EP1755110A2 - Procédé et dispositif destinés à la réduction adaptative de signaux de bruit et de fond dans un système de traitement vocal - Google Patents

Procédé et dispositif destinés à la réduction adaptative de signaux de bruit et de fond dans un système de traitement vocal Download PDF

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Publication number
EP1755110A2
EP1755110A2 EP06014433A EP06014433A EP1755110A2 EP 1755110 A2 EP1755110 A2 EP 1755110A2 EP 06014433 A EP06014433 A EP 06014433A EP 06014433 A EP06014433 A EP 06014433A EP 1755110 A2 EP1755110 A2 EP 1755110A2
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Prior art keywords
prediction
filter
coefficients
input signal
audio input
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EP06014433A
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German (de)
English (en)
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EP1755110A3 (fr
Inventor
Jörn Fischer
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Entropic Communications LLC
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TDK Micronas GmbH
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Publication of EP1755110A2 publication Critical patent/EP1755110A2/fr
Publication of EP1755110A3 publication Critical patent/EP1755110A3/fr
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    • 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
    • G10L21/00—Speech or voice signal processing techniques 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
    • 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
    • G10L21/00—Speech or voice signal processing techniques 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

Definitions

  • the invention relates to a method for reducing noise and background signals in a speech processing system with the preamble features of claim 1 and to an apparatus for performing such a method with the preamble features of claim 18.
  • noise such as noise and non-speech background noise reduce the quality of speech processing, e.g. in terms of the detection or compression of the speech or speech signal components contained in an input signal.
  • noise e.g. in terms of the detection or compression of the speech or speech signal components contained in an input signal.
  • filter devices For reducing noise and background signals in speech processing systems, filter devices are employed which include at least one audio input, an audio output, a memory and a processor or a field programmable device or an application-specific integrated circuit (ASIC) Perform filtering procedure.
  • ASIC application-specific integrated circuit
  • FIR filter Finite Impulse Response / Finite impulse response
  • LPC Linear Predictive coding / linear prediction coding
  • the object of the invention is to improve a method for the reduction of noise and background signals in a voice-processing system or a device for carrying out such a method in terms of applicability, in particular to make it more flexible.
  • a method of reducing noise and background signals in a voice processing system in which an audio input signal is filtered by means of filtering using an adaptive filter to generate a predicted output signal with reduced noise, wherein the filtering is performed using is accordingly preferred a plurality of coefficients for forming a plurality of prediction errors and for forming one of the plurality of prediction errors, wherein by means of a plurality of reduction parameters, the amounts of the coefficients are continuously reduced.
  • a method is preferred in which the continuous reduction of the coefficients is produced by multiplying the coefficients by a factor smaller than 1, in particular by multiplying by a factor between 0.8 and 1.0.
  • a method is preferred in which the predicted output signal is used as a prediction of the reduced noise audio input signal as an input to a subsequent second filtering to produce a second prediction.
  • the second filtering is carried out by means of predictive filtering with a second, known per se filtering with a set of second coefficients, wherein a learning rate for adjusting the coefficients by a few powers of ten is chosen to be less than a learning rate of first filtering.
  • a method is preferred in which the second prediction is then subtracted from the prediction output signal in order to eliminate long-lasting background noise.
  • a method is preferred in which a learning rule for determining the further coefficients is asymmetrical, so that the magnitude of the further coefficients falls more sharply in magnitude than can rise and drop rapidly to zero, but increases only with a small slope.
  • a method is preferred in which, instead of the audio input signal for determining individual prediction errors, only its sign is used so as not to disadvantage small signals.
  • Particularly preferred is a method in which the coefficients are limited to avoid drifting of the coefficients, in particular to a range of e.g. -4 ... 4, if the audio input signal is standardized from -1 ... 1.
  • a method is preferred in which a maximum of a speech signal component of the audio input signal is detected and the output signal is normalized, in particular sluggishly, to this maximum.
  • a method is preferred in which the output signal of the first and / or the second filtering in relation to their input signal is used in particular simultaneously as a measure of the presence of speech in the input signal.
  • a method is preferred in which a filter is used for the first and / or the second filtering, which performs an error prediction by means of an LMS adaptation (Least Mean Squares Adaption).
  • LMS adaptation Least Mean Squares Adaption
  • an FIR filter is used for the first and / or the second filtering.
  • a method is preferred in which a sigmoid function is multiplied by the predicted output signal to avoid overdriving the signal in the case of a bad prediction.
  • Particularly preferred is a method in which the predicted output signal as the original signal, the audio input signal is mixed to produce a more natural sound.
  • a method is preferred in which a field-programmable component or an ASIC (Application-Specified-Integrated-Circuit) is correspondingly programmed for carrying out the method.
  • ASIC Application-Specified-Integrated-Circuit
