CN107112003B

CN107112003B - Acoustic processor with low latency

Info

Publication number: CN107112003B
Application number: CN201580064466.0A
Authority: CN
Inventors: A.库马尔; T.伊尔冈; X.赵
Original assignee: Avnera Corp
Current assignee: Avnera Corp
Priority date: 2014-09-30
Filing date: 2015-09-30
Publication date: 2021-11-19
Anticipated expiration: 2035-09-30
Also published as: WO2016054186A1; KR20170084054A; CN107112003A; EP3201911A1; TW201626361A; TW201944390A; JP2017530413A; TWI672689B; TWI734176B

Abstract

An audio system with low latency includes a digital audio processor and a sensor input coupled to the processor. The sensor input may be a microphone input. The audio processor operates at the same frequency as the sensor input, which is typically much higher than the audio signal provided to the audio processor. In some aspects, the audio processor operates as a noise cancellation processor and does not include an audio input.

Description

Acoustic processor with low latency

Technical Field

The present disclosure relates to acoustic processing, and more particularly, to reconfigurable acoustic processors capable of real-time or near real-time operation.

Background

In general, the noise present in the listening environment almost always impairs the experience of listening to audio through headphones. For example, in an aircraft cabin, in addition to audio programming, noise from the aircraft generates unwanted sound waves (i.e., noise) that can propagate to the ears of a listener. Other examples include computer and air conditioning noise in an office or home, vehicle and passenger noise in public or private vehicles, or other noisy environments.

To reduce the amount of noise received by the listener, two main ways of noise reduction have been developed: passive noise reduction and active noise cancellation. Passive noise reduction refers to noise reduction caused by placing a physical barrier, typically a headphone or an earplug, between the ear cavity and the noisy external environment. The amount of noise that is reduced depends on the quality of the barrier. In general, noise reducing headphones with greater mass provide higher passive noise reduction. However, large, heavy earphones may be uncomfortable to wear for extended periods of time. For a given headphone, passive noise reduction can better reduce the noise at higher frequencies, while low frequencies may still pass through the passive noise reduction system.

Active noise reduction systems, also known as Active Noise Cancellation (ANC), refer to noise reduction by means of playing an anti-noise signal through an earpiece speaker. The anti-noise signal is generated as an approximation of the negative value of the noise signal that would be in the ear cavity without ANC. Then, when combined with the anti-noise signal, the noise signal is neutralized.

In a typical noise cancellation process, one or more sensors (e.g., a microphone) monitor ambient noise or noise in the earmuffs (earups) of the headset in real time, and the system then generates an anti-noise signal from the ambient or residual noise. The anti-noise signal may be generated in different ways depending on factors such as, for example, the physical shape and size of the ANC system (e.g., headphones, etc.), the frequency response of the transducer and transducer (e.g., speaker), the time delay of the transducer at various frequencies, the sensitivity of the transducer, and the placement of the transducer and transducer. The variation of the above factors between different sensors and transducers (e.g. headphones) and even between two ear cups of the same headphone system means that the optimal filter design for generating anti-noise also varies.

The time delay in processing the anti-noise signal prevents the active noise cancellation system from operating efficiently. For example, digitizing the sensor signal at a rate common in audio processing (such as 44.1KHz or 48 KHz) and processing the signal introduces large time delays. A large time delay is disadvantageous for the acoustic noise cancellation process because the performance of an acoustic processor such as an ANC depends on the ability to detect noise fast enough in time and generate an anti-noise signal to cancel the noise.

Embodiments of the present invention address this and other limitations of the prior art.

Drawings

Fig. 1 is a circuit diagram illustrating a conventional topology of feedforward active noise cancellation.

Fig. 2 is a circuit diagram illustrating a conventional topology of feedback active noise cancellation.

Fig. 3 is a circuit diagram illustrating a conventional topology of combined feedforward and feedback active noise cancellation.

