AUDIO SIGNAL COMPONENT COMPENSATION SYSTEM
A system compensating audio signal components in a communication system is disclosed, the method comprising the steps of detecting, by a microphone, a sound signal, the sound signal comprising audio signal components resulting from reproducing an audio signal of an audio source, the sound signal further comprising speech signal components corresponding to a speech signal from a person, filtering the sound signal to whiten the sound signal, compensating the audio signal components in the whitened sound signal, and removing the whitening of the compensated sound signal, where the filtering of the audio signal is performed using at least two filters in an alternating way, each filter using time-dependent filter coefficients.
Latest Harman Becker Automotive Systems GmbH Patents:
This application claims priority to European Patent Application Serial No. 06 014 366.6, filed on Jul. 11, 2006, titled METHOD FOR COMPENSATION OF AUDIO SIGNAL COMPONENTS IN A VEHICLE COMMUNICATION SYSTEM AND SYSTEM THEREFOR, the application of which is incorporated by reference in its entirety in this application.
BACKGROUND OF THE INVENTION1. Field of the Invention
This invention relates to a communication system. More particularly, the invention relates to a method and a system for compensation of audio signal components in a communication system, such as in a vehicle communication system.
2. Related Art
The use of various types of communication systems has been proliferating over the last few years. For example, communication systems are often incorporated in vehicles for different purposes. For example, it is possible to use speech recognition and voice commands of the driver for controlling predetermined electronic devices inside the vehicle. Additionally, telephone calls, such as in a conference call, are possible with two or more passengers within the vehicle. For example, a person sitting on a front seat and a person sitting on one of the back seats may talk to a third person on the other end of the line using a hands-free communication system inside the vehicle. Moreover, it is possible to use the communication system inside the vehicle for the communication of the different vehicle passengers to each other,
In vehicle communication systems, it may be difficult to hear speech audibly and clearly due to noise, other sounds in the vehicle or attenuation of the speech sound waves. Accordingly, the voice of one of the passengers may be detected using one or more microphones positioned in different locations in the vehicle. The signal detected by the microphone can be processed and then output using the loudspeakers of an audio module that is normally located in the vehicle. The signal emitted from the loudspeaker, however, is normally also detected by the microphone. To avoid acoustic feedback or other undesirable effects, the signals detected by the microphone have to be processed and such signal components have to be filtered out. Otherwise, undesirable feedback can occur in the system.
In vehicle audio systems, it has become possible to select different modes for reproducing an audio signal. By way of example, state of the art audio systems provide the possibility to either reproduce the sound in a stereo mode or in a surround sound mode. In the surround sound mode, additional time delays may be introduced in the different audio channels of the audio signal, so that the person sitting inside the vehicle has the impression of a surround sound audio system. When this audio system having a variable time delay in the different audio channels is used in connection with a vehicle communication system, the audio signal component emitted from the loudspeakers and then detected by the microphone should be removed to avoid unwanted echoes. In a surround sound mode, the signal amplifier introduces an additional time delay into the audio channel and the audio signal component detected by the microphone is delayed by the time delay introduced by the amplifier. Accordingly, an echo compensation unit for compensating acoustic echoes, by simulating the signal path from the loudspeaker to the microphone, should be able to simulate this signal path with a variable time delay. For the echo compensation of audio signal components from a signal detected by a microphone, a method with high computing power may be necessary. The required computer power mainly depends on the length of the filter of the echo compensation units. Thus generally, the greater the length of the filter, the more computer power needed.
Furthermore, it is possible that several microphones may be used for one seat to detect the speech signal of a passenger. Negative feedback can be avoided when adaptive filters are used for filtering out echoes and feedback signal components of the signals.
In addition to the communication signals output via the loudspeakers of the vehicle, audio modules reproducing audio signals, such as radio signals or signals from a music storage device such as a compact disc, are provided in the vehicles. These audio signals are output via the same loudspeakers, and they are also recorded by the microphones and again output via the loudspeaker. If these audio signal components are not attenuated before being output as part of the signal detected by the microphone, the driver has the impression of an audio sound signal having reverberation.
The above-described vehicle communication systems are often incorporated into expensive and highly sophisticated vehicles having highly sophisticated audio components. When the audio module is used in connection with a vehicle communication system, the sound quality is deteriorated by the feedback of the audio signal components picked up by the microphone and again fed to the loudspeakers. To avoid this signal quality degradation, the audio signal may be disabled during the in-vehicle communication, or the audio signal components detected by the microphone may be filtered out in an effective way.
