WO2009081184A1

WO2009081184A1 - Noise cancellation system and method with adjustment of high pass filter cut-off frequency

Info

Publication number: WO2009081184A1
Application number: PCT/GB2008/051176
Authority: WO
Inventors: Anthony James Magrath
Original assignee: Wolfson Microelectronics Plc
Priority date: 2007-12-21
Filing date: 2008-12-11
Publication date: 2009-07-02
Also published as: GB0810996D0; GB2455823B; GB0725112D0; GB2455823A

Abstract

There is provided a noise cancellation system, comprising: an input, for receiving a digital signal; a digital filter, having at least a high pass filter characteristic, for receiving the digital signal and generating a filter output signal; and an amplitude detector, for generating a detection signal based on an amplitude of a representation of said filter output signal, wherein the detection signal is applied to the digital filter to control a cut- off frequency thereof.

Description

NOISE CANCELLATION SYSTEM AND METHOD WITH ADJUSTMENT OF HIGH PASS FILTER CUT-OFF FREQUENCY

This invention relates to a noise cancellation system, and in particular to a noise cancellation system having a filter that can easily be adapted based on an input signal in order to improve the noise cancellation performance.

BACKGROUND

Noise cancellation systems are known, in which an electronic noise signal representing ambient noise is applied to a signal processing circuit, and the resulting processed noise signal is then applied to a speaker, in order to generate a sound signal. In order to achieve noise cancellation, the generated sound should approximate as closely as possible the inverse of the ambient noise, in terms of its amplitude and its phase.

In particular, feedforward noise cancellation systems are known, for use with headphones or earphones, in which one or more microphones mounted on the headphones or earphones detect an ambient noise signal in the region of the wearer's ear. In order to achieve noise cancellation, the generated sound then needs to approximate as closely as possible the inverse of the ambient noise, after that ambient noise has itself been modified by the headphones or earphones. One example of modification by the headphones or earphones is caused by the different acoustic path the noise must take to reach the wearer's ear, travelling around the edge of the headphones or earphones.

The microphone used to detect the ambient noise signal and the loudspeaker used to generate the sound signal from the processed noise signal will in practice also modify the signals, for example being more sensitive at some frequencies than at others. One example of this is when the speaker is closely coupled to the ear of a user, causing the frequency response of the loudspeaker to change due to cavity effects.

The signal processing circuit is therefore required to have a frequency-dependent characteristic. However, there may be situations where an attempt to cancel fully the detected noise signal would require the noise signal, that is to be applied to the speaker, to have a higher amplitude than can be handled by the speaker. This would have disadvantageous effects on the listener. SUMMARY OF INVENTION

According to a first aspect of the present invention, there is provided a noise cancellation system, comprising: an input, for receiving a digital signal; a digital filter, having at least a high pass filter characteristic, for receiving the digital signal and generating a filter output signal; and an amplitude detector, for generating a detection signal based on an amplitude of a representation of said filter output signal, wherein the detection signal is applied to the digital filter to control a cut-off frequency thereof.

This has the advantage that the amplitude of the noise signal, that is to be applied to the speaker, can be limited.

According to a second aspect of the present invention, there is provided a method of controlling a filter for a noise cancellation system. The method comprises receiving a digital signal; filtering the digital signal in a digital filter, to generate a filter output signal; detecting an amplitude of a representation of said filter output signal; generating a detection signal based on said detected amplitude; and applying said detection signal to said digital filter, to control a cut-off frequency thereof.

BRIEF DESCRIPTION OF THE DRAWINGS

For a better understanding of the present invention, and to show more clearly how it may be carried into effect, reference will now be made, by way of example, to the following drawings, in which:

Figure 1 illustrates a noise cancellation system in accordance with an aspect of the invention;

Figure 2 illustrates a signal processing circuit in accordance with an aspect of the invention in the noise cancellation system of Figure 1 ;

Figure 3 is a flow chart, illustrating a process in accordance with an aspect of the invention; and

Figure 4 illustrates a signal processing circuit appropriate for use in a feedback noise cancellation system in accordance with the present invention. DETAILED DESCRIPTION

Figure 1 illustrates in general terms the form and use of a noise cancellation system in accordance with the present invention.