Fig. 4 is a block diagram of an audio system including a reconfigurable acoustic processor according to an embodiment of the present invention.

Fig. 5 is a functional block diagram of the example reconfigurable acoustic processor of fig. 4.

Fig. 6 is a block diagram illustrating the reconfigurable acoustic processor of fig. 4 configured to implement combined feed-forward and feedback active noise cancellation operations.

Fig. 7 is a functional block diagram of components of the example reconfigurable acoustic processor of fig. 4, in accordance with an embodiment of the present invention.

Detailed Description

Embodiments of the present invention relate to digital acoustic processors, such as Reconfigurable Acoustic Processors (RAPs), for use in audio systems using digitized sensor inputs.

There are three main types of Active Noise Cancellation (ANC) that are distinguished based on sensor or microphone placement within the system. In feed-forward ANC, a sensor senses ambient noise but does not appreciably sense the signal produced by a transducer (such as a speaker). Such a system is illustrated in fig. 1. Referring to FIG. 1, a feed-forward ANC system 10 includes a sensor 12 that senses ambient noise but does not monitor the signal directly from a transducer 14. The output from the sensor 12 is filtered in a feedforward filter 16 and the output of the filter is coupled to a feedforward mixer 18, where the filtered signal is mixed with the input audio signal. The filtered signal from filter 16 is an anti-noise signal generated from the output of transducer 12. When the anti-noise signal is mixed with the desired signal in the mixer 18, the output of the transducer 14 (which is the combination of the input signal and the filtered, anti-noise signal mix) has less noise than when the anti-noise signal is not generated.

In feedback ANC, a sensor is placed in a position to sense the total audio signal present in the ear cavity. In other words, the sensor senses the sum of both the ambient noise and the audio played back by the transducer. Such a system is illustrated in fig. 2. Referring to FIG. 2, in the feedback ANC system 20, the sensor 32 directly monitors the output from the transducer 24. In the feedback mixer 30, the output from the sensor 32 is mixed with the audio input signal, and the combined signal is then sent to the feedback filter 34, where it is filtered to produce the anti-noise signal. This anti-noise signal from filter 34 is mixed with the original audio signal in mixer 28, and the combined output is then fed to transducer 24. The feedback ANC system 20 also reduces the noise heard by the listener of the speaker 24.

A combined feed-forward and feedback ANC system uses two or more sensors, with a first position for the sensor being in the feed-forward path as illustrated in fig. 1 and a second position of the sensor being in the feedback path as illustrated in fig. 2. A combined feed-forward and feedback ANC system 40 is illustrated in fig. 3 and includes

sensor locations

42, 52 and one or more transducers at the illustrated location 44. In feedback mixer 50, the signals sensed from the feedback sensor(s) at location 52 are mixed and the combined signal is filtered by feedback filter 54. Similarly, the sensed signal from the feedforward sensor(s) at location 42 is filtered in a feedforward filter 46, and the filtered signal is combined with the input audio signal in a feedforward mixer 48. The output of the transducer(s) at location 44 has reduced noise through filtering and mixing operations.

Embodiments of the present invention use alternative systems to cover many different applications, as opposed to prior systems that use fixed topologies and filters, as described in detail below.

Typical audio processing rates are 44.1kHz or 48kHz, which is a frequency range based on typical human hearing. At these sampling rates, the sampling time period is about 20 μ s. The digitization and filtering in ANC systems always take multiple samples. At these rates, the resulting delay is on the order of hundreds of microseconds. This significantly reduces the performance of ANC, as any delay in processing degrades the generation of the anti-noise signal. This usually appears to limit the maximum noise frequency that can be eliminated.

Fig. 4 is a block diagram of an audio system 100 including a low-latency or ultra-low latency acoustic processor. In some embodiments, the acoustic processor may be reconfigurable and referred to as Reconfigurable Audio Processor (RAP) 150. The audio system in fig. 4 is divided into three general parts-an analog part 102, a digital part 104 that runs at the rate of an analog-to-digital converter (ADC), and a digital part 106 that runs at a standard audio sampling rate, such as 44.1 or 48 KHz. These portions may also be referred to as domains.

