Hearing assistance devices with echo cancellation

According to various embodiment of a method of operating a hearing instrument, an acoustic signal pathway is provided from a microphone through a signal processor to a hearing instrument (HI) receiver in an acoustic mode of operation. An RF transmit signal pathway is provided from the microphone to an RF transmitter, and an RF receive signal pathway is provided from the RF receiver through the signal processor to the HI receiver in an RF mode of operation. An input signal representative of sound detected by the microphone is adaptively filtered. A first adaptation rate is used for the acoustic mode of operation and a second adaptation rate is used for the RF mode of operation. The second adaptation rate is faster than the first adaptation rate.

TECHNICAL FIELD

This application relates generally to hearing assistance devices, and more particularly to cancelling, removing or diminishing echo in hearing assistance devices.

BACKGROUND

Examples of hearing assistance devices, also referred to herein as hearing instruments, include both prescriptive devices and non-prescriptive devices. Examples of hearing assistance devices include, but are not limited to, hearing aids, headphones, assisted listening devices, and earbuds.

Modern hearing instruments are typically equipped with a feedback cancellation circuit which helps prevent oscillations due to feedback from the receiver output to the microphone input. The feedback canceller is designed to adapt slowly on signals that are time stationary over relatively long intervals compared with speech. However, if the hearing instrument is used in two-way communication system, the feedback canceller does not address echo produced by the acoustic and/or electrical feedback of signals.

SUMMARY

An embodiment of a hearing instrument, comprises a microphone, a hearing instrument (HI) receiver, a signal processor, an RF antenna, an RF receiver and an RF transmitter. The hearing instrument is configured to operate in an acoustic mode and an RF mode. In the acoustic mode, the signal processor processes a signal representative of a sound detected by the microphone into a processed signal, and the HI receiver outputs an audio signal representative of the processed signal. In the RF mode, the RF transmitter transmits, using the antenna, a signal representative of a sound detected by the microphone, the signal processor processes a signal representative of a signal received by the RF receiver using the antenna, and the HI receiver outputs an audio signal representative of the processed signal.

A hearing instrument embodiment comprises a microphone, a hearing instrument (HI) receiver, a signal processor, an RF receiver, an RF transmitter, and a mode specific adjuster adapted to adjust signal pathways when switching between acoustic and RF modes of operation. For the RF mode of operation, the mode specific adjuster provides an RF receive signal path from the RF receiver to the signal processor, and an RF transmit signal path from the microphone to the RF transmitter. For the acoustic mode of operation, the mode specific adjuster provides an acoustic signal path from the microphone to the signal processor.

According to various embodiment of a method of operating a hearing instrument, an acoustic signal pathway is provided from a microphone through a signal processor to a hearing instrument (HI) receiver in an acoustic mode of operation. An RF transmit signal pathway is provided from the microphone to an RF transmitter, and an RF receive signal pathway is provided from the RF receiver through the signal processor to the HI receiver in an RF mode of operation.

DETAILED DESCRIPTION

An example of a hearing instrument is a hearing aid. A hearing aid generally amplifies or processes sound to compensate for poor hearing and is typically worn by a hearing impaired individual. In some instances, the hearing aid adjusts or modifies a frequency response to better match the frequency dependent hearing characteristics of a hearing impaired individual. Individuals may use hearing aids to receive audio data, such as digital audio data and voice messages, which may not be available otherwise for those who have a hearing impairment.

FIG. 1illustrates an environment for a hearing instrument, according to various embodiments. In addition to processing acoustic sounds detected near the wearer of the hearing instrument, the illustrated hearing instrument also is adapted to communicate with other devices, enabling two-way wireless communication. The hearing instrument100includes a microphone101and a receiver (speaker)102. When operating in an acoustic mode, the microphone picks up ambient sounds and converts the sounds into an electrical signal. The receiver converts a processed electrical signal into an audible signal for the wearer of the hearing instrument. Acoustic feedback occurs if the sound produced by the receiver is directly or indirectly coupled to the microphone, as represented in a simplified manner at103. Sustained feedback occurs at certain frequencies where the gain through the forward path is greater than the attenuation through a feedback path. The feedback path alters with changes in the environment. Feedback cancellers may include notch filters or phase-cancellers.

The network may be a single network or a combination of different networks and different types of networks, including but not limited to various combinations of a Local Area Network (LAN), Wireless LAN (WLAN), Wide Area Networks (WAN), Cellular Networks, PSTN networks, and VoIP networks such as the Internet. The wireless communication device105(e.g. a Bluetooth device or a Wireless Audio Controller (WAC) module) provides the interface between the hearing instrument100and consumer devices or the network106. Through this network, by way of example and not limitation, the illustrated hearing instrument100is capable of communicating voice data with, by way of example and not limitation, landline telephone systems107, cell phones108, and computers109. The illustrated computer109is shown connected to the network105via a wireless router and modem110. The computer109may be equipped with a microphone along with its speakers, to enable voice communication similar the voice communication enabled by the microphones and receivers for the phones107and108. In a communication channel between phone107and hearing instrument100, for example, echoes may be caused by a number of factors, including electrical echo in the network devices and acoustic echo for the hearing instrument100and phone107. Thus, a person speaking into phone107over a channel111to a wearer of a hearing instrument100may hear his or her own voice with an unacceptable delayed echo.

