EP1103956A2 - Exponential reduction of echo and noise during speech pauses - Google Patents

Abstract

The exponential echo (E) and noise (N) reduction system samples the received signal regularly and has a speech pause detector (SPD) to scale down the cancellation constant (a0(k)) used to weight the stored best estimate of noise power spectral density (g(S/N)) during pauses.

Description

die Abtastwerte durchzählt und T die Periodendauer von einem Abtastwert zum nächsten bedeutet. The invention relates to a method for reducing echo and / or noise signals in telecommunications (= TC) systems for the transmission of acoustic useful signals, in particular human speech, in which it is determined by means of speech pause detection when in the mixture to be transmitted a speech signal is included from useful signals and interference signals or when there is a speech pause, the useful signals in their amplitude, which are usually disturbed by echo and / or noise signals, being amplified by a time-dependent control signal a ₀ (t) or by a multiplier with two inputs in the rhythm of a sampling rate f _T = 1 / T clocked control signal a ₀ (k) are changed, where k ∈

the samples are counted and T means the period from one sample to the next.

Such a method is known for example from DE 42 29 912 A1.

During a natural communication between people you fit in usually the amplitude of the spoken language automatically to the acoustic environment. With voice communication between distant In places, however, the conversation partners are not in the same acoustic Environment and are therefore not aware of the acoustic situation at the location of the aware of other interlocutors. This is why it occurs particularly sharply Problem when one of the partners due to its acoustic environment is forced to speak very loudly while the other partner in one quiet acoustic environment generates voice signals with low amplitude.

Added to this is the problem that an "electronic generated "noise arises and as a background to the useful signal is also transmitted. Furthermore, it is also advantageous to produce interference signals such as unwanted background noise (street noise, factory noise, office noise, To reduce or completely suppress canteen noise, aircraft noise etc.). To the To increase comfort when making a call, one is generally anxious to do any kind of To keep noise as low as possible.

Finally, so-called echoes also arise in TC connections in two-wire telecommunications networks as line echoes and, for example, in simple and more uncomfortable telecommunications terminals occur in the form of acoustic echoes.

It is therefore general when transferring from a mixture Voice signal and interference signals via telecommunications networks important, the interference signals like Reduce the amplitude of noise and echo as much as possible.

A known method for noise reduction is the so-called "spectral Subtraction ", for example in the publication" A new approach to noise reduction based on auditory masking effects "by S. Gustafsson and P. Jax, ITG conference, Dresden, 1998. It is about a spectral noise reduction method in which an acoustic Masking threshold (e.g. according to the MPEG standard) is taken into account. The disadvantage of such methods is that they are complex Determination of this acoustic masking threshold and the execution of all arithmetic operations associated with this method.

In the case of spectral subtraction, the noise is initially in the Speech breaks measured and in the form of a power density spectrum continuously stored in a memory. The power density spectrum is about won a Fourier transformation. When language occurs then the stored noise spectrum "as the best current estimate" subtracted from the current disturbed language spectrum, then into the Time domain transformed back to reduce noise in this way to get for the disturbed signal.

Another disadvantage of spectral subtraction is that Process of a generally inaccurate spectral noise estimate and Subsequent subtraction also errors in the output signal occur make it noticeable as "musical tones". In addition, this is well known Hardly any method for suppressing echo signals in telecommunications connections suitable.

In the extended spectral signal processing, which is also described in the citation mentioned, the power density spectra for the noise and for the speech itself are first estimated with the aid of a spectral subtraction. From the knowledge of these partial spectra, a spectral acoustic masking threshold R _T (f) for the human ear is then calculated using, for example, the rules from the MPEG standard. With the aid of this masking threshold and the estimated spectra for noise and speech, a filter pass curve H (f) is then calculated according to a simple rule, which is designed in such a way that essential spectral parts of the speech are passed through as unchanged as possible and spectral parts of the noise are reduced as much as possible.