Specifically, Figure 1 shows an earphone 10, being worn on the outer ear 12 of a user 14. Thus, Figure 1 shows a supra-aural earphone that is worn on the ear, although it will be appreciated that exactly the same principle applies to circumaural headphones worn around the ear and to earphones worn in the ear such as so-called ear-bud phones. The invention is equally applicable to other devices intended to be worn or held close to the user's ear, such as mobile phones and other communication devices.

Ambient noise is detected by microphones 20, 22, of which two are shown in Figure 1 , although any number more or less than two may be provided. Ambient noise signals generated by the microphones 20, 22 are combined, and applied to signal processing circuitry 24, which will be described in more detail below. In one embodiment, where the microphones 20, 22 are analogue microphones, the ambient noise signals may be combined by adding them together. Where the microphones 20, 22 are digital microphones, i.e. where they generate a digital signal representative of the ambient noise, the ambient noise signals may be combined alternatively, as will be familiar to those skilled in the art. Further, the microphones could have different gains applied to them before they are combined, for example in order to compensate for sensitivity differences due to manufacturing tolerances.

This illustrated embodiment of the invention also contains a source 26 of a wanted signal. For example, where the noise cancellation system is in use in an earphone, such as the earphone 10 that is intended to be able to reproduce music with a relatively high quality, the source 26 may be an inlet connection for a wanted signal from an external source such as a sound reproducing device. In other applications, for example where the noise cancellation system is in use in a mobile phone or other communication device, the source 26 may include wireless receiver circuitry for receiving and decoding radio frequency signals. In other embodiments, there may be no source, and the noise cancellation system may simply be intended to cancel the ambient noise for the user's comfort. The wanted signal, if any, from the source 26 is applied through the signal processing circuitry 24 to a loudspeaker 28, which generates a sound signal in the vicinity of the user's ear 12. In addition, the signal processing circuitry 24 generates a noise cancellation signal that is also applied to the loudspeaker 28.

An aim of the signal processing circuitry 24 is to generate a noise cancellation signal, which, when applied to the loudspeaker 28, causes it to generate a sound signal in the ear 12 of the user that is the inverse of the ambient noise signal reaching the ear 12.

In order to achieve this, the signal processing circuitry 24 needs to generate from the ambient noise signals generated by the microphones 20, 22 a noise cancellation signal that takes into account the properties of the microphones 20, 22 and of the loudspeaker 28, and also takes into account the modification of the ambient noise that occurs due to the presence of the earphone 10.

Figure 2 shows in more detail the form of the signal processing circuitry 24. An input 40 is connected to receive an input signal, for example directly from the microphones 20, 22. This input signal is amplified in an amplifier 41 and the amplified signal is applied to an analog-digital converter 42, where it is converted to a digital signal. The digital signal is applied to an adaptable digital filter 44, and the filtered signal is applied to an adaptable gain device 46. Those skilled in the art will appreciate that in the case where the microphones 20, 22 are digital microphones, wherein an analog-digital converter is incorporated into the microphone capsule and the input 40 receives a digital input signal, the analog-digital converter 42 is not required.

The resulting signal is applied to an adder 48, where it is summed with the wanted source signal received from a second input 49, to which the source 26 may be connected.

Thus, the filtering and level adjustment applied by the filter 44 and the gain device 46 are intended to generate a noise cancellation signal that allows the detected ambient noise to be cancelled.

The output of the adder 48 is applied to a digital-analog converter 50, so that it can be passed to the loudspeaker 28. As mentioned above, the noise cancellation signal is produced from the input signal by the adaptable digital filter 44 and the adaptable gain device 46. These are controlled by a control signal, which is generated by applying the digital signal output from the analog-digital converter 42 to a decimator 52 which reduces the digital sample rate, and then to a microprocessor 54. In a preferred embodiment, the sample rate of the input digital signal is 2.4 MHz, and the decimator reduces this sample rate to 8 kHz.