For compensation of audio signal components in a sound signal (also referred to as “echo compensation”), a filter may be used to simulate the audio signal components of a sound signal that has been emitted from the loudspeaker and then detected by the microphone. However, the audio signal component may be, for example, an audio signal of a classical piece of music, a pop piece of music, or perhaps an interview without music. For all these different kinds of music, the echo compensation may have to be carried out in a different way to be effective. The audio signal components of the audio signal can have, in the case of a stereo signal for example, completely independent audio channels. In other situations, such as, for example, in the case of speaking interviews or one speaking person, the two audio signal parts of the stereo signal may be completely linear, depending on the signals. The echo compensation for linearly dependent signals is a difficult task, as the adaptation algorithms for calculating filter coefficients generally do not have a well-defined solution. When the audio signal changes from a piece of music to a person speaking, it is desirable for the filters to be adapted to the new signal characteristics. This adaptation of the filter tales a certain amount of time and during this time unwanted echoes can occur.
Moreover, echo compensation filters seek to simulate the path of the sound wave in the vehicle by calculating the pulse response. The approximation step may not result in a non-ambiguous and definite answer. Particularly in cases where the audio signal may be either a mono signal or a multi-channel signal, the different channels being completely linearly dependent from each other, a multi-channel stereo echo compensation filter may have the problem of finding the correct result. In other words, the stereo echo compensation filter may not be able to accurately simulate the interior of the vehicle through which the sound passed before it is detected by the microphone in a correct way.
Accordingly, a need exists to effectively cope with the different situations that can occur in the compensation of audio signal components in an echo compensation unit, and generally for an improved system and method for compensation of audio signal components in a vehicle communication system. A need further exists to reduce the length of filters while maintaining a length sufficient to allow the echo compensation unit to be able to simulate the signal path of a stereo signal or of a signal in a surround sound mode. Yet a further need exists to effectively cope with the different situations that can occur in the compensation of audio signal components in an echo compensation unit.
SUMMARYAn echo compensation system for compensating audio signal components in a communication system is provided. The communication system may include (i) an audio unit for generating an audio signal, (ii) a microphone for receiving a sound signal, (iii) a loudspeaker for outputting the sound signal detected by the microphone and outputting the audio signal itself, (iv) an echo compensation unit for compensating the audio signal components of the sound signal, and (v) a filter for whitening the sound signal, the audio signal, or both signals. Applicants note that the term “sound signals” (which may also be referred to as “detected sound signals”) refers to the signals detected by a microphone, including both audio signal components and speech signal components. The system may further include at least two filters used in an alternating way for whitening the sound signal, the audio signal, or both signals. The system may further include a sound signal having different audio channels, the time delay of the different audio channels relative to each other being adjustable.
A calculating unit may also be provided for calculating time-dependent filter coefficients. Additionally, a switch may be provided for switching the supply of the time-dependent filter coefficients to various audio signal filters. Furthermore, a second switch may also be provided to supply the simulated audio signal components to a subtracting unit, where the signal output from the echo compensation unit may be subtracted from the detected signal output. In addition, an inverse filter may be provided for removing the whitening of the whitened error signal resulting in the echo compensated sound signal, where this inverse filter may also be connected to the calculating unit.
A method for compensating audio signal components in a communication system is also provided. According to one implementation, a sound signal, comprising audio signal components and speech signal components, is detected by a microphone. The detected sound signal is then filtered in order to whiten the sound signal. After whitening the detected sound signal, the audio signal components in the sound signal are compensated. After compensation, the whitening of the compensated sound signal may be removed.
According to another implementation, filter coefficients may be calculated and supplied to two audio filters in an alternating way, to be used for whitening of signals. In such an implementation, the calculated filter coefficients may be supplied to a first filter for a first set of N cycles, and the calculated filter coefficients may be supplied to the other filter for a next set of N cycles resulting in a renewal of the filter coefficients of each filter every 2N cycles (i.e., new filter coefficients for a given filter calculated every 2N cycles).
According to another implementation of the invention, a system and method for compensating audio signal components in a communication system is provided using a mono echo compensation unit and a multi-channel (or stereo) echo compensation unit in combination. The provided echo compensation system may comprise a mono echo compensation unit for receiving one channel of an audio signal, and a multi-channel compensation unit for receiving at least two channels of the audio signal. When the audio signal changes its characteristic (for example, from music to a person speaking), either the mono echo compensation unit or the multi-channel echo compensation unit achieves the best echo compensation result. Accordingly, effective echo compensation can be achieved for any kind of audio signal.
According to yet another implementation, an echo compensation system is provided that is able to suppress audio signal components of an audio source having a variable time delay. In one implementation, the adaptation of the length of the variable time delay may be used alone, or in connection with other aspects or implementations of the invention. It is also possible that the variation of the length of the delay element may be used in combination with the time-dependent filter coefficients and/or in combination with the dual echo compensation structure of a mono echo compensation unit in combination with a multi-channel echo compensation unit, as described above.