Then the original disturbed speech signal is only through this filter given in this way a noise reduction for the disturbed signal to obtain. The advantage of this method is that from disturbed signal "nothing added or subtracted" and therefore errors in the Estimates are less or hardly noticeable. The disadvantage is again considerable computing effort for the spectral noise suppression as well adaptive filters to be connected upstream for echo suppression.

In the known compander method, as it is for example in the DE 42 29 912 A1 cited at the beginning, the degree of Noise and echo reduction according to a predefined Transfer function set, which also includes a level reduction very small input signals.

The compander initially has the property of speech signals with a certain (pre-set) "normal speech signal level" (possibly normal Called volume) practically unchanged from its entrance to the exit transferred to.

If the input signal becomes too loud, e.g. because a speaker too comes close to its microphone, a dynamic compressor limits it Output level to almost the same value as normal by using the current gain in the compander linearly with increasing input volume is lowered. This property keeps the language at the exit of the Compander systems about the same loud - regardless of how strong the Input volume fluctuates.

On the other hand, a signal with a level that is lower than that Normal level is given to the input of the compander, so it will be Signal attenuated by reducing the gain to To reduce background noise, if possible, only attenuated.

The compander thus consists of a compressor for speech signal levels, which are greater than or equal to a normal level and an expander for Signal levels that are lower than the normal level. The gain reduction the expander becomes stronger with increasingly smaller input levels.

The disadvantage of the compander solution is the considerable computing effort that is necessary to carry out the known method. Through the Compression of the speech signal level on the one hand and through the expansion on the other hand there is also a modulation in the speech volume that changes the speech signal in such a way that the Subjectively, the result is often perceived as unsatisfactory, i.e. one leaves an unsatisfactory listening impression.

In contrast, the object of the present invention is to provide a method with to introduce the features described at the outset, in which as possible uncomplicated and inexpensive way without large Computing effort and with little need for computing memory and Data storage space causes an echo and noise reduction, which with simple means one as pleasant as possible for the human ear overall acoustic impression, which also depends on the taste individual needs can be adjusted.

According to the invention, this object is achieved in a simple and effective manner in that the control signal a ₀ (t) or a ₀ (k) is varied in such a way that during the presence of speech signals in the useful signal, the amplitude of the control signal a ₀ (t ) or a ₀ (k) is set to a predetermined constant gain value c ₀ and with the beginning of a speech pause in the useful signal, the amplitude of the control signal a ₀ (t) or a ₀ (k) from one sample value to the next according to the recursion formula a 0 (k + 1) = a 0 (k) · β with β <1 is steadily lowered
and that after the end of a speech pause, a ₀ (k) = c _{0 is} set again.

This provides a very simple and very inexpensive method that also gives surprisingly good quality in terms of interference reduction, by preferring the interfering echo and Noise signals reduced. During the language phases themselves Noise at least partially masked and therefore from the human ear perceived far less clearly. By omitting compression the original speech signal is produced using the known compander method changed significantly less, so the result is usually a better one sounding voice signal arrives at the other end of the line. Moreover the method according to the invention requires less computing power than the compander process, since at least the compression is omitted. Correspondingly, there are smaller capacities in data storage and Computing required what the inventive method in In contrast to the known methods easier and cheaper designed.

In order to achieve effective noise reduction, the one to be transmitted Signal during the speech pauses in its performance value according to one temporal exponential function, in contrast to one from the input level dependent lowering, as in the compander process. In order to a substantial reduction in noise is already achieved. On top of that a reduction in noise during a pause in hearing significantly less burdened by the deafness effect after louder Sound exposure significantly reduced. The ear can be reinserted React language more sensitively and listen more carefully.

The factor β is advantageously chosen such that the constant decrease in time corresponds approximately to a time constant τ _{1 of} the perceptibility of the human ear. This means that after a strong sound event, the human ear does not perceive new sound events in time and in their amplitude below a curve shape that decays with the time constant τ ₁ after the end of the strong sound event. A variant of the method according to the invention is therefore preferred, in which the factor β is determined from the sampling rate f _T , from a time constant τ ₁ and from a predetermined constant pre-factor c ₁ according to the relationship β = c ₁ · exp (-1 / τ ₁ f _T )

The value of the time constant τ ₁ is usually between 50 ms to 150 ms in humans and is preferably about 65 ms.