The microprocessor 54 contains a block 56 that emulates the filter 44 and gain device 46, and produces an emulated filter output which is applied to an adder 58, where it is summed with the wanted signal from the second input 49, via a decimator 89. The sample rate reduction performed by the decimator 52 allows the emulation to be performed with lower power consumption than performing the emulation at the original 2.4 MHz sample rate.

The resulting signal is applied to a control block 60, which generates control signals for adjusting the properties of the filter 44 and the gain device 46. The control signal for the filter 44 is applied through a frequency warping block 62, a smoothing filter 64 and sample-and-hold circuitry 66 to the filter 44. The same control signal is also applied to the block 56, so that the emulation of the filter 44 matches the adaptation of the filter 44 itself.

The purpose of the frequency warping block 62 is to adapt the control signal output from the control block 60 for the high-frequency adaptive filter 82. That is, the high- frequency filter 82 will generally be operating at a frequency that is much higher than that of the low-frequency filter emulator 86, and therefore the control signal will generally need to be adapted in order to be applicable to both filters.

In an alternative embodiment, the sample-and-hold circuitry 66 may be replaced by an interpolation filter.

The control block 60 further generates a control signal for the adaptive gain device 46. In the illustrated embodiment, the gain control signal is output directly to the gain device 46. In this illustrated embodiment of the invention, the digital filter 44 is formed from a fixed first filtering stage, in the form of an infinite impulse response filter (NR) 80 and an adaptive second filtering stage, in the form of an adaptive high-pass filter 82.

The filter emulation 56 is therefore similarly formed from an NR emulator 84 and a high- pass filter 86.

In one embodiment of the invention, the NR filter 80 has a filter characteristic which preferentially passes signals at relatively low frequencies. For example, although the noise cancellation system may seek to cancel ambient noise as far as possible across the whole of the audio frequency band, the particular arrangement of the microphones 20, 22, and the speaker 28, and the size and shape of the earphone 10, may mean that it is preferred for the NR filter 80 to have a filter characteristic which boosts signals at frequencies in the 250 - 750 Hz region. However, in other embodiments, the NR filter 80 may have a significant boost below 250 Hz as well. This boost may be needed to compensate for small speakers mounted in small enclosures, which generally have a poor low-frequency response.

However, this means that, when there is an ambient noise signal having a large component within this frequency range, there is a danger that the noise signal generated by the filter 80 will be larger than the speaker 28 can comfortably handle without distortion, etc, i.e. the speaker 28 may be overdriven. Should this occur, the speaker cone may move beyond its excursion limit, resulting in physical damage to the speaker.

Therefore, in order to prevent this, the frequency characteristic of the adaptive high- pass filter 82 is adapted, based on the amplitude of the input signal. In fact, in this preferred embodiment, the frequency characteristic of the adaptive high-pass filter 82 is adapted, based on the output signal from the emulated filter 56. Moreover, in this preferred embodiment, the frequency characteristic of the adaptive high-pass filter 82 is adapted, based on the sum of the wanted signal from the second input 49 and the output signal from the emulated filter 56. This means that the frequency characteristic of the adaptive high-pass filter 82 is adapted based on a representation of the signal that would actually be applied to the speaker 28. More specifically, in this illustrated embodiment of the invention, the adaptive high-pass filter 82 is a first-order high pass filter, with a cut-off frequency, or corner frequency, which can be adjusted based on the control signal applied from the microprocessor 54. The filter 82 has a generally constant gain, which may be unity or may be some other value provided that there is suitable compensation elsewhere in the filter path, at frequencies above the corner frequency, and has a gain that reduces below that corner frequency.

In one embodiment, the corner frequency may be adjustable in the range from 10 Hz to 1.4 kHz.

Figure 3 is a flow chart, illustrating the process performed in the control block 60.

In step 90, the process is initialized, by setting an initial value for a control value K, which is used to control the corner frequency of the high pass filter 82.

In step 92, the input value to the control block 60, namely the absolute value of the sum H of the emulated filter block 56 and the wanted source input 49, is compared with a threshold value T. If the sum H exceeds the threshold value T, the process passes to step 94, in which an attack coefficient K_A is added to the current control value K. After adding these values together, it is tested in step 96 whether the new control value exceeds an upper limit value and, if so, this upper limit value is applied instead. If the new control value does not exceed the upper limit value, the new control value is used.