The analog portion 102 does not require a clock and typically the signal in that portion is generally a continuous analog signal. For example, the transducer or speaker 110 may produce an analog audio signal, such as from a headphone or other speaker. A sensor such as digital microphone 112 automatically generates a digital output from an analog input signal, while a standard analog sensor such as microphone 114 may be combined with ADC 124 to generate a digital signal from analog sensor 114. A sensor 116, such as a microphone, may be placed in a feedback position and coupled to the ADC 126. The

ADCs

124, 126 may use sigma delta processing, for example. In other embodiments, the

ADCs

124, 126 may be of the Pulse Code Modulation (PCM) or Successive Approximation Register (SAR) type. A

single sensor

112, 114, 116 may be used for multiple purposes, such as sampling ambient noise, while also serving as an input microphone for a telephone, for example. One or more filters 128 may be present to filter the outputs from the

ADCs

124, 126, but are not required in all embodiments.

A Digital Signal Processor (DSP) 130 or other audio source operates in the digital section 106 and at a frequency of standard audio sampling rate. Typically, the operating frequency of the digital portion 106 of the audio system 100 may be 44.1 or 48 KHz.

Conversely, the operating frequency of the digital section 104 may operate at a rate from as low as about 50KHz to about 100MHz, and preferably in a range such as 2-100 MHz. In some embodiments, the digital portion 104 may operate at 50KHz, 96KHz, in the range of hundreds of KHz, at frequencies in the low MHz range (such as 1-6), in the range of tens of MHz (such as 10-20 MHz), up to about 100 MHz. In embodiments of the invention, each of the components of a particular domain operates at the frequency of that domain. For example, referring to fig. 4, the

ADCs

124, 126 operate at the same frequency as the audio processor or RAP 150. This is in stark contrast to previous systems that typically use a decimation filter to down sample the sensor signal before processing in the audio processor.

The interpolator 140 converts the audio signal operating at 48KHz, for example, from the DSP 130 into an audio signal operating at 3MHz or 6MHz as an input signal to the RAP 150. In contrast, the decimator 144, which need not be present in all audio systems 100, converts the signal from the RAP 150, for example at 3 or 6MHz, to the operating frequency of the digital portion 106. Because the RAP 150 processes the signals at the same rate as the signals generated by the sensors or

microphones

112, 114, 116 (whether the sensors are not digital microphones or whether the sensor signals are converted to digital signals by the ADCs 124, 126), the resulting latency of the RAP 150 is very low, e.g., less than 2.5 μ s, and preferably less than 0.5 μ s.

As described in more detail below, the RAP 150 controls in real time the acoustic signals transmitted, for example, from the transducer 110. As described above, the RAP 150 is configured to operate on raw sensor samples from the

microphones

112, 114, and/or 116 without any intermediate processing like decimation filters or other sample rate converters. This allows the microphone signal to be responded to with zero or near zero computational delay in the RAP 150, which enables real-time audio processing algorithms. The effect of using real-time sensor sampling is to eliminate the delay from the decimation filter of previous systems, which in turn greatly increases the responsiveness of the control loop.

The sampling rate of digital portion 104 may vary depending on the sampling rate of digital sensor 112 or ADC 124 coupled to analog sensor 114. There is a linear tradeoff between the sampling rate and the amount of processing that can be handled per sample.