In order to dimension the factor β exactly according to the time constant τ ₁ , it is expedient if c ₀ = 1 is selected.

If the constant exponential reduction of the interference signals is not limited according to the recursion formula described above, the value of a ₀ (k) very quickly becomes very small with increasing k and goes towards 0. However, this is not always desirable, since in many cases one prefers want to hear a little residual noise to avoid the impression during a pause in speech that the telecommunications line is suddenly "dead" or interrupted. A variant of the method according to the invention is therefore preferred, in which a ₀ (k + 1) assumes a predetermined constant value c ₂ during a speech pause and / or a value c ₃ <c ₂ if an echo signal is present if the previous value a ₀ (k ) Has become ≤ c ₂ .

It is also desirable to consider the level of signal level reduction in the Adapt speech pauses to the current situation in the TK channel.

For example, noise is preferred depending on instantaneous noise level N or depending on a function g (S / N) lower the signal-to-noise ratio S / N, but short-term ones Lower echoes more and after the end of the echoes the lowering to the reduce the lower value of the noise reduction.

A method variant is therefore particularly preferred, which is characterized in that, in the presence of a noise signal and / or echo signal and for a ₀ (k) c c ₂ , where c _{2 is} a predetermined constant, the power value of the noise level N in the currently used TC Channel is continuously measured and / or estimated, and that, depending on the current noise level N, the control signal a ₀ (k + 1) is continuously set according to a ₀ (k + 1) = f (N), where f (N) is a predetermined one Is a function of N.

The degree of noise reduction is thus different from that currently occurring Power value N of the noise automatically controlled and the current Noise level in the telephone channel adjusted and in a predefined defined Wisely tracked. The subjective function can also be selected via the function f (N) Impression of the total signal generated can be adjusted. Another advantage This variant of the method consists in that with a bundle of Telephone channels, for example between international exchanges, the noise situation in each individual channel, which is very different from channel to channel can be different, automatically set and individually optimized can.

A variant of the method according to the invention is particularly preferred, the is characterized in that the predetermined function f (N) is a function g (S / N), which is the quotient S / N from the power value of the signal level S the useful signals to be transmitted and the power level of the noise level N depends, or that the given function f (N) is a function g '(N / S), which depends on the reciprocal N / S of this quotient. For the sake of one simpler practical implementation can also be a function of Use (S + N) / N or from (S + N) / S.

The advantage of the above process variant is that with strong Varying useful signal level S in the telephone channels of a bundle always the correct setting for noise reduction is found. At a The control of the noise reduction proportional to the reciprocal N / S can be done Function g '(N / S) easily on a digital signal processor (= DSP) with fixed Computer word lengths of, for example, 16 bit using Implement particularly simple software, because for N / S preferably a Numerical range 0 <N / S <1 relevant for controlling noise reduction or interesting is.

Acoustic hearing tests have shown that at S / N = 0 db the speech is already so disturbed that the noise may only be reduced by a value f ₀ or g ₀ between 5 and 10 dB, preferably between 6 and 8 dB in order not to worsen the overall acoustic impression with regard to the naturalness of the language. With even more unfavorable values of the signal-to-noise ratio S / N <0 dB, the value f ₀ or g ₀ can then be retained, since any further noise reduction only worsens the overall impression.

With medium S / N, a stronger noise reduction can be carried out according to these studies. A maximum results in the range of 10 to 15 dB. The _maximum value of the noise reduction f _max or g _max should be between 20 and 30, preferably about 25 dB.

With very good noise values S / N> 40 dB, only a minimal one should be left Reduction between 0 and 3 dB can be set to the naturalness of the to get transmitted language as good as possible.

The sound of the language and the intelligibility are particularly good when the Function f (N) or g (S / N) across the three areas discussed above in is interrelated, with rapid changes in N or in S (N) can advantageously be smoothed by filtering.