The microphone101and receiver102of the hearing instrument100are collocated in the hearing instrument and are subject to an acoustic feedback that should be cancelled out using the hearing instrument. An embodiment of the hearing instrument is operated in a normal acoustic mode and a wireless mode. When operating in the normal acoustic mode, an acoustic signal from the microphone is processed in such a way so as to “enhance” the wearer's ability to hear. This usually involves amplification and equalization, which can produce acoustic feedback via path103. The feedback is canceled using an adaptive filter.

The hearing instrument100operates in the wireless mode to communicate audio through a communication link. The illustrated communication link has a relatively long delay. When the hearing instrument100operates in the wireless mode, the microphone input is decoupled from the output, the wireless input signal is used as the training signal for the adaptive filter. The hearing instrument wearer is limited in this mode to hearing only the wireless input and no microphone input or local sidetone. The user listens to the wireless input in one ear and his or her own acoustic sidetone in the other ear. In the wireless mode the echo cancellation algorithm is modified by changing the adaption coefficient and the adaptation filter such that it can sufficiently cancel the echo for the wireless communication. These modifications transform the acoustic feedback canceller into an effective echo canceller. According to an embodiment, the adaptation rate is set equal to or approximately equal to the sample rate of the incoming wireless samples.

An echo may occur in voice transmissions because of many causes (e.g. line impedance discontinuities or the interaction between microphones and loudspeakers in an environment). These perturbations are annoying and reduce the intelligibility of conversations. Echo in a communication link may be caused by an introduced delay as audio samples are accumulated and packetized for transmission, an introduced delay as packets are transferred over the wireless hearing instrument interface, an introduced delay as received packets are supplied at a consistent rate for conversion back into audio samples, a processing delay within a hearing instrument, and acoustic feedback within the hearing instrument. All of these may contribute to the echo cancellation problem, and add to the complexity of solving this problem. Some sources of echo are an multi-path effect, group delay, and a processing delay over the wireless hearing interface, which can occur in both one-way and two-way communications. Two-way communication applications have more potential sources of echo, which combine to increase the length of the round trip delay. In digital mobile phone systems, for example, voice signals are digitized, compressed and coded within the mobile handset, and then processed at the radio frequency interface of the network. The total delay introduced by the various stages of digital signal processing range from 80 ms to 100 ms, resulting in a total round-trip delay of 160-200 ms for any echo. A delay of this magnitude will make any appreciable echo disruptive to the communication process. For example, a person talking on phone107to the wearer of hearing instrument100may hear his or her own voice with an unacceptable echo over the communication circuit111.

The perceived effect of an echo depends on its amplitude and time delay. In general, echoes with an appreciable amplitude and a delay of more than 1 ms are noticeable. If the round-trip delay is on the order of a few milliseconds, echo gives a telephone call a sense of “liveliness”. However, echoes become increasingly annoying and objectionable with the increasing round-trip delay and amplitude in particular for delays of more than 20 ms. The two echo types differ in cause and effect and offer a challenging problem when both combine on the same call in the same network. In wireless networks, or digital-cellular calls, there is no hybrid on the mobile end of the call. The source of echo is acoustic coupling in the mobile handset, and this ambient echo depends on the environment in which the handset is located. Multiple delays can occur from the speaker's voice rebounding off surfaces at various distances from the handset. By its very nature, the “wireline” end of the call has a “leaky” hybrid interface. As voice signals pass from the four-wire to the two-wire portion of the network, the higher energy level in the four-wire section reflects back on itself within the hybrid, creating echo. Echo cancellation creates a model of the echo path, synthesizes a replica estimate of the echo, and cancels the echo by subtracting the estimated echo from the true echo. This process allows full-duplex speech between the near and distant callers and results in natural, interactive speech.

Two general kinds of echoes in communication systems are the electric echo and the acoustic echo. The electric echo is also referred to as a hybrid echo or line echo. This echo can be found in the public-switched telephone network (PSTN), mobile, and IP phone systems. The electric echo can be generated from both the near end and the far end electric devices.

Acoustic echo may occur when a speaker/receiver and microphone are placed such that the microphone picks up the signal radiated from the speaker/receiver and its reflections. In the case of telecommunications systems, users are annoyed by listening to their own speech delayed by the round trip time of the signal to the receiver, to the microphone and back. To avoid these problems, the attenuation of the acoustic path between the loudspeaker and the microphone must be sufