1. Field of the Invention
The present invention relates generally to personal audio devices such as wireless telephones that include adaptive noise cancellation (ANC), and more specifically, to a filter architecture for implementing ANC in a personal audio device.
2. Background of the Invention
Wireless telephones, such as mobile/cellular telephones, cordless telephones, and other consumer audio devices, such as mp3 players, are in widespread use. Performance of such devices with respect to intelligibility can be improved by providing noise canceling using a microphone to measure ambient acoustic events and then using signal processing to insert an anti-noise signal into the output of the device to cancel the ambient acoustic events.
The acoustic environment around personal audio devices such as wireless telephones provides a challenge for the implementation of ANC. In particular, conditions such as nearby voice activity, wind, mechanical noise on the device housing or unstable operation of the ANC system typically requires reset of the adaptive filter that generates the noise-canceling (anti-noise) signal. Since resetting the adaptive results in no noise canceling until the adaptive filter re-adapts, any time an event occurs that disrupts the operation of the ANC system, cancellation of ambient noise is disrupted, as well.
Therefore, it would be desirable to provide a personal audio device, including a wireless telephone, that provides noise cancellation that provides adequate performance under dynamically changing operating conditions. It would further be desirable to provide a mechanism for resetting an ANC system that does not cause the total loss of noise canceling while the ANC system re-adapts.
The above stated objective of providing a personal audio device providing adequate noise cancellation performance in dynamically changing operating conditions and that does not cause total loss of the correct anti-noise signal when the adaptive filter is reset, is accomplished in a personal audio device, a method of operation, and an integrated circuit.
The personal audio device includes a housing, with a transducer mounted on the housing for reproducing an audio signal that includes both source audio for playback to a listener and an anti-noise signal for countering the effects of ambient audio sounds in an acoustic output of the transducer, which may include the integrated circuit to provide adaptive noise-canceling (ANC) functionality. The method is a method of operation of the personal audio device and integrated circuit. A reference microphone is mounted on the housing to provide a reference microphone signal indicative of the ambient audio sounds. The personal audio device further includes an ANC processing circuit within the housing for adaptively generating an anti-noise signal from the reference microphone signal using one or more adaptive filters, such that the anti-noise signal causes substantial cancellation of the ambient audio sounds.
At least one of the one or more adaptive filters is partitioned into a first filter portion having a fixed frequency response that is combined with a variable frequency response of a second filter portion. The partitioned filter may be the adaptive filter that filters the reference microphone signal to generate the anti-noise signal. An error microphone may be included for controlling the adaptation of the anti-noise signal to cancel the ambient audio sounds and for correcting for the electro-acoustic path from the output of the processing circuit through the transducer. A secondary path adaptive filter may be used to generate an error signal from the error microphone signal and the secondary path adaptive filter may be partitioned, alone or in combination with partitioning of the adaptive filter that filters the reference microphone signal to generate the anti-noise signal.
The foregoing and other objectives, features, and advantages of the invention will be apparent from the following, more particular, description of the preferred embodiment of the invention, as illustrated in the accompanying drawings.
The present invention encompasses noise canceling techniques and circuits that can be implemented in a personal audio device, such as a wireless telephone. The personal audio device includes an adaptive noise canceling (ANC) circuit that measures the ambient acoustic environment and generates an anti-noise signal that is injected in the speaker (or other transducer) output to cancel ambient acoustic events. A reference microphone is provided to measure the ambient acoustic environment and an error microphone may be included for controlling the adaptation of the anti-noise signal to cancel the ambient audio sounds and for correcting for the electro-acoustic path from the output of the processing circuit through the transducer. Under certain operating conditions, e.g., when the ambient environment is one that the ANC circuit cannot adapt to, one that overloads the reference microphone, or causes the ANC circuit to operate improperly or in an unstable/chaotic manner, the adaptive filter(s) implementing the ANC circuit must generally be reset. The present invention uses one or more partitioned filters having a fixed frequency response portion and a variable frequency response portion to implement the adaptive filters that control generation of the anti-noise signal. When the response of the partitioned filter is reset, the filter response is restored to a nominal response, or another response selected for recovery from the disruptive condition, providing an immediate anti-noise response that, while initially not adapted to the ambient audio condition, provides some degree of noise-cancellation while the ANC circuit re-adapts. Further, the partitioned filter configuration can provide increased stability, since only a portion of the filter adapts, the amount of deviation from a nominal response can be reduced. Leakage can also be introduced to provide a time-dependent restoration of the adaptive filter response to a nominal response, which provides further stability in operation.