A relatively simple implementation in hardware and / or software results by the functions f (N) or g (S / N) or g '(N / S) by straight Characteristic pieces between the three operating points described above approximated (linear approximation in sections).

In a somewhat more complex variant of the method according to the invention, but the result is a better sound, becomes a Polynomial function to implement the continuous functions f (N) or g (S / N) or g '(N / S) in the three areas discussed, which is the result leads to a kind of asymmetrical bell function.

A variant of the method according to the invention is particularly preferred for functions f (N) or g (S / N) or g '(N / S) are selected so that the Reduction of the noise level N according to the psychoacoustic Averages of the human hearing spectrum are made. The value for S and / or N not only from the current performance value alone, but also determined from a weighted spectral profile of S or N and Overall, an aurally correct, i.e. a achieved a pleasant sounding noise reduction that is psychoacoustically pleasing. Since there is no easy-to-display dimension for an acoustically pleasant sounding Noise reduction there, all quality assessments are extensive Hearing tests instructed, which are then optimized using statistical Methods are evaluated to an evaluation scale, (similar to Voice codecs).

A good noise level estimate requires a good speech pause detector, because you can only be sure that in the speech pause sections just annoying noise and not any mix between noise and scraps of speech, as is often the case in practice occurs.

A method variant which is characterized thereby is particularly preferred distinguishes that in the speech pause detector from the input signal x a short-term output signal sam (x) using a short-term level estimator, a middle time output signal mam (x) by means of a middle time level estimator and by means of a long-term level estimator a long-term output signal lam (x) is formed that the three output signals sam (x), mam (x) and lam (x) over suitable gain coefficients are set so that they are approximately equal large if the input signal x is a pure noise signal, where sam (x) < mam (x) <lam (x) that the three output signals sam (x), mam (x) and lam (x) are monitored by comparators and that the existence of a Speech signal is assumed as the input signal x if sam (x) and mam (x) initially become larger than lam (x), and the existence of a Speech pause when sam (x) and / or mam (x) then becomes smaller than lam (x).

With the help of these relatively simple types of formation of various Average values of the time signal can already be a surprisingly good one Speech pause detection can be carried out, which is only a very small one Computational effort required.

A further development of this method variant provides that the three output signals sam (x), mam (x) and lam (x) for speech pause estimation a neural network can be supplied with a variety was trained by scenarios with different input signals x. On Neural network can advantageously linear and non-linear relationships between a large amount of input parameters and the desired ones Show output values. A prerequisite for this is that the neural Network with a sufficient amount of input values and associated output values were trained. Therefore neural are suitable Networks especially for the task of speech pause detection Presence of different disturbing noises.

In addition to the detection and reduction of noise signals, the presence of echo signals is preferably also detected and / or predicted and the corresponding echo signals are suppressed or reduced. If, in addition to noise, echoes also occur in a telephone channel, these can generally be predicted on the basis of a previously determined signal propagation time τ _{E of} an echo as well as the previously determined echo coupling ERL in the channel and the signal strength ES which triggers the echo in the return channel. This estimation can be carried out in such a way that the size of the delayed arriving echoes is estimated as a function of the transmitted speech signal and its instantaneous power. If the respectively estimated echo signal exceeds a predetermined threshold value thrs in certain short time segments, then this echo-laden signal is preferably briefly attenuated, for example by the exponential reduction mentioned above, to a value which is necessary for a substantial reduction in the echo signal. In the same sense, a compander characteristic curve can also be briefly shifted in the direction of greater input volume in the case of echoes and returned to its original position after the echoes have subsided.

A development of this method variant is particularly preferred, in which the control signal a ₀ (k + 1) is continuously set according to a ₀ (k + 1) = h (N, S, ES, τ _E , ERL), where h (N, S, ES, τ _E , ERL) is a predetermined function of N, S, the useful signal ES in the opposite direction of a speaking TC partner, τ _{E is} a constant delay time of the echo signal and ERL is a damping constant of the amplitude of the echo signal.

You can advantageously use an ear-sound reduction connect independent echo reduction. It is special important if there is almost no background noise in the telephone channel exists because no noise reduction is effective, and thus occurring echoes can reach the speaker unhindered.