If in step 92 the absolute value of the sum H is lower than the threshold value T, the process passes to step 98, in which a decay coefficient K_D is added to the current control value K. It should be noted that the decay coefficient K_D is negative, and so adding it to the current control value K reduces that value. After adding these values together, it is tested in step 100 whether the new control value falls below a lower limit value and, if so, this lower limit value is applied instead. If the new control value does not fall below the lower limit value, the new control value is used.

When the new control value has been determined, the process returns to step 92, where the new sum H of the emulated filter block 56 and the wanted source input 49 is compared with the threshold value T. In one embodiment, the attack coefficient K_A is larger in magnitude that the decay coefficient K_D, so that if a transient low frequency signal occurs, the cut-off frequency can be increased rapidly, resulting in a fast reduction in output amplitude to prevent the speaker exceeding its excursion limit. Further, a relatively smaller decay coefficient minimizes any ripple in the cut-off frequency, so that the cut-off frequency effectively tracks the envelope of the input signal, rather than the absolute value.

Further, it will be apparent to those skilled in the art that other implementations of the control algorithm performed in control block 60 are possible, in order to alter the cut-off frequency appropriately to prevent speaker overload. For example, the attack and decay coefficients K_A and K_D could be varied in a non-linear (e.g. exponential) way.

As described above, the control process is performed at a lower sample rate than the sample rate of the input digital signal. In order to ensure that this is not a source of errors, the control value is passed through a frequency warping function 62.

Further, the control value is passed through a smoothing filter 64, which is provided to smooth any unwanted ripple in the signal. In this embodiment, the filter determines whether the control value is increasing or decreasing. If the control value is increasing, the output of the filter 64 tracks the input directly, without any time lag. However, if the control value is decreasing, the output of the filter 64 decays exponentially towards the input, in order to smooth any unwanted ripple in the output signal.

The output of the smoothing filter 64 is passed to sample-and-hold circuitry 66, from which it is latched out to the adaptive filter 82. The corner frequency of the filter 82 is then determined by the filtered control value applied to the filter. For example, when the control value takes the lower limit value, the corner frequency can take its minimum value, of 10 Hz in the illustrated embodiment, while, when the control value takes the upper limit value, the corner frequency can take its maximum value, namely 1.4 kHz in the illustrated embodiment.

In an alternative embodiment, the sample-and-hold circuitry 66 may be replaced by an interpolation filter. The control block 60 further generates a control signal for the adaptive gain device 46. In the illustrated embodiment, the gain control signal is output directly to the gain device 46.

Thus, the amplitude of the filter output can be determined, based on an emulation of the filter, and, when that amplitude exceeds a threshold, the corner frequency of the high-pass filter can be increased, so that the low frequency component of the filtered signal is reduced in amplitude, until the amplitude is below the threshold. In this manner, the danger of driving the speaker 28 with an excessively large signal can be reduced.

In this embodiment, the determination of the filter output is made based on the sum of the emulated filter output and the source signal. For example, the determination may be based on an absolute value of the sum, or a root mean square value of the sum. In other embodiments, the determination can be made based on the emulated filter output alone, without taking into account the contribution of the wanted signal.

In other embodiments, the determination may be made based not on the output of the emulation of the filter, but based on the output of the filter itself, either directly or after the wanted source signal has been added thereto. For example,

Various modifications may be made to the embodiments described above without departing from the scope of the claims appended hereto. For example, the source signal input to the signal processor 24 may be digital, as described above, or analogue, in which case an analog-digital converter may be necessary to convert the signal to digital.

It will also be apparent to those skilled in the art that the present invention is equally applicable to so-called feedback noise cancellation systems.

The feedback method is based upon the use, inside the cavity that is formed between the ear and the inside of an earphone shell, or between the ear and a mobile phone, of a microphone placed directly in front of the loudspeaker. Signals derived from the microphone are coupled back to the loudspeaker via a negative feedback loop (an inverting amplifier), such that it forms a servo system in which the loudspeaker is constantly attempting to create a null sound pressure level at the microphone. Figure 4 shows an example of signal processing circuitry according to the present invention when implemented in a feedback system.