Referring now to
Wireless telephone 10 includes adaptive noise canceling (ANC) circuits and features that inject an anti-noise signal into speaker SPKR to improve intelligibility of the distant speech and other audio reproduced by speaker SPKR. A reference microphone R is provided for measuring the ambient acoustic environment and is positioned away from the typical position of a user's mouth, so that the near-end speech is minimized in the signal produced by reference microphone R. A third microphone, error microphone E, is provided in order to further improve the ANC operation by providing a measure of the ambient audio combined with the audio reproduced by speaker SPKR close to ear 5, when wireless telephone 10 is in close proximity to ear 5. Exemplary circuit 14 within wireless telephone 10 includes an audio CODEC integrated circuit 20 that receives the signals from reference microphone R, near speech microphone NS and error microphone E and interfaces with other integrated circuits such as an RF integrated circuit 12 containing the wireless telephone transceiver. In other embodiments of the invention, the circuits and techniques disclosed herein may be incorporated in a single integrated circuit that contains control circuits and other functionality for implementing the entirety of the personal audio device, such as an MP3 player-on-a-chip integrated circuit.
In general, the ANC techniques of the present invention measure ambient acoustic events (as opposed to the output of speaker SPKR and/or the near-end speech) impinging on reference microphone R, and by also measuring the same ambient acoustic events impinging on error microphone E, the ANC processing circuits of illustrated wireless telephone 10 adapt an anti-noise signal generated from the output of reference microphone R to have a characteristic that minimizes the amplitude of the ambient acoustic events at error microphone E. Since acoustic path P(z) extends from reference microphone R to error microphone E, the ANC circuits are essentially estimating acoustic path P(z) combined with removing effects of an electro-acoustic path S(z) that represents the response of the audio output circuits of CODEC IC 20 and the acoustic/electric transfer function of speaker SPKR including the coupling between speaker SPKR and error microphone E in the particular acoustic environment, which is affected by the proximity and structure of ear 5 and other physical objects and human head structures that may be in proximity to wireless telephone 10, when wireless telephone is not firmly pressed to ear 5. While the illustrated wireless telephone 10 includes a two microphone ANC system with a third near speech microphone NS, some aspects of the present invention may be practiced in a system that does not include separate error and reference microphones, or a wireless telephone uses near speech microphone NS to perform the function of the reference microphone R. Also, in personal audio devices designed only for audio playback, near speech microphone NS will generally not be included, and the near-speech signal paths in the circuits described in further detail below can be omitted, without changing the scope of the invention.
Referring now to
Referring now to
Leaky W coefficient control block 31 is leaky in that response WADAPT(z) normalizes to flat or otherwise predetermined response over time when no error input is provided to cause leaky LMS coefficient controller 31 to adapt. A flat response, WADAPT(z)=0, allows response WFIXED(z) to be set to a desired default, i.e., start-up or reset, response so that the total response of fixed filter portion 32A and adaptive filter portion 32B tends toward response WFIXED(z) over time. Providing a leaky response adaptation prevents long-term instabilities that might arise under certain environmental conditions, and in general makes the system more robust against particular sensitivities of the ANC response. An exemplary leakage control equation is given by:
Wk+1=(1−Γ)·Wk+μ·ek·Xk
where μ=2-normalized
The step size implemented by LMS coefficient controller 31 may have a fixed or selectable rate, as well as a fixed or selectable degree of leakage, as mentioned above. If the leakage is set to restore the response of adaptive filter portion 32B to a zero response, then the response of fixed filter portion 32A with respect to the maximum possible response variation of the adaptive filter portion 32B determines the degree to which the leakage can affect the anti-noise signal generation. The response of fixed filter portion 32A may also be made selectable, such that although the response of fixed filter portion 32A is not dynamically adapted as for adaptive filter portion 32B, the response of fixed filter portion 32A may be selected for particular environments, particular devices, particular users or in response to detection of particular audio events. To customize the device, historical values of the combined response of adaptive filter portion 32B and fixed filter portion 32A may be applied as the response to fixed filter portion 32A, at start-up or in response to an audio event, so that adaptive filter portion 32B only needs to adapt to vary the combined response from that of the historic response, which may be selected from among multiple historic values. Similarly, the initial response of the adaptive filter portion 32B may also be selected, alone or in combination with the selection of the initial response of the adaptive filter portion 32B. A coefficient storage 37 is coupled to LMS coefficient controller 31 to record and subsequently select historical and/or predetermined coefficient sets, which may be selected in response to an event detection block 39 detecting an ambient audio event.