It is advisable to separate the control of a noise reduction from that of an echo reduction, since noises and echoes occur independently of one another and generally also have completely different physical causes. But one can mathematically state a general reduction function R that describes a reduction in signal levels for both noise and echoes: R (S, N, ES, τ E , ERL, thrs) ∼ g (S / N) · d (ES, τ E , ERL, thrs), where g (S / N) means the noise reduction described above and d (...) the independent additional echo reduction when the estimated echo signal exceeds the predetermined threshold thrs.

A method variant is particularly advantageous in which during the Duration of an echo reduction in addition to the useful signal is an artificial one Noise signal is added.

A noise reduction is also with the same noise level constant. A sudden additional echo reduction in the rhythm of the Language also means a reduction in noise (at least in the short Period) in the rhythm of speech. This leads to a pulsed Background noise, which does not sound natural. Therefore, it is advantageous in the moments of an additional echo reduction synthetic noise of a suitable noise generator in the Magnitude of normal background noise to the processed signal to add. This is to ensure that the background noise remains as constant as possible be conveyed to the listener.

The noise generator can be designed so that the artificial Sound signal a psycho-acoustically perceived as pleasant acoustic Signal sequence (= comfort noise) includes.

Instead of a synthetic background noise, a Section of a previously recorded real background noise in appropriate strength can be inserted in the echo periods. The added noise then hardly differs from that at all previous noise and therefore will not be an annoying acoustic Cause a change in the listener.

The addition of sounds to acoustically mask effects and the measures of separate treatment of noise and Echoes, when properly coordinated, make you special understandable and pleasant language impression even with "difficult" Effect environment (echoes plus noise).

A variant of the invention is also particularly preferred Method in which the useful signal to be transmitted is spectral Subtracted. The advantage of using spectral subtraction downstream level reduction in the speech pauses is that first by means of spectral subtraction some of the noise from the Speech signal itself is eliminated and only then the speech pauses in the described type of noise and echoes are freed. Total results this combination gives better auditory impressions than subjective tests simple spectral subtraction.

Another particularly advantageous variant of the invention Finally, the method provides that the useful signal to be transmitted is a spectral filtering adapted to human hearing. Here too, spectral subtraction is used to create one Estimation of noise, speech and echoes performed, then determines an audible masking threshold and then the whole Signal processed via an appropriately set transmission filter so that the Speech parts as pristine as possible and the echo and noise parts be suppressed as much as possible.

A combination with the downstream level reduction in the Language breaks further improve the listening experience.

A server unit also falls within the scope of the present invention Supporting the inventive method described above and a computer program for performing the method. The procedure can both as a hardware circuit and in the form of a computer program will be realized. Nowadays software programming for powerful DSP's preferred because of new knowledge and additional functions easier by changing the software on an existing hardware basis are implementable. Processes can also be used as hardware modules for example, implemented in telecommunications terminals or telephone systems.

Further advantages of the invention result from the description and the Drawing. Likewise, the above and the others Features listed according to the invention individually for themselves or to several can be used in any combination. The shown and The described embodiments are not to be considered as a conclusive list understand, but rather have exemplary character for the description the invention.

Fig. 1

das Steuersignal ao bei Vorliegen von Sprachsignalen, während einer Sprachpause und bei erneutem Einsetzen der Sprachsignale;

Fig. 2

ein Schema einer Anordnung zur gesteuerten Signalabsenkung;

Fig. 3a

die Funktion g(S/N) in der linearen Näherung;

Fig. 3b

die entsprechende Funktion g'(N/S);

Fig. 4a

die Funktion g(S/N) als unsymmetrische Glockenkurve; und

Fig. 4b

die entsprechende Funktion g'(N/S).

The invention is illustrated in the drawing and is explained in more detail using exemplary embodiments. Show it:

Fig. 1

the control signal ao when speech signals are present, during a speech pause and when the speech signals are used again;

Fig. 2

a diagram of an arrangement for controlled signal reduction;

Fig. 3a

the function g (S / N) in the linear approximation;

Fig. 3b

the corresponding function g '(N / S);

Fig. 4a

the function g (S / N) as an asymmetrical bell curve; and

Fig. 4b

the corresponding function g '(N / S).