Fig. 5 is a functional block diagram of an example Reconfigurable Acoustic Processor (RAP) 250, which example Reconfigurable Acoustic Processor (RAP) 250 may be an embodiment of RAP 150 of fig. 4. The RAP 250 of FIG. 5 includes six biquad or biquad filter circuits BQ0-BQ6, the functions of which are described below. Biquad filters are well known in electrical processing, particularly audio processing. A biquad filter typically includes 2 zeros and 2 poles. The biquad circuits BQ0-BQ6 each include a cascade of biquad filters. In some embodiments, the circuits BQ0-BQ6 may include 4, 6, 8, 12, or 16 cascaded biquad filters, preferably 8. The biquad filter circuits BQ0-BQ6 are programmable so that their filtered values may vary depending on the desired implementation. They may also be set to a transmit or unit one setting, meaning that they do not significantly affect the signals passing through them.

Connected to each biquad filter circuit BQ0-BQ6 is a gain unit M0-M6, respectively, and an additional gain unit M7, the purpose of which is described below. The gain cells M0-M7 are programmable in that the amount of gain generated between their inputs and outputs is controllable. The outputs of the particular biquad filter circuit BQ0-BQ6 may be controlled by the gain cells M0-M6 to which they are coupled. Setting the gain of any of the gain cells M0-M6 to zero will effectively turn off that particular circuit branch. It is not absolutely necessary to maintain a one-to-one relationship between the biquad filter circuit and the gain cell, but maintaining this relationship provides flexibility in setting the RAP. The RAP 250 of fig. 5 shows a single audio channel. For two or more channels (such as for stereo processing) additional hardware will be used.

By programming specific filter coefficients in the biquad filter circuits BQ0-BQ6 and specific gain values in the gain cells M0-M6, different audio applications (such as audio noise cancellation) may be performed in the RAP 250, as described below.

Also, the inputs coupled to the RAP 250 may include inputs from the digital sensors 212, 214 (which may be microphones, decimators 218, and interpolators 220). Either or both of the

sensor inputs

212, 214 may be created by having an analog microphone coupled to an ADC. Decimator 218 and interpolator 220 operate as described with reference to fig. 4.

In operation, the RAP 250 accepts input from the sensor 212 at biquad filter circuits BQ0 and BQ3, and from the sensor 214 at biquad filter circuits BQ1 and BQ 5. The audio signals are received at biquad filter circuits BQ2 and BQ 6. In some embodiments, the audio signal is not entirely necessary. For example, in a noise canceling headphone for hunters or industry, the audio signal may not be present.

The gain unit M7 may be used as a controllable gain for the processed audio signal before it is finally combined with the unprocessed audio signal from the interpolator 220 in combiner a 2. The gain unit M7 may be controlled to gradually increase its gain so that noise cancellation or other processing may be gradually added to the unprocessed audio signal to cancel pops or other rapid changes in the output audio signal, which may be uncomfortable to the listener.

Adders or combiners a0, a1, and a2 combine the intermediate signal outputs from the biquad filter circuit, as illustrated in fig. 5.

In one embodiment, the RAP 350 operates at 49.152MHz, which is the standard rate for audio processing. The input sample rate is typically 3.072Msps and the filter section can also operate at the same rate.

A simple operational example of the RAP 250 is a simple audio processor that does not use input from either of the

sensors

212, 214. In such an example, the gain unit M7 is set to 0 (i.e., turned off) while the audio signal from the interpolator is filtered by the biquad filter circuit BQ 6. The output signal level of the filtered audio signal is controlled by controlling the gain unit M6, sending the filtered audio signal to the transducer 210, which may be a speaker or other transducer output.

In a more complex example, the RAP 250 may be configured as feed-forward/feedback ANC, which has the same functionality as the feed-forward and feedback ANC circuits illustrated in fig. 3. Fig. 6 illustrates how the RAP 250 is set for such a configuration. In this configuration, gain cells M0 and M5 are set to 0, illustrated in fig. 6 as having an "x," indicating that they do not contribute to the processing. Gain cells M2, M4, M6, and M7 are set to 1. The gain cells M1 and M3 are set to-1, which means their outputs are subtracted. The biquad filter circuits BQ1, BQ2, and BQ6 are set to a pass setting. Referring to fig. 3 and 6, the biquad filter circuit BQ3 has the function of the feedforward filter 46, and the biquad filter circuit BQ4 has the function of the feedback filter 54.