These and other objects, features and advantages of the present invention, as well as other devices, apparatuses, systems, methods, features and advantages of the invention, will be or will become apparent to one with skill in the art upon examination of the following figures and detailed description. It is intended that all such additional systems, methods, features and advantages be included within this description, be within the scope of the invention, and be protected by the accompanying claims.
BRIEF DESCRIPTION OF THE FIGURESThe invention may be better understood by referring to the figures described below. The components in the figures are not necessarily to scale, emphasis instead being placed upon illustrating the principles of the invention. In the figures, like reference numerals designate corresponding parts throughout the different views.
While the present invention may be used in various types of communication systems, the invention will be described below with specific reference to an in-vehicle communication system as an example application of the invention.
When more than two microphones are used for one vehicle seat, a beam forming for the different vehicle seat positions can be done. In the example implementation illustrated in
The filtered audio signal channels xL(n) and xR(n) are then transmitted to an audio amplifier 22 for amplifying the audio signals before they are emitted via the loudspeakers 11. The filtered audio signal channels are also supplied to an echo compensation unit 23 where the audio signal components of a detected sound signal (not shown) may be removed. The audio signals emitted from the loudspeakers 11 propagate in the environment and may be diffracted different times before they are detected by one or more the microphones 13. The detected sound signal, comprising audio signal components as emitted by the loudspeaker 11 and also comprising speech signal components (such as from one or more of the passengers) are then fed to a processing unit 24 where linear processing (beam forming etc.) of the detected sound signal can be done. The output signals of the two units 23 and 24 are then fed to a subtracting unit 25 where the signal output from the echo compensation unit 23, {circumflex over (d)}(n), is subtracted from the detected signal output from the processing unit 24, d(n). The subtraction results in an error signal as discussed further below. The better the echo compensation can simulate the signal path from the loudspeakers 11 to the microphone 13, the smaller is the error signal e(n).
In the following, an example of the compensation of audio signal components according to the implementation illustrated in
hL(n)=[hL,0(n),hL,1(n), . . . , hL,L-1(n)]T (1)
hR(n)=[hR,0(n),hR,1(n), . . . , hR,L-1(n)]T (2)
The index n in equations (1) and (2) indicate the time dependence of the pulse responses. In one example, the signal path from the loudspeaker 11 to the microphone 13 is simulated by filtering the audio signal in such a way that after filtering, the filtered audio signal corresponds substantially to the audio signal as it was detected by the microphone 13. In this case, the unwanted audio signal component can be removed from the sound signal by subtracting the simulated audio signal component from the detected sound signal.
For compensating the acoustic echoes, one or more adaptive filters having the following pulse responses can be used:
ĥL(n)=[ĥL,0(n),ĥL,1(n), . . . , ĥL,N-1(n)]T (3)
ĥR(n)=[ĥR,0(n),ĥR,1(n), . . . , ĥR,N-1(n)]T (4)
Normally, digital filters are used having a large number of filter coefficients, e.g. 300-500 coefficients. The audio signal components as received by the microphones 13 can then be removed by subtracting the simulated signal component from the detected sound signal. The resulting signal is called an error signal e(n) and is defined as follows:
The signal d(n) is either the signal from the microphone 13 or the signal of a linear time invariant processing. A good compensation of the audio signal component can be achieved when the estimated pulse response corresponds to the actual pulse responses and when a sufficient number of coefficients are used. In echo compensation systems, the left and the right audio signal channels can have very different cross correlation characteristics. When music is reproduced as an audio sound signal, the square of the modulus of the coherence may be defined as:
C(Ω) normally has values of C(Ω)<1. When reproducing a news signal or other signal comprising one speaker, the left and the right audio signals may be linearly dependent signals, meaning that the coherence is approximately 1. In the above-shown equation (6) the values SxLxR(Ω),SxLxL(Ω) and SxRxR(Ω) are called the cross power spectral density or auto power spectral density of the left and right audio signal channels xL(n) and xR(n). When one of the audio signal components is an audio component that depends linearly on the other component, the adaptation algorithm compensating the acoustic echoes may not have a non-ambiguous single solution.
In the example illustrated in
The audio signals are filtered by the echo compensation filters 35a, 35b in such a way that the signal path in the vehicle is simulated. The echo compensation filters 35a, 35b determine the pulse response between the loudspeaker and the microphone. This can be done by using gradient methods and using least mean square (LMS) algorithms or normalized least mean square algorithms (NLMS). These methods and algorithms are known in the art and will not be discussed in detail.