The control signal a ₀ shown in FIG. 1 as a function of the time t or the number of samples k is kept at a value c ₀ = 1 during a first phase T1, in which speech signals are detected. During a speech pause in the time period T2, the control signal a _{0 is} lowered exponentially to a constant value c ₂ , which is just above 0, and then jumps back to the value c ₀ = 1 (or another, arbitrary) when the speech signals are used again during a phase T3 selectable constant). As a result, no (or, in other examples, only a slight) suppression of interference signals in the overall signal is carried out during the speech phases T1, T3, so that the speech signal is passed on as unadulterated and unimpeded as possible. During the speech pause in phase T2, echo and noise signals are suppressed as effectively as possible (exponentially), but in the present example it is not reduced to the value 0, but to a small residual value c ₂ , so as not to cause the other end To give the impression of a "dead" line. If echoes occur, they are reduced to a residual value c ₃ <c ₂ .

2 schematically shows the functioning of an arrangement for noise and echo reduction in accordance with the above-mentioned reduction function R (S, N, ES, τ _E , ERL, thrs) with a speech pause detector SPD.

It applies to all the curves shown in FIGS. 3a to 4b that the function value g or g 'for the case S / N <0 dB, that is to say with an extremely high noise background, changes to a constant value g _{0 of} the noise reduction of approximately 6 dB. Starting from S / N = 0 dB, with increasing improvement in the signal-to-noise ratio S / N, an increased noise reduction is achieved, which reaches a maximum g _max ≈ 25 dB at approximately S / N ≈ 12 dB. As the S / N continues to increase, the degree of noise reduction finally drops to zero in order to carry out as little manipulation as possible in the transmitted useful signal with low background noise.

Claims (23)

Method for reducing echo and / or noise signals in telecommunications (= TC) systems for the transmission of acoustic useful signals, in particular human speech, in which it is determined by means of speech pause detection when in the mixture of useful signals and interference signals to be transmitted a speech signal is included or when there is a speech pause, with the aid of a multiplier with two inputs, the amplitude of the useful signals, which are generally disturbed by echo and / or noise signals, by a time-dependent control signal a ₀ (t) or by a rhythmic sampling rate f _T = 1 / T clocked control signal a ₀ (k) can be changed, where k ∈