By configuring the RAP 250, and in particular the gain units M0-M7 and biquad filter circuits BQ1-BQ6, the RAP may be configured to perform most any type of audio processing. For example, the RAP 250 may be configured as an ANC processor for an active noise cancelling headset in a feedback, feedforward or combined feedforward feedback configuration. The RAP 250 may be used for active noise cancellation in the handset by using input from the handset microphone and generating audio output for one or more speakers in the handset. The RAP 250 may further enhance the input audio signal while simultaneously performing noise cancellation. The RAP 250 may also be used for ambient sound enhancement by accepting ambient sound at one of the microphone inputs, modifying it through one or more biquad filter circuits, setting the appropriate gain level, and then outputting the modified ambient signal.

In practice, the RAP 250 of fig. 6 or the RAP 150 of fig. 5 includes a function, procedure or operation for modifying an audio signal input. In practice, these functions may be implemented by specifically formed hardware circuitry, as programmed functions operating on a general-purpose or special-purpose processor, such as a Digital Signal Processor (DSP), or in a Field Programmable Gate Array (FPGA) or Programmable Logic Device (PLD). Other variations are also possible.

Fig. 7 is a functional block diagram of components of the example reconfigurable acoustic processor of fig. 4, in accordance with an embodiment of the present invention. In FIG. 7, the RAP 350 includes a biquad engine 310 and a multiplier accumulator 320. Multiplier accumulator 320 implements all of the multipliers and adders in the functional block diagrams of fig. 5 and 6. In one embodiment, there are seven multiply-add operations per sample. The bi-quad engine 310 includes inputs from one or more sensors (such as microphones) and inputs for audio signals to be processed. The biquad engine may also accept inputs from the multiplier accumulator output. The input from the sensor is clocked at the same rate as the biquad engine. In other words, the sensor input may be processed without any decimation or rate reduction. The biquad engine 310 may be sized to operate on 16 biquad filters. A biquad descriptor section (section) 330 contains filter values for implementing a biquad filter circuit, while a biquad state memory 332 is a memory for storing intermediate values during biquad processing. Gain table 322 stores values for gain cells, while feathering control, such as that provided by gain cell M7 of FIG. 5, is provided separately by feathering control 334. The RAP 350 is programmed and configured by writing specific values into the biquad descriptor 330 and the gain table 322, as illustrated in FIG. 7.

By using such programmable techniques, filters can be selected to enhance rather than reduce certain sounds or noise. For example, as described above, instead of the biquad circuit filter parameters (as described above) selected to reduce the ability of the sound sensed by a particular microphone, parameters may be selected that enhance the particular sound. For example, one may use noise canceling headphones in a noisy work environment with various rumble machines, but still wish to be able to talk to colleagues without removing the noise canceling headphones. By using adaptive filter coefficients, when the microphone detects noise in the vocal cords, different parameters may be automatically loaded to the RAP system that enhances the colleagues' voices. Thus, the listener will have noise canceling headphones that adaptively enhance a particular sound. For example, sounds such as speech, audio television signals and traffic may be enhanced. When such sound disappears (e.g., colleagues stop speaking), the standard filter coefficients can be dynamically loaded into the filters of the RAP system again.

Embodiments of the invention may be incorporated into an integrated circuit such as a sound processing circuit or other audio circuit. Further, the integrated circuit may be used in audio devices such as headsets, mobile phones, portable computing devices, bar phones, audio docks (audio docks), loudspeakers, speakers, and the like.

Having described and illustrated the principles of the invention with reference to illustrated embodiments, it will be recognized that the illustrated embodiments can be modified in arrangement and detail without departing from such principles, and can be combined in any desired manner. Also, while the foregoing discussion has focused on particular embodiments, other configurations are contemplated.