  • an apparatus in particular apparatus for carrying out a method for reducing noise and background signals in a speech processing system, having an audio input for inputting an audio input signal to an adaptive filter for filtering audio input signal to produce a prediction output signal with reduced noise, with a memory for storing a plurality of coefficients for the filter, wherein the filter is configured or switched to form a plurality of prediction errors and to form an error of the plurality of prediction errors, wherein a coefficient providing arrangement is formed or switched by means of at least one reduction parameter to reduce the amounts of the coefficients continuously.
  • a device is preferred in which the coefficient providing arrangement for multiplying the coefficients by the reduction parameter as a factor less than 1, in particular with a factor between 0.8 and 1.0 is formed or connected.
  • a device in which a first filter stage with the filter as the first filter is followed by a second filter stage with a second filter for supplying the predicted output signal as a prediction of the audio input signal with reduced noise as an input signal for the second filter for generating a second prediction.
  • Particularly preferred is a device having a subtraction circuit for subtracting a sum of error predictions of the second filtering from the prediction output signal to produce the prediction.
  • a device is preferred in which the second filter is designed or switched by an LMS adaptation filter for carrying out an error prediction.
  • a device is preferred in which the first filter and / or the second filter is formed or switched by an FIR filter for performing a signal prediction.
  • a device which is formed by a field-programmable component or an ASIC is preferred.
  • a device is provided with a multiplier for weighting the optionally time delayed audio input signal or for weighting the prediction output signal with a weighting factor less than one, in particular about 0.1 and an adder for adding the weighted signal to the prediction output signal or the Prediction for generating a noise-reduced audio output signal.
  • the computational effort is much lower.
  • the computation outlay is O (n log (n)) and the computational cost of an autocorrelation is O (n 2 )
  • the computational effort of the preferred method of the entire method of both filter stages is only O (n), where n a number of sampled samples (nodes) of the input signal and O is a general function of the filter overhead.
  • a speech signal is only delayed by a single sample.
  • An adaptation is for noises instantaneously and for long-lasting background noises, the adaptation is preferably delayed about 0.2 s to 5.0 s.
  • the method is much less computationally expensive than conventional methods. Especially with only four coefficients one obtains respectable results, so that only four multiplications and four additions have to be calculated for the prediction of one sample and only four to five further operations for the adaptation of the filter coefficients are required.
  • the particularly preferred method consists of two adaptive filters F1, F2, which are connected in series as a first and a second filter stage.
  • Autonomously advantageous is already the use of only the first filter stage.
  • an audio input signal s (t) is input via an audio input 1.
  • the audio input signal is applied to a group of delay elements 2, which z. B. are formed as a buffer and delay the respective applied value of the audio input signal s (t) by one clock.
  • the audio input signal s (t) is supplied to a first adder 3.
  • the values s (t-1) -s (t-4) delayed by means of the delay elements 2 are output from the respective delay element 2 to the next of the delay elements 2 and respectively two corresponding multipliers of two groups of multipliers 4 created.
  • the group of second multipliers 5 is applied to a further multiplication input in each case a coefficient c1 - c4 as a filter coefficient of an adaptive filter.
  • the multiplication results of the group of second multipliers 5 are output to a second adder 6 as individual prediction errors sv1-sv4.
  • a time sequence of the addition values of the second adder 6 forms a prediction output sv (t).
  • the sequence of values of the predicted output signal sv (t) are directly output according to a first advantageous embodiment to form an output signal o (t) (FIG. 2).
  • the sequence of values of the predicted output sv (t) are also applied to the first adder 3 formed as a subtraction circuit at a subtraction input to subtract these values from the current later value of the audio input signal s (t).
  • the subtraction result of the first adder 3 forms an error e from a corresponding sequence of individual error values.
  • This error e is applied to a third multiplier 8, at the second multiplication input of which a value of a learning rate ⁇ with preferably ⁇ 0.01 is applied.
  • the multiplication result is applied to the inputs of the group of first multipliers 4 for multiplication by the delayed values s (t-1) -s (t-4).
  • the multiplication results of the group of first multipliers 4 are supplied to a group of third adders 10, which form an input of a coefficient providing arrangement 9.
  • the output values of the group of third adders 10 form the coefficients c1 - c3, which are applied to the corresponding multipliers 5 of the group of second multipliers 5.
  • these coefficients c1-c4 are each applied to an adder 11 of a group of fourth adders 11 and in each case to a multiplier 12 of a group of fourth multipliers 12.
  • a reduction parameter k is applied to a multiplication input, wherein the value of the reduction parameter k is, for example, 0.0001.
  • the respective value of the coefficients c1-c4 is correspondingly reduced by this factor.
  • the corresponding multiplication result of the fourth multipliers 12 is applied to the respective one of the fourth adder 11 formed as a subtraction circuit, to which the corresponding coefficient c1-c4 has previously been applied, at a subtraction input.
  • the output value of the respective adders 11 of the fourth group adder 11 is applied to another input of the corresponding third adder of the group of third adders 10.
  • the group of third adders 10 adds the respective addition value of the group of fourth adders 11 to the respectively applied and delayed audio signal input value s (t-1) -s (t-4) in order to learn the coefficients.