The feedback system comprises a microphone 120 positioned substantially in front of a loudspeaker 128. The microphone 120 detects the output of the loudspeaker 128, with the detected signal being fed back via an amplifier 141 and an analog-to-digital converter 142. A wanted audio signal is fed to the processing circuitry via an input 140. The fed back signal is subtracted from the wanted audio signal in a subtracting element 188, in order that the output of the subtracting element 188 substantially represents the ambient noise, i.e. the wanted audio signal has been substantially cancelled.

Thereafter, the processing circuitry is substantially similar to the processing circuitry 24 in the feed forward system described with respect to Figure 2. The output of the subtracting element 188 is fed to an adaptive digital filter 144, and the filtered signal is applied to an adaptable gain device 146.

The resulting signal is applied to an adder 148, where it is summed with the wanted audio signal received from the input 140.

Thus, the filtering and level adjustment applied by the filter 144 and the gain device 146 are intended to generate a noise cancellation signal that allows the detected ambient noise to be cancelled.

The output of the adder 148 is applied to a digital-analog converter 150, so that it can be passed to the loudspeaker 128.

As mentioned above, the noise cancellation signal is produced from the input signal by the adaptive digital filter 144 and the adaptable gain device 146. These are controlled by a control signal, which is generated by applying the digital signal output from the analog-digital converter 142 to a decimator 152 which reduces the digital sample rate, and then to a microprocessor 154.

The microprocessor 154 contains a block 156 that emulates the filter 144 and gain device 146, and produces an emulated filter output which is applied to an adder 158, where it is summed with the wanted audio signal from the input 140 via a decimator 190.

The resulting signal is applied to a control block 160, which generates control signals for adjusting the properties of the filter 144 and the gain device 146. The control signal for the filter 144 is applied through a frequency warping block 162, a smoothing filter 164 and sample-and-hold circuitry 166 to the filter 144. The same control signal is also applied to the block 156, so that the emulation of the filter 144 matches the adaptation of the filter 144 itself.

In an alternative embodiment, the sample-and-hold circuitry 166 is replaced by an interpolation filter.

The control block 160 further generates a control signal for the adaptive gain device 146. In the illustrated embodiment, the gain control signal is output directly to the gain device 146.

Further, the microprocessor 154 may comprise an adaptive gain emulator (not shown in Figure 4), located in between the filter emulator 156 and the adder 158. In this instance, the control block 160 will also output the gain control signal to the adaptive gain emulator.

Similarly to the feedforward case, the fixed filter 180 may be an NR filter, and the adaptive filter 182 may be a high pass filter.

It will be clear to those skilled in the art that the implementation may take one of several hardware or software forms, and the intention of the invention is to cover all these different forms.

Noise cancellation systems according to the present invention may be employed in many devices, as would be appreciated by those skilled in the art. For example, they may be employed in mobile phones, headphones, earphones, headsets, etc.

The skilled person will recognise that the above-described apparatus and methods may be embodied as processor control code, for example on a carrier medium such as a disk, CD- or DVD-ROM, programmed memory such as read only memory (firmware), or on a data carrier such as an optical or electrical signal carrier. For many applications, embodiments of the invention will be implemented on a DSP (digital signal processor), ASIC (application specific integrated circuit) or FPGA (field programmable gate array). Thus the code may comprise conventional program code or microcode or, for example code for setting up or controlling an ASIC or FPGA. The code may also comprise code for dynamically configuring re-configurable apparatus such as reprogrammable logic gate arrays. Similarly the code may comprise code for a hardware description language such as Verilog TM or VHDL (very high speed integrated circuit hardware description language). As the skilled person will appreciate, the code may be distributed between a plurality of coupled components in communication with one another. Where appropriate, the embodiments may also be implemented using code running on a field-(re-)programmable analogue array or similar device in order to configure analogue/digital hardware.