When the acoustic path of the vehicle is simulated in the echo compensation filters 35a and 35b, the output signal is then fed to another switch 36, the switch 36 switching every N cycles, so that the filtered signals from echo compensation filter 35a are transmitted to the subtracting unit 37 for N cycles, before the switch 36 is switched and the signal from the echo compensation filter 35b is fed to the subtracting unit 37 for the next N cycles.
In the foregoing example, the two switches 34 and 36 change their respective states every N cycles, while at the same time each respectively maintaining a different actual state. Thus, when the switch 34 supplies data to the upper branch 33a and 35a, the switch 36 receives signal data from the lower branch 33b and 35b. In this example, the signal parameters in the filters 33a and 33b are renewed every 2N cycles, where the signal parameters in the filter 32 are renewed every N cycle. The output signal of filter 32 and the output signal of the echo compensation filters 35a or 35b are then used in the subtracting unit where the simulated signal from the respective echo compensation filter 35a, 35b is subtracted from the filtered sound signal as detected by the microphone 13. The result is a whitened error signal {tilde over (e)}(n). As it is known in adaptive filter systems, this whitened error signal {tilde over (e)}(n) is then used as a feedback control signal to adapt the audio signal echo compensation filters. The whitened error signal {tilde over (e)}(n) is then transmitted to an inverse filter 38 for removing the decorrelation. This inverse filter 38 also receives the calculated filter parameters every N cycles. The resulting error signal e(n) output from the inverse filter 38 then corresponds to the signal that will be output through the loudspeakers of the communication system. In this error signal e(n), the audio signal component is removed or suppressed. With the system shown in
In the example shown in
After whitening 44 (also referred to as decorrelating, since the whitening of a signal decorrelates the different channels of the signal), the acoustic echoes are compensated by compensating the audio signal components in the sound signal (step 45). This compensation may be carried out as explained in connection with
As illustrated in
Next, the filter coefficients calculated by the calculation unit 31 based, in this example, on the last 500 (N) cycles or input samples are transmitted to the first decorrelation filter 33a (step 52), which will use and/or store this set of filter coefficients for 2N cycles. During the time the filter coefficients are being calculated for the decorrelation filter 33a (i.e., the first N cycles), the other echo compensation filter 35b is being used (step 52a). The calculated filter coefficients calculated for the next N cycles are calculated in step 53 and are then transmitted to the other decorrelation filter 33b (step 54). For this next N cycles during which new filter coefficients are being calculated, the first echo compensation filter 35a is used (step 54a). In the method described with respect to
When the filter coefficients are supplied to the first decorrelation filter 33a as shown in
In the example of
The system of
The echo compensation unit shown in
When a mono audio signal or a multi-channel (stereo) audio signal having two linearly dependent signal channels is emitted through the loudspeakers, a mono echo compensation unit may achieve more desirable results than a multi-channel stereo echo compensation unit. When the sound signal has non-linearly depending signal channels, the stereo echo compensation unit can compensate the audio signal components in the sound signal and therefore the acoustic echoes more effectively. As both filters in the example described with respect to
Furthermore, in the case of a linearly dependent stereo signal or a mono signal, (e.g., an interview or other speech-only audio signal), the use of two different compensation units may increase the speed of echo compensation, as the mono echo compensation unit finds a solution in the approximation method much faster than the multi-channel echo compensation unit. Further, when the audio signal changes, for example, from a piece of music to a person speaking, the echo compensation may be adapted more quickly with a mono and multi-channel echo compensation unit operating in parallel, than it would be if only a multi-channel echo compensation unit were used. Moreover, the output from the echo compensation unit that would achieve the best echo compensation result (e.g., the mono echo compensation unit or the multi-channel echo compensation unit) may be selected.
In accordance with the system described with respect to
According to one implementation of the invention, the delay element comprises a delay element 92 of variable length, the delay element of variable length being connected to a signal memory 93 of the filter filtering the audio signal, the signal memory 93 of the filter having a constant length. With the delay element 92 of variable length it is possible to simulate the different time delays introduced by the amplifier of the audio signal. At the same time the signal memory 93 of the filter compensating the acoustic echoes can be of a relatively short length. In one example, the length of the delay element 92 is selected in such a way that the maximum of the pulse response calculated by the filter is located within a predetermined range of filter coefficients.