In addition to error microphone signal err, the signal compared to the output of filter 35 by W coefficient control block 31 includes an inverted amount of downlink audio signal ds that has been processed by filter response SE(z), of which response SECOPY(z) is a copy. By injecting an inverted amount of downlink audio signal ds, adaptive portion filter 32B is prevented from adapting to the relatively large amount of downlink audio present in error microphone signal err, and by transforming that inverted copy of downlink audio signal ds with the estimate of the response of path S(z), the downlink audio that is removed from error microphone signal err before comparison should match the expected version of downlink audio signal ds reproduced at error microphone signal err, since the electrical and acoustical path of S(z) is the path taken by downlink audio signal ds to arrive at error microphone E. Filter 35 is not an adaptive filter, per se, but has an adjustable response that is tuned to match the response of an adaptive filter 34 that is used to estimate the response of acoustical path S(z), so that the response of filter 35 tracks the adapting of adaptive filter 34.
To implement the above, adaptive filter 34 has coefficients controlled by SE coefficient control block 33, which compares downlink audio signal ds and error microphone signal err after removal of the above-described filtered downlink audio signal ds, that has been filtered by adaptive filter 34 to represent the expected downlink audio delivered to error microphone E, and which is removed from the output of adaptive filter 34 by a combiner 36. SE coefficient control block 33 correlates the actual downlink speech signal ds with the components of downlink audio signal ds that are present in error microphone signal err. Adaptive filter 34 is thereby adapted to generate a signal from downlink audio signal ds, that when subtracted from error microphone signal err, contains the content of error microphone signal err that is not due to downlink audio signal ds.
Referring now to
Referring now to
Referring now to
In the system depicted in
The above arrangement of baseband and oversampled signaling provides for simplified control and reduced power consumed in the adaptive control blocks, such as leaky LMS controllers MA and 54B, while providing the tap flexibility afforded by implementing adaptive filter stages 44A-44B, 55A-55B and filter 51 at the oversampled rates. The remainder of the system of
In accordance with an embodiment of the invention, the output of combiner 46D is also combined with the output of adaptive filter stages 44A-44B that have been processed by a control chain that includes a corresponding hard mute block 45A, 45B for each of the filter stages, a combiner 46A that combines the outputs of hard mute blocks 45A, 45B, a soft mute 47 and then a soft limiter 48 to produce the anti-noise signal that is subtracted by a combiner 46B with the source audio output of combiner 46D. The output of combiner 46B is interpolated up by a factor of two by an interpolator 49 and then reproduced by a sigma-delta DAC 50 operated at the 64× oversampling rate. The output of DAC 50 is provided to amplifier A1, which generates the signal delivered to speaker SPKR.
Each or some of the elements in the system of
While the invention has been particularly shown and described with reference to the preferred embodiments thereof, it will be understood by those skilled in the art that the foregoing and other changes in form, and details may be made therein without departing from the spirit and scope of the invention.
This U.S. Patent Application Claims priority under 35 U.S.C. §119(e) to U.S. Provisional Patent Application Ser. No. 61/493,162 filed on Jun. 3, 2011.
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Number | Date | Country | |
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20120308026 A1 | Dec 2012 | US |
Number | Date | Country | |
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61493162 | Jun 2011 | US |