It should be noted that the above-mentioned embodiments illustrate rather than limit the invention, and that those skilled in the art will be able to design many alternative embodiments without departing from the scope of the appended claims. The word "comprising" does not exclude the presence of elements or steps other than those listed in a claim, "a" or "an" does not exclude a plurality, and a single processor or other unit may fulfil the functions of several units recited in the claims. Any reference signs in the claims shall not be construed so as to limit their scope.

Claims

1. A noise cancellation system, comprising: an input, for receiving a digital signal; a digital filter, having at least a high pass filter characteristic, for receiving the digital signal and generating a filter output signal; and an amplitude detector, for generating a detection signal based on an amplitude of a representation of said filter output signal, wherein the detection signal is applied to the digital filter to control a cut-off frequency thereof.

2. A noise cancellation system as claimed in claim 1 , further comprising: a source input, for receiving a wanted signal; and an adder for forming a sum of amplitudes of the filter output signal and the wanted signal, wherein the representation of the filter output signal is said sum, and wherein the amplitude detector generates the detection signal based on a comparison of said sum with a threshold.

3. A noise cancellation system as claimed in claim 2, wherein the amplitude detector generates the detection signal based on a comparison of an absolute value of said sum with a threshold.

4. A noise cancellation system as claimed in claim 2, wherein the amplitude detector generates the detection signal based on a comparison of a root mean square value of said sum with a threshold.

5. A noise cancellation system as claimed in claim 1 , further comprising: an emulation of the digital filter, connected to the input to receive the digital signal, and generating an emulation output signal, wherein the amplitude detector generates the detection signal based on an amplitude of the emulation output signal as the representation of the filter output signal.

6. A noise cancellation system as claimed in claim 5, further comprising: an adder, for forming a sum of amplitudes of the emulation output signal and the wanted signal, wherein the amplitude detector generates the detection signal based on a comparison of said sum with a threshold.

7. A noise cancellation system as claimed in claim 6, wherein the amplitude detector generates the detection signal based on a comparison of an absolute value of said sum with a threshold.

8. A noise cancellation system as claimed in claim 6, wherein the amplitude detector generates the detection signal based on a comparison of a root mean square value of said sum with a threshold.

9. A noise cancellation system as claimed in any one of claims 5 to 8, wherein the digital signal has a first sampling rate, and further comprising: a decimator, for producing a decimated input signal having a second sampling rate lower than the first sampling rate, and wherein the decimated input signal is applied to the emulation of the digital filter.

10. A noise cancellation system as claimed in any one of claims 1 to 9, wherein said noise cancellation system is a feedforward system.

11. A noise cancellation system as claimed in any one of claims 1 to 9, wherein said noise cancellation system is a feedback system.

12. An integrated circuit, comprising: a noise cancellation system as claimed in any one of claims 1 to 1 1.

13. A mobile phone, comprising: an integrated circuit as claimed in claim 12.

14. A pair of headphones, comprising: an integrated circuit as claimed in claim 12.

15. A pair of earphones, comprising: an integrated circuit as claimed in claim 12.

16. A headset, comprising: an integrated circuit as claimed in claim 12.

17. A method of controlling a filter for a noise cancellation system, comprising: receiving a digital signal, the digital signal having a first sample rate; filtering the digital signal in a digital filter, to generate a filter output signal; detecting an amplitude of a representation of said filter output signal; generating a detection signal based on said detected amplitude; and applying said detection signal to said digital filter, to control a cut-off frequency thereof.

18. A method as claimed in claim 17, wherein said detecting step further comprises: receiving a wanted signal; and forming a sum of amplitudes of the filter output signal and the wanted signal, wherein the representation of the filter output signal is said sum, and wherein the detection signal is generated based on a comparison of said sum with a threshold.

19. A method as claimed in claim 17, further comprising: emulating said digital filter, to generate an emulation output signal; wherein the representation of the filter output signal is said emulation output signal.

20. A method as claimed in claim 19, further comprising: forming a sum of amplitudes of the emulation output signal and the wanted signal, wherein the detection signal is generated based on a comparison of said sum with a threshold.

21. A method as claimed in claim 19 or 20, wherein the digital signal has a first sampling rate, and further comprising: producing a decimated input signal having a second sampling rate lower than the first sampling rate, wherein the decimated input signal is applied to the emulation of the digital filter.