At the beginning the filter coefficients are 0. This pulse response was calculated based on the predetermined length of the delay memory. Above, the part 91a of the audio signal 91 is shown, which is comprised in the delay element 92. The other part 91b of the audio signal 91 is comprised in the signal memory 93 of the filter. With the length of the delay element 92 shown in
When it is detected that the maximum 95a of the pulse response is not located at a predetermined filter coefficient, the pulse response is shifted as shown in
This means that the direct sound as it is simulated by the echo compensation filter is situated at a predetermined filter coefficient of the filter. By way of example, the maximum of the pulse response can be arranged at a filter coefficient which is between one tenth and one twentieth of the maximum filter coefficient. By way of example, it is supposed that the filter compensating the acoustic echoes has a length of 500 coefficients. In this example the delay element may be controlled in such a way that the maximum of the pulse response in the calculated pulse response is positioned between the 20th and the 40th filter coefficient, preferably between the 25th and 35th filter coefficient, even preferably between the 28th and the 32nd filter coefficient.
Preferably, the maximum of the pulse response can be calculated by the following equation:
iD(n)=arg max(|hi(n)|γi). (7)
As can be seen by equation (7), the coefficient representing the direct sound can be found by searching for the maximum of a weighted modulus of the pulse response. Preferably, the parameter γ is chosen to be between 0 and 1. By introducing this parameter γ, reflections of the sound signal may be attenuated relative to the direct sound. When the maximum of the pulse response in the simulated signal path in the echo compensation filter is found to be at a much larger filter coefficient, this means that the simulated time delay may be smaller than desired. In this case, a further time delay may be introduced. If, however, it is determined that the maximum of the pulse response is located at a filter coefficient having a number which is smaller than the number of the predetermined range, it can be followed that the simulated time delay may be larger than desired. In this case, the delay introduced by the delay element may be made shorter.
It should be understood that the implementations described in connection with
Although the invention has been shown and described with respect to example implementations thereof, it should be understood by those skilled in the art that the description is example rather than limiting in nature, and that many changes, additions and omissions are all possible without departing from the scope and spirit of the present invention, which should be determined from the following claims.
Claims
1. A method for compensating an audio signal in a communication system comprising the steps of:
- detecting a sound signal, the sound signal comprising a detected audio signal component from an audio signal and a speech signal component;
- filtering the sound signal in order to create a whitened sound signal;
- filtering the audio signal component in order to create a whitened audio signal component, where the step of filtering the audio signal component is performed using at least two filters in an alternating way, each filter using time-dependent filter coefficients;
- compensating the audio signal component in the whitened sound signal; and
- inverse filtering the compensated whitened sound signal.
2. The method of claim 1, further comprising the step of calculating the time-dependent filter coefficients.
3. The method of claim 2, where the step of filtering the sound signal is performed using the time-dependent filter coefficients.
4. The method of claim 3, further comprising the step of renewing the time-dependent filter coefficients every N cycles.
5. The method of claim 3, further comprising alternately supplying the time-dependent filter coefficients to a first filter and a second filter, where time-dependent filter coefficients calculated for a first set of N cycles are supplied to the first filter, and time-dependent filter coefficients calculated for a next set of N cycles are supplied to the second filter, such that the time-dependent filter coefficients for the first filter and the second filter, respectively, are renewed every 2N cycles.
6. The method of claim 1, where the step of compensating the audio signal component in the whitened sound signal further comprises: simulating the whitened audio signal component to create a whitened simulated audio signal component; and determining a whitened error signal by subtracting the simulated audio signal component from the sound signal.
7. The method of claim 6, further comprising determining an estimated sound signal component using the error signal as a feedback control signal.
8. The method of claim 7, where the step of simulating the audio signal component of the sound signal is performed by an echo compensation filter having a length equal to N.
9. The method of claim 6, where the step of determining a whitened error signal comprises alternately subtracting a first whitened simulated audio signal component from the whitened sound signal, and a second whitened simulated audio signal component from the whitened sound signal.
10. The method of claim 9, further comprising inverse filtering the whitened simulated error signal, resulting in an error signal corresponding to an echo compensated sound signal.
11. The method of claim 9, where the step of determining a whitened error signal comprises alternately subtracting a first whitened simulated audio signal component from the whitened sound signal for a first set of N cycles, and a second whitened simulated audio signal component from the whitened sound signal for a next set of N cycles.
12. A method for compensating an audio signal comprising the steps of:
- detecting a sound signal, the sound signal comprising a detected audio signal component from an audio signal, and a speech signal component;
- calculating time dependent filter coefficients;
- filtering the sound signal to create a whitened sound signal using a decorrelation filter, said decorrelation filter being supplied with the time dependent filter coefficients every N cycles;
- filtering the audio signal component to create a whitened audio signal component where the step of filtering the audio signal component is performed using at least two filters in an alternate way, said two filters being supplied alternately every 2N cycles with said time dependent filter coefficients;
- compensating the audio signal component in the whitened sound signal; and
- inverse filtering the compensated whitened sound signal.
13. An echo compensation system comprising:
- at least one microphone for detecting a sound signal, the sound signal comprising a detected audio signal component from an audio signal and a speech signal component;
- at least one loudspeaker for outputting the sound signal and the audio signal;
- a filter unit comprising a first filter and a second filter used in an alternating way for filtering the audio signal component, where each filter uses time-dependent filter coefficients; and
- an echo compensation unit for compensating the audio signal component received by the microphone.
14. The echo compensation system of claim 13, further comprising a calculating unit for calculating the time-dependent filter coefficients.
15. The echo compensation system of claim 14, further comprising a first switch in communication with the calculating unit and the first and second filters, for alternately supplying the time-dependent filter coefficients to the first filter and the second filter.
16. The echo compensation system of claim 15, where the first switch switches between a first position and a second position every N cycles.
17. The echo compensation system of claim 13, where the filter unit further comprises a third filter for filtering the sound signal and outputting a whitened sound signal component, where the third filter receives the time-dependent filter coefficients calculated by the calculating unit, and the time-dependent filter coefficients received by the third filter are refreshed every N cycles.
18. The echo compensation system of claim 13, where the echo compensation unit comprises a first echo compensation filter in communication with the first filter and a second echo compensation filter in communication with the second filter, where the first echo compensation filter receives a first whitened audio signal component from the first filter and outputs a first whitened simulated audio signal component, and the second echo compensation filter receives a second whitened audio signal component from the second filter and outputs a second whitened simulated audio signal component.
19. The echo compensation system of claim 18, where the echo compensation unit further comprises a subtracting unit where the first and second whitened simulated audio signal components are subtracted from the whitened sound signal component, resulting in a whitened error signal.
20. The echo compensation unit of claim 19, where the whitened error signal is used as a feedback control signal for the echo compensation filters.
21. The echo compensation unit of claim 13, further comprising an inverse filter for inverse filtering the whitened error signal, and outputting an echo compensated sound signal.
22. The echo compensation unit of 13, further comprising a third filter and a fourth filter, where the first and third filters correspond to a first audio signal component, and the second and fourth filters correspond to a second audio signal component; and four echo compensation filters, where two of the echo compensations filters correspond to the first audio signal component and the other two echo compensation filters correspond to the second audio signal component.
23. The echo compensation unit of claim 18, further comprising a switch for alternately supplying the first whitened simulated audio signal component and the second whitened simulated audio signal component to the subtracting unit, the switch alternating every N cycles.
24. An echo compensation system comprising:
- at least one microphone for detecting a sound signal, the sound signal comprising a detected audio signal component from an audio signal and a speech signal component;
- at least one loudspeaker for outputting the sound signal and the audio signal;
- a calculating unit for calculating time dependent filter coefficients based on the audio signal;
- at least two filters for alternately filtering the audio signal, said filters being alternately supplied with said time-dependent filter coefficients every 2N cycles;
- a sound signal filter for filtering the sound signal, the sound signal filter being supplied with said time dependent filter coefficients every N cycles;
- an echo compensation unit for compensating the audio signal component.
25. A method for compensating audio signal components comprising the steps of:
- detecting a sound signal, the sound signal comprising a detected audio signal component from an audio signal comprising a first channel and a second channel, and a speech signal component;
- generating an echo compensated sound signal to compensate acoustic echoes in the sound signal due to the detected audio signal component in the sound signal, where the generating step comprises the steps of:
- supplying the first channel of the audio signal to a mono echo compensation unit;
- supplying the first and second channels of the audio signal to a multi channel echo compensation unit;
- outputting a first output associated with a first signal power from the mono echo compensation unit, and a second output associated with a second signal power from the multi channel echo compensation unit;
- comparing the first signal power and the second signal power;
- selecting the first output if the first signal power is smaller than the second signal power; and
- selecting the second output if the second signal power is smaller than the first signal power.
26. The method of claim 25, further comprising the step of filtering the sound signal in order to obtain a whitened sound signal before the step of generating the echo compensated sound signal, and inverse filtering the selected output.
27. The method of claim 25, where the step of generating an echo compensated sound signal further comprises: generating a first simulated audio signal component for the first channel and a second simulated audio signal component for the second channel using the multi channel echo compensation unit; and adding the first and second simulated audio signals to obtain a combined simulated audio signal component.
28. The method of claim 26, further comprising calculating time-dependent filter coefficients to be used for obtaining the whitened sound signal.
29. The method of claim 27, further comprising subtracting a mono simulated audio signal component from the sound signal to obtain the first output, and subtracting the combined simulated audio signal component from the sound signal to obtain the second output.