In particular, although expressions such as "an embodiment in accordance with the invention" or the like are used herein, these phrases are intended to generally reference embodiment possibilities, and are not intended to limit the invention to particular embodiment configurations. As used herein, these terms may reference the same or different embodiments that are combinable into other embodiments.

Thus, the detailed description and accompanying materials are intended for purposes of illustration only and are not intended to limit the scope of the present invention in any way, given the various permutations of embodiments described herein.

Claims

1. An audio system, comprising:

a sensor that generates an analog sensor signal;

an analog-to-digital converter coupled to the sensor, the analog-to-digital converter generating a digital sensor signal from the analog sensor signal at a sampling rate greater than 50 KHz;

a digital audio processor that performs active noise cancellation in the digital domain at a sampling rate of the analog-to-digital converter without an intermediate sampling rate converter, the digital audio processor having a first input for receiving an input audio signal at the sampling rate, having a second input for receiving the sensor signal at the sampling rate, and having an output.

2. The audio system of claim 1, wherein the sensor is a microphone.

3. The audio system of claim 2, wherein the microphone is part of a system that includes a microphone coupled to the analog-to-digital converter, and the output of the analog-to-digital converter is the digital sensor signal.

4. The audio system of claim 3, wherein the analog-to-digital converter is configured to perform sigma-delta processing.

5. The audio system of claim 3, wherein the analog-to-digital converter is configured to perform successive approximation register processing.

6. The audio system of claim 1, wherein the output is coupled to a transducer.

7. The audio system of claim 6, wherein the transducer is a speaker.

8. The audio system of claim 1, wherein the sampling rate is a 50KHz or faster rate.

9. The audio system of claim 1, wherein the sampling rate is a 96KHz or faster rate.

10. The audio system of claim 1, wherein the sampling rate is a rate of 200KHz or faster.

11. The audio system of claim 1, wherein the sampling rate is a rate of 350KHz or faster.

12. The audio system of claim 1, wherein the sampling rate is a rate of 750KHz or faster.

13. The audio system of claim 1, wherein the sampling rate is a rate of 1MHz or faster.

14. The audio system of claim 1, wherein the sampling rate is a rate of 3MHz or faster.

15. The audio system of claim 1, wherein the sampling rate is a rate of 6MHz or faster.

16. The audio system of claim 1, further comprising:

an interpolator having an input for receiving the digital audio signal at the second rate and converting it to an audio signal having the sampling rate.

17. The audio system of claim 16, wherein the second rate is lower than the sampling rate.

18. The audio system of claim 16, wherein the second rate is 100KHz or slower.

19. The audio system of claim 1, wherein the audio processor comprises:

a plurality of programmable filters;

a plurality of controllable gain stages, at least some of the plurality of controllable gain stages being respectively coupled to at least some of the plurality of programmable filters.

20. A reconfigurable noise cancellation system comprising:

an input for receiving a digital audio signal at an audio sampling rate;

an interpolator to change a sampling rate of the digital audio signal from the audio sampling rate to an analog-to-digital converter sampling rate that is higher than the audio sampling rate, the analog-to-digital converter sampling rate comprising a sampling rate in excess of 50 KHz;

at least one sensor that generates an analog sensor signal;

an analog-to-digital converter coupled to the sensor, the analog-to-digital converter generating a digital sensor signal from the analog sensor signal at the analog-to-digital converter sampling rate; and

a reconfigurable digital audio processor coupled to the interpolator and the sensor, the reconfigurable digital audio processor performing active noise cancellation in the digital domain at the analog-to-digital converter sampling rate without an intermediate sampling rate converter and comprising:

a plurality of programmable filters;

a plurality of controllable gain stages, at least some of the plurality of controllable gain stages being respectively coupled to at least some of the plurality of programmable filters;

an adder configured to combine the outputs of one or more of the plurality of controllable gain stages; and

an audio output coupled to at least one of the adders for conveying an output audio signal modified from the input audio signal.