the samples are counted and T means the period from one sample to the next,
characterized,
that the control signal a ₀ (t) or a ₀ (k) is varied such that during the presence of speech signals in the useful signal, the amplitude of the control signal a ₀ (t) or a ₀ (k) to a predetermined constant value c ₀ is set and with the beginning of a speech pause in the useful signal, the amplitude of the control signal a ₀ (t) or a ₀ (k) from one sample value to the next according to the recursion formula a 0 (k + 1) = a 0 (k) · β with β <1 is steadily lowered
and that after the end of a speech pause a ₀ (k) = c _{0 is} set. Method according to claim 1, characterized in that the factor β is determined from the sampling rate f _T , from a time constant τ ₁ and from a predetermined constant pre-factor c ₁ according to the relationship β = c 1 Exp (-1 / τ 1 f T ). Method according to claim 2, characterized in that the time constant τ _{1 is} chosen between 50 ms and 150 ms, preferably τ ₁ ≈ 65 ms. Method according to one of the preceding claims, characterized in that the constant value c ₀ = 1 is selected. Method according to one of the preceding claims, characterized in that a ₀ (k + 1) assumes a predetermined constant value c ₂ during a pause in speech and / or the presence of an echo signal if the previous value has become a ₀ (k) ≤ c ₂ . Method according to one of claims 1 to 4, characterized in that during a pause in speech and / or the presence of an echo signal and for a ₀ (k) ≤ c ₂ , where c _{2 is} a predetermined constant, the power value of the noise level N currently used TC channel is continuously measured and / or estimated, and that depending on the current noise level N, the control signal a ₀ (k + 1) is continuously set according to a ₀ (k + 1) = f (N), where f (N) is a given function of N. A method according to claim 6, characterized in that the predetermined function f (N) is a function g (S / N), which is from Quotient S / N from the power value of the signal level S to transmitting useful signals and the power value of the Noise level depends on N, or that the given function f (N) is a function g '(N / S) which has the reciprocal N / S of this Quotient depends. A method according to claim 7, characterized in that the function f (N) or g (S / N) at 1 / N «1 or S / N = 0 dB with a constant value f ₀ > 0 or g ₀ > 0 begins, in the range between N or S / N = 10 dB to 15 dB, preferably for N or S / N ≈ 12 dB, increases to a maximum f _max or g _max and then to a minimum value f _min or g _min , preferably drops to 0 dB, where 5 dB ≤ f ₀ , g ₀ ≤ 10dB, preferably 6dB ≤ f ₀ , g ₀ ≤ 8dB, and where 20 dB ≤ f _max , g _max ≤ 30 dB, preferably f _max , g _max ≈ 25 dB. Method according to one of claims 6 to 8, characterized characterized in that the function f (N) or g (S / N) at least piece by piece, preferably linear in all subsections with N or S / N runs. Method according to one of claims 6 to 8, characterized characterized that the function f (N) or g (S / N) from polynomials is constructed and as an asymmetrical bell curve over N or S / N runs. Method according to one of claims 6 to 10, characterized characterized in that the functions f (N) or g (S / N) or g '(N / S) be chosen so that the reduction of the noise level N accurate according to the psychoacoustic mean values of the human hearing spectrum. Method according to one of the preceding claims, characterized characterized that in addition to the detection and reduction of Noise signals detected the presence of echo signals and / or is predicted and the echo signals are suppressed or be reduced. Method according to claim 12 and one of claims 6 to 11, characterized in that the control signal a ₀ (k + 1) is continuously set according to a ₀ (k + 1) = h (N, S, ES, τ _E , ERL) , where h (N, S, ES, τ _E , ERL) is a predetermined function of N, S, the useful signal ES in the opposite direction of a speaking TC partner, τ _{E is} a constant delay time of the echo signal and ERL is a damping constant of the amplitude of the echo signal . A method according to claim 12, characterized in that the Control of noise signal reduction and reduction separated from echo signals. Method according to one of claims 12 to 14, characterized characterized that during the period of an echo reduction an artificial noise signal is added to the useful signal becomes. A method according to claim 15, characterized in that the artificial noise signal a psychoacoustic as pleasant perceived acoustic signal sequence (= comfort noise). A method according to claim 15, characterized in that the artificial noise signal previously during the current telecommunications connection recorded sound signal includes. Method according to one of the preceding claims, characterized in that that in a speech pause detector (SPD) a short-term output signal sam (x) from the input signal x by means of a short-term level estimator, a medium-term output signal mam (x) by means of a medium-term level estimator and a long-term by means of a long-term level estimator -Output signal lam (x) is formed such that the three output signals sam (x), mam (x) and lam (x) are set via suitable amplification coefficients so that they are approximately the same size when the input signal x is a pure noise signal, whereby sam (x) <mam (x) <lam (x), that the three output signals sam (x), mam (x) and lam (x) are monitored by comparators, and that the presence of a speech signal is assumed to be the input signal x if sam (x) and mam (x) initially become larger than lam (x), and the presence of a speech pause if sam (x) and / or mam (x ) becomes smaller than lam (x) again. A method according to claim 18, characterized in that the three output signals sam (x), mam (x) and lam (x) for speech pause estimation be fed to a neural network that with a variety of scenarios with different Input signals x was trained. Method according to one of the preceding claims, characterized characterized in that the useful signal to be transmitted is a undergoes spectral subtraction. Method according to one of the preceding claims, characterized characterized in that the useful signal to be transmitted is one of the subjected to spectral filtering adapted to human hearing becomes. Server unit to support the method according to one of the Claims 1 to 21. Computer program for performing the method according to a of claims 1 to 21.

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