30. A method for compensating audio signal components comprising the steps of:
- detecting a sound signal, the sound signal comprising a detected audio signal component from an audio signal comprising a first channel and a second channel, and a speech signal component;
- filtering the sound signal to obtain a whitened sound signal;
- filtering the first channel to obtain a first whitened audio signal component;
- filtering the second channel to obtain a second whitened audio signal component;
- supplying the first whitened audio signal component and the whitened sound signal to a mono echo compensation unit;
- outputting a first output having a first signal power from the mono echo compensation unit;
- supplying the first whitened audio signal component, the second whitened audio signal component, and the whitened sound signal to a multi channel echo compensation unit;
- outputting a second output having a second signal power from the multi channel echo compensation unit;
- comparing the first signal power and the second signal power;
- selecting the first output if the first signal power is smaller than the second signal power; and
- selecting the second output if the second signal power is smaller than the first signal power.
31. An echo compensation system comprising:
- at least one microphone for detecting a sound signal, the sound signal comprising a detected audio signal component from an audio signal comprising a first channel and a second channel, and a speech signal component;
- at least one loudspeaker for outputting the sound signal;
- a mono echo compensation unit for receiving the first channel of the audio signal and outputting first output having a first signal power;
- a multi channel echo compensation unit for receiving the first and second channels of the audio signal and outputting a second output having a second signal power; and
- a comparison unit for comparing the first signal power and the second signal power; and
- selecting the first output if the first signal power is lower than the second signal power, or the second output if the second signal power is lower than the first signal power.
32. The echo compensation system of claim 31, further comprising a plurality of filters to whiten the audio signal and the sound signal, and an inverse filter for inverse filtering at least one of the first output and the second output.
33. The echo compensation system of claim 32, where the plurality of filters includes at least one filter for the first channel and at least one filter for the second channel.
34. An echo compensation system comprising:
- at least one microphone for detecting a sound signal, the sound signal comprising a detected audio signal component from an audio signal comprising a first channel and a second channel, and a speech signal component;
- at least one loudspeaker for outputting the sound signal;
- a filter unit for generating a whitened sound signal;
- a plurality of filter units for generating a whitened audio signal, the whitened audio signal comprising a first whitened audio signal corresponding to the first channel and a second whitened audio signal corresponding to the second channel;
- a mono echo compensation unit being supplied with the first whitened audio signal and with the whitened sound signal, and outputting a first output;
- a multi channel echo compensation unit being supplied with the first whitened audio signal, the second audio signal, and the whitened sound signal, and outputting a second output; and
- a comparison unit for comparing a signal power of the first output and a signal power of the second output, and selecting whichever of the first output or second output has a lower signal power.
35. An echo compensation system comprising:
- an audio source for generating an audio signal having a first channel with a first time delay and a second channel with a second time delay, where the first and second time delays are adjustable relative to each other;
- at least one microphone for detecting a sound signal, the sound signal comprising a detected audio signal component from the audio signal, and a speech signal;
- a loudspeaker unit for outputting the audio signal and the sound signal;
- an echo compensation unit for simulating the audio signal component to obtain a simulated audio signal component, and subtracting the simulated audio signal component from the sound signal, the echo compensation unit comprising:
- a filter for filtering the audio signal to obtain a pulse response of the audio signal, the pulse response having a maximum value;
- a plurality of filter coefficients corresponding to the filter;
- a delay element for introducing a variable time delay corresponding to the audio signal; and
- a delay control unit for controlling the delay element so that the maximum value of the pulse response is located within a predetermined range of the filter coefficients.
36. The echo compensation system of claim 35, where the delay element comprises a delay memory having a variable length.
37. The echo compensation system of claim 36, where the delay element is in communication with a signal memory, the signal memory of the filter having a constant length.
38. The echo compensation system of claim 35, where a filter coefficient corresponding to the maximum value of the pulse response is located between a tenth and a twentieth filter coefficient.
39. The echo compensation system of claim 35, where the maximum value of the pulse response is located between a twentieth and a fortieth filter coefficient
40. The echo compensation system of claim 35, where the maximum value of the pulse response is located between a twenty-fifth and a thirty-fifth filter coefficient.
41. The echo compensation system of claim 35, where the maximum value of the pulse response is located between a twenty-eighth and a thirty-second filter coefficient.
42. The echo compensation system of claim 35, where the delay control unit determines at which filter coefficient the maximum value of the impulse response is positioned.
43. The echo compensation system of claim 35, further comprising:
- a filter for generating a whitened sound signal using time-dependent filter coefficients;
- a plurality of filter units for generating a whitened delayed audio signal, where at least one filter unit comprises two audio signal filters, each audio signal filter using time-dependent filter coefficients for generating corresponding whitened delayed audio signals, said at least two audio signal filters being used in an alternating way for generating the whitened delayed audio signals;
- a mono echo compensation unit comprising at least two echo compensation filters alternately receiving the whitened delayed audio signals from the audio signal filters, and outputting a mono compensated sound signal having a first signal power;
- a multi channel echo compensation unit comprising at least two echo compensation filters alternately receiving the whitened delayed audio signals from the audio signal filters, and outputting a multi channel compensated sound signal having a second signal power; and
- a comparison unit for comparing the first signal power and the second signal power, and selecting whichever of the mono compensated sound signal and the multi channel compensated sound signal has a lower signal power.