21. The reconfigurable noise cancellation system of claim 20, further comprising a digital sampling microphone operating at the analog-to-digital converter sampling rate.

22. The reconfigurable noise cancellation system of claim 20, wherein the at least one sensor is an analog microphone.

23. The reconfigurable noise cancellation system of claim 20 in which the analog-to-digital converter is configured to perform sigma-delta processing.

24. The reconfigurable noise cancellation system of claim 20 in which the analog-to-digital converter is configured to perform successive approximation register processing.

25. The reconfigurable noise cancellation system of claim 20 in which the programmable filter is structured to be programmed during operation of the noise cancellation system.

26. The reconfigurable noise cancellation system of claim 20 wherein at least some of the plurality of controllable gain stages are structured to be updated during operation of the noise cancellation system.

27. A method of operating an audio system, comprising:

operating the digital audio processor at an analog-to-digital converter sampling rate of 50KHz or higher;

receiving a digital input audio signal having the analog-to-digital converter sample rate at the digital audio processor,

receiving, at the digital audio processor via an analog-to-digital converter without an intermediate sample rate converter, a digital sensor signal having the analog-to-digital converter sample rate;

performing active noise cancellation on the digital input audio signal in the digital audio processor in the digital domain while running at the analog-to-digital converter sampling rate by combining the digital input audio signal with a signal derived from the digital sensor signal; and

the processed digital input audio signal is output at an output.

28. The method of operating an audio system of claim 27 in which receiving a digital sensor signal having the analog-to-digital converter sampling rate includes receiving a microphone signal at the analog-to-digital converter sampling rate.

29. The method of operating an audio system of claim 27 in which outputting the processed digital input audio signal at an output comprises outputting the digital input audio signal to a transducer.

30. The method of operating an audio system of claim 27 in which the analog-to-digital converter sampling rate is a 96KHz or faster rate.

31. The method of operating an audio system of claim 27 in which the analog-to-digital converter sampling rate is a rate of 1MHz or faster.

32. The method of operating an audio system of claim 27 in which the analog-to-digital converter sampling rate is a rate of 3MHz or faster.

33. The method of operating an audio system of claim 27 in which the analog-to-digital converter sampling rate is a rate of 6MHz or faster.

34. The method of operating an audio system of claim 27, further comprising:

receiving a digital audio signal at an audio sampling rate;

converting the digital audio signal from the audio sampling rate to the analog-to-digital converter sampling rate; and

sending the digital audio signal to the digital audio processor at the analog-to-digital converter sampling rate.

35. The method of operating an audio system of claim 34 in which the audio sampling rate is lower than the analog-to-digital converter sampling rate.

36. A method of operating an audio system according to claim 34 in which the audio sampling rate is 100KHz or slower.

37. A method of operating a reconfigurable noise cancellation processor, comprising:

receiving an audio signal at an analog-to-digital converter sampling frequency through an audio input;

receiving, without an intermediate sample rate converter, one or more sensor signals of a monitored environment through one or more sensor inputs via an analog-to-digital converter at the analog-to-digital converter sampling frequency, the analog-to-digital converter sampling frequency comprising a sampling rate in excess of 50 KHz;

configuring filter parameter sections of a plurality of programmable filters in the reconfigurable noise cancellation processor;

configuring a plurality of controllable gain stages in the reconfigurable noise cancellation processor, at least some of the plurality of controllable gain stages being respectively coupled to at least some of the plurality of programmable filters; and

performing active noise cancellation at the analog-to-digital converter sampling frequency by mixing selected outputs of the plurality of controllable gain stages with the audio signal to produce a modified audio signal output.

38. The method of operating a reconfigurable noise cancellation processor according to claim 37, further comprising:

modifying the filter parameter sections of the plurality of programmable filters during operation of the reconfigurable noise cancellation processor.