Fig. 3
- eine Sprachverarbeitungsvorrichtung für drei Sprachsignale,
- Fig. 4 und 5
- Blockschaltbilder mit Steuervorrichtungen zur Einstellung von Zeitversätzen zwischen den drei Sprachsignalen nach Fig. 3,
- Fig. 6 und 7
- ein Blockschaltbild und ein Flußdiagramm zur Bestimmung des Signal-/ Rauschleistungsverhältnisses eines Sprachsignals,
- Fig. 8
- eine Einteilung von geglätteten Leistungswerten eines Sprachsignals in Gruppen und Untergruppen und
- Fig. 9
- ein Mobilfunkendgerät mit einer Sprachverarbeitungsvorrichtung nach Fig. 1 bis 8.
Claims (8)
- Sprachverarbeitungsvorrichtung zur Verarbeitung eines ersten (x2(i)) und mindestens eines weiteren (x1(i), x3(i)) aus Rausch- und Sprachsignalanteilen bestehenden und als Abtastwerte vorliegenden Sprachsignals mit Verzögerungsmitteln (4, 23, 24) zur Verzögerung des abgetasteten weiteren Sprachsignals (x1(i), x3(i)), mit Steuermitteln (3, 26)zur Bildung von Gradientenschätzwerten (grad(i), sgrad(i)) durch Multiplikation von Fehlerwerten (e12(i), e32(i), e13(i), e31(i)) für zwei Sprachsignale (z.B. x1(i) und x2(i)) mit den Ausgangswerten eines Digitalfilters (6), das eine Phasenverschiebung von 90 Grad bewirkt und zur Filterung eines der zwei Sprachsignale (z.B. x2(i)) dient,zur rekursiven Ermittlung von Verzögerungsschätzwerten (T1'(i), T3'(i)) aus den Gradientenschätzwerten (grad(i), sgrad(i)), wobei aus den Verzögerungsschätzwerten (T1'(i), T3'(i)) durch Rundung die Verzögerungswerte (T2(i), T3(i)) gebildet werden, die zur Einstellung der Verzögerungsmittel (4, 23, 24) dienen undzur Bildung jeweils wenigstens eines Fehlerwertes (e12(i), e32(i), e13(i), e31(i)) für einen bestimmten Abtastzeitpunkt (i) aus der Differenz zwischen einem Sprachsignalschätzwert (x1int(i), x3int(i)), der zur Abschätzung des weiteren Sprachsignals (x1(i), x3(i)) zu einem gegenüber dem bestimmten Abtastzeitpunkt (i) um den Verzögerungsschätzwert (T1'(i), T3'(i)) verschobenen Zeitpunkt dient und durch Interpolation von Abtastwerten des weiteren Sprachsignals (x1(i), x3(i)) gebildet wird, und dem Abtastwert eines anderen der zu verarbeitenden Sprachsignale (x1(i), x2(i), x3(i)) zu dem bestimmten Abtastzeitpunkt (i) und mit einer Addiervorrichtung (5, 25) zum Addieren der gegeneinander zeitversetzten Sprachsignale (x1(i), x2(i), x3(i)).
- Mobilfunkendgerät mit einer Sprachverarbeitungsvorrichtung nach Anspruch 1.
- Mobilfunkendgerät nach Anspruch 2,
dadurch gekennzeichnet,
daß das Digitalfilter (6) ein digitaler Hilbert-Transformator ist. - Mobilfunkendgerät nach Anspruch 3,
dadurch gekennzeichnet,
daß Mittel (12) zur Glättung der Gradientenschätzwerte (grad(i)) vorgesehen sind. - Mobilfunkendgerät nach einem der Ansprüche 2 bis 4,
dadurch gekennzeichnet,
daß die Sprachverarbeitungsvorrichtung zur Verarbeitung von drei Sprachsignalen (x1(i), x2(i), x3(i)) vorgesehen ist. - Mobilfunkendgerät nach einem der Ansprüche 2 bis 5,
dadurch gekennzeichnet,
daß zur Ermittlung eines Verzögerungsschätzwertes (T1'(i), T3'(i)) für das weitere Sprachsignal (x1(i), x3(i)) die Verwendung einer Linearkombination von Fehlerwerten (e12(i) mit e13(i), e31(i) mit e32(i)) vorgesehen ist. - Mobilfunkendgerät nach einem der Ansprüche 2 bis 6,
dadurch gekennzeichnet,
daß Verzögerungsmittel (16,27) zur Verzögerung des ersten Sprachsignals (x2(i)) mit einer festen Verzögerungszeit (Tmax) vorgesehen sind. - Mobilfunkendgerät nach einem der Ansprüche 2 bis 7,
dadurch gekennzeichnet,
daß die Sprachverarbeitungsvorrichtung in eine Freisprecheinrichtung (M1, M2, M3, 51, 52, 53) integriert ist.