  • the prediction output signal sv (t) for forming the output signal o (t) can optionally be added with a weighted value formed directly from the instantaneous or optionally from a correspondingly delayed value of the audio input signal s (t).
  • the weighted value is provided by a weighting multiplier 15, which multiplies the input signal s (t) by a factor ⁇ ⁇ 1, in particular ⁇ ⁇ 0.1.
  • the predicted output sv (t) or the output o (t) is not output as a final output but provided as an input to a second filter stage with the second filter F2.
  • the second filter F2 is again an adaptive filter arrangement, the construction of which is preferably substantially equal to the structure of the first filter stage. Therefore, only differences to the first filter stage will be described below.
  • the respective components and signals or values are with a Star for distinguishing the corresponding components and signals or values of the first filter stage characterized.
  • the output result of the second filter F2 is correspondingly provided by a second adder 6 * of the second filter F2 and added to the input signal or the corresponding input value of the input signal sv (t) of the second filter F2 by means of a fifth adder 13 * or preferably subtracted therefrom Case of an adder 6 *, preferably designed as a subtraction circuit.
  • the output of the fifth adder 13 * forms a second prediction sv * (t) as a second prediction output.
  • the values of the prediction sv * (t) are added to the optionally time-delayed and weighted audio input signal s (t) or sv (t) by means of a sixth adder 14 * to produce a noise-reduced audio output signal o * (t).
  • the arrangement has further components in a conventional manner or is connected to further components such as a processor for control functions and a clock generator for providing a clock signal.
  • the arrangement has a memory or can access a memory.
  • the first filter F1 reduces the noise over the entire perceived frequency range.
  • a modified adaptive FIR filter is trained to predict the audio input signal s (t) containing eg speech and noise as well as possible from the past n values.
  • the output is the predicted values as the prediction output sv (t).
  • the amounts of the coefficients c i (t) are continuously reduced, resulting in smaller predicted amplitudes in noise signals than in speech signals. It is determined with the reduction parameter k, how much the noise should be suppressed.
  • the second filter F2 reduces long-lasting background noise. It exploits the fact that the energy of speech signal components in the audio input signal s (t) repeatedly falls to zero in individual frequency bands, whereas long-lasting tones tend to have a constant energy in the frequency band.
  • the prediction sv * (t) thus obtained in the second filter F2 is subtracted from the input signal s (t), so that the long-lasting signals from the input signal s (t) are eliminated or at least greatly reduced.
  • the first and second filters F1, F2 are particularly efficient when executed one after the other on the input signal s (t), as shown in FIG.
  • first the first filter F1 is executed and its output or predicted output signal sv (t) is passed as an input signal to the second filter F2 for further additional filtering.
  • FIG. 1 schematically shows an amplitude curve a over the time t of an exemplary input signal s (t) in the time domain before and after the filtering by the first filter F1 for noise suppression. While the input signal s (t) includes speech and noise, the prediction output sv (t) of the first filter F1 includes speech and a reduced noise.
  • Fig. 2 shows schematically an amplitude curve a over the time t of an exemplary input signal s (t) and the prediction output signal sv (t) in the frequency range before and after the filtering by the second filter F2 for the suppression of long-lasting background noise.
  • the x-axis corresponds to the time t
  • the y-axis corresponds to a frequency f
  • a brightness corresponds to an amplitude.
  • Recognizable is a spectrum of a prominent 2 kHz tone in the background in front of the second filter F2 compared to a spectrum with a reduced 2 kHz tone after the second filter F2.
  • the reduction of the coefficients c i (t) can alternatively or additionally also be produced by the coefficients c i (t) having a fixed or variable factor between in particular 0, 8 and 1.0 are multiplied.
  • a method and apparatus respectively, in which a sigmoidal function, e. G. (T), is applied after the first filter F1 has been used with its predicted output sv (t).
  • a hyperangular hyperbaric which avoids overdriving the signal in case of a bad prediction.
  • a method and a device are advantageous if the audio input signal (s (t)) is added to the prediction output signal (sv (t)) as the original signal in order to produce a more natural sound.
  • a plurality of reduction parameters for the different coefficients c1-c4 can also be determined or determined individually.
  • the reduction parameter or k can also be dependent on e.g. be varied according to the received audio input signal.

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  • Engineering & Computer Science (AREA)
  • Computational Linguistics (AREA)
  • Quality & Reliability (AREA)
  • Signal Processing (AREA)
  • Health & Medical Sciences (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • Human Computer Interaction (AREA)
  • Physics & Mathematics (AREA)
  • Acoustics & Sound (AREA)
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  • Compression, Expansion, Code Conversion, And Decoders (AREA)
  • Soundproofing, Sound Blocking, And Sound Damping (AREA)
EP06014433A 2005-08-19 2006-07-12 Procédé et dispositif destinés à la réduction adaptative de signaux de bruit et de fond dans un système de traitement vocal Withdrawn EP1755110A3 (fr)

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DE102005039621A DE102005039621A1 (de) 2005-08-19 2005-08-19 Verfahren und Vorrichtung zur adaptiven Reduktion von Rausch- und Hintergrundsignalen in einem sprachverarbeitenden System

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US8352256B2 (en) 2013-01-08
DE102005039621A1 (de) 2007-03-01
EP1755110A3 (fr) 2009-05-06
US20070043559A1 (en) 2007-02-22
US7822602B2 (en) 2010-10-26
US20110022382A1 (en) 2011-01-27

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