44. An echo compensation system comprising:
- an audio source for generating an audio signal having a first channel with a first time delay and a second channel with a second time delay, where the first and second time delays are adjustable relative to each other;
- at least one microphone for detecting a sound signal, the sound signal comprising a detected audio signal component from the audio signal, and a speech signal;
- a loudspeaker unit for outputting the audio signal and the sound signal;
- a delay element comprising a delay memory for generating a delayed audio signal;
- a calculating unit for calculating a pulse response corresponding to the delayed audio signal and the sound signal, the pulse response having a maximum value;
- an echo compensation unit for obtaining a compensated audio signal component and
- a delay control unit for controlling the delay element so that the maximum value of the pulse response is located within a predetermined range of filter coefficients.
45. A method for compensating audio signal components comprising the steps of:
- reproducing an audio signal, the audio signal having a first channel with a first time delay and a second channel with a second time delay, where the first and second time delays are adjustable relative to each other;
- outputting the audio signal;
- detecting a sound signal, the sound signal comprising a detected audio signal component from the audio signal, and a speech signal component;
- outputting the sound signal;
- generating a simulated audio signal component;
- subtracting the simulated audio signal components from the sound signal;
- simulating a signal path of the audio signal from the loudspeaker to the microphone by determining a pulse response of the audio signal; the pulse response having a maximum value; and
- introducing a variable time delay selected so that the maximum value of the pulse response is located within a predetermined range of filter coefficients of the filter.
46. The method of claim 45, further comprising the step of adding a variable time delay to the audio signal by supplying the audio signal to a delay element of variable length.
47. The method of claim 45, further comprising the step of determining the maximum value of the pulse response and determining at which filter coefficient the determined maximum value is located.
48. The method of claim 47 where the step of determining the maximum value of the pulse response comprises determining the maximum value of a weighted modulus of the pulse response.
49. The method of claim 45, further comprising varying the length of a delay element.
50. The method of claim 49, further comprising increasing the delay element when it is determined that the maximum value of the pulse response is positioned at a filter coefficient having a number larger than the predetermined range.
51. The method of claim 50, further comprising decreasing the delay element when it is determined that the maximum value of the pulse response is positioned at a filter coefficient having a number smaller than the predetermined range.
52. The method of claim 45, further comprising shifting the pulse response so that the maximum of the pulse response is located within the predetermined range of filter coefficients, if the determined pulse response is not located within a predetermined range of filter coefficients.
53. The method of claim 45, further comprising the steps of:
- filtering the sound signal to obtain a whitened sound signal;
- filtering the first channel and the second channel to obtain a first whitened delayed channel and a second whitened delayed channel, where the filtering of the first and second channels is performed using at least two audio filters per channel in an alternating way, each audio filter using time-dependent filter coefficients;
- obtaining a compensated mono sound signal having a first signal power using a mono echo compensation unit;
- obtaining a compensated multi channel sound signal having a second signal power using a multi channel echo compensation unit;
- comparing the compensated first signal power and the second signal power; and
- selecting whichever of the mono sound signal and the compensated multi channel sound signal has a lower signal power.
54. A method for compensating audio signal components comprising the steps of:
- reproducing an audio signal, the audio signal having a first channel with a first time delay and a second channel with a second time delay, where the first and second time delays are adjustable relative to each other;
- outputting the audio signal;
- detecting a sound signal, the sound signal comprising a detected audio signal component from the audio signal, and a speech signal component;
- outputting the sound signal;
- delaying the audio signal using a delay element;
- calculating a pulse response from the delayed audio signal and the sound signal, the pulse response having a maximum value;
- generating a simulated audio component using the delayed audio signal and a filter, said filter using the calculated pulse response as filter coefficients;
- controlling a delay introduced by the delay element so that the maximum value of the pulse response is positioned within a predetermined range of filter coefficients; and
- subtracting the simulated audio component from the sound signal.
Type: Application
Filed: Jul 11, 2007
Publication Date: Jan 17, 2008
Applicant: Harman Becker Automotive Systems GmbH (Karlsbad)
Inventors: Gerhard Schmidt (Ulm), Tim Haulick (Blaubeuren), Harald Lenhardt (Ulm)
Application Number: 11/776,432
International Classification: G10L 11/00 (20060101); H04B 15/00 (20060101);