Applications Claiming Priority (2)
Application Number | Priority Date | Filing Date | Title |
---|---|---|---|
DE4421853 | 1994-06-22 | ||
DE4421853A DE4421853A1 (de) | 1994-06-22 | 1994-06-22 | Mobilfunkendgerät |
Publications (3)
Publication Number | Publication Date |
---|---|
EP0689191A2 EP0689191A2 (de) | 1995-12-27 |
EP0689191A3 EP0689191A3 (de) | 1997-05-28 |
EP0689191B1 true EP0689191B1 (de) | 2001-05-23 |
Family
ID=6521236
Family Applications (1)
Application Number | Title | Priority Date | Filing Date |
---|---|---|---|
EP95201578A Expired - Lifetime EP0689191B1 (de) | 1994-06-22 | 1995-06-14 | Sprachverarbeitungsvorrichtung und Mobilfunkendgerät |
Country Status (4)
Country | Link |
---|---|
US (1) | US5647006A (de) |
EP (1) | EP0689191B1 (de) |
JP (1) | JPH0818473A (de) |
DE (2) | DE4421853A1 (de) |
Families Citing this family (3)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
US6535609B1 (en) * | 1997-06-03 | 2003-03-18 | Lear Automotive Dearborn, Inc. | Cabin communication system |
DE60010457T2 (de) * | 2000-09-02 | 2006-03-02 | Nokia Corp. | Vorrichtung und Verfahren zur Verarbeitung eines Signales emittiert von einer Zielsignalquelle in einer geräuschvollen Umgebung |
JP5931108B2 (ja) * | 2014-03-20 | 2016-06-08 | 本田技研工業株式会社 | ナビゲーションサーバ及びプログラム |
Family Cites Families (12)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
US3997772A (en) * | 1975-09-05 | 1976-12-14 | Bell Telephone Laboratories, Incorporated | Digital phase shifter |
DE3173306D1 (en) * | 1981-09-08 | 1986-02-06 | Ibm | Data receiving apparatus with listener echo canceller |
ES2145737T5 (es) * | 1989-09-01 | 2007-03-01 | Motorola, Inc. | Codificador digital de voz con predictor a largo plazo mejorado por resolucion de submuestreos. |
US5126681A (en) * | 1989-10-16 | 1992-06-30 | Noise Cancellation Technologies, Inc. | In-wire selective active cancellation system |
WO1992020170A1 (fr) * | 1991-04-30 | 1992-11-12 | Kabushiki Kaisha Toshiba | Appareil de radiotelephonie avec eliminateur d'echo |
EP0517525A3 (en) * | 1991-06-06 | 1993-12-08 | Matsushita Electric Ind Co Ltd | Noise suppressor |
US5519637A (en) * | 1993-08-20 | 1996-05-21 | Mcdonnell Douglas Corporation | Wavenumber-adaptive control of sound radiation from structures using a `virtual` microphone array method |
US5359663A (en) * | 1993-09-02 | 1994-10-25 | The United States Of America As Represented By The Secretary Of The Navy | Method and system for suppressing noise induced in a fluid medium by a body moving therethrough |
US5473701A (en) * | 1993-11-05 | 1995-12-05 | At&T Corp. | Adaptive microphone array |
NL9302013A (nl) * | 1993-11-19 | 1995-06-16 | Tno | Systeem voor snelle convergentie van een adaptief filter bij het genereren van een tijdvariant signaal ter opheffing van een primair signaal. |
US5581495A (en) * | 1994-09-23 | 1996-12-03 | United States Of America | Adaptive signal processing array with unconstrained pole-zero rejection of coherent and non-coherent interfering signals |
US5526426A (en) * | 1994-11-08 | 1996-06-11 | Signalworks | System and method for an efficiently constrained frequency-domain adaptive filter |
-
1994
- 1994-06-22 DE DE4421853A patent/DE4421853A1/de not_active Withdrawn
-
1995
- 1995-06-14 EP EP95201578A patent/EP0689191B1/de not_active Expired - Lifetime
- 1995-06-14 DE DE59509271T patent/DE59509271D1/de not_active Expired - Fee Related
- 1995-06-22 JP JP7156504A patent/JPH0818473A/ja not_active Ceased
- 1995-06-22 US US08/493,401 patent/US5647006A/en not_active Expired - Fee Related
Also Published As
Publication number | Publication date |
---|---|
EP0689191A3 (de) | 1997-05-28 |
EP0689191A2 (de) | 1995-12-27 |
DE59509271D1 (de) | 2001-06-28 |
JPH0818473A (ja) | 1996-01-19 |
DE4421853A1 (de) | 1996-01-04 |
US5647006A (en) | 1997-07-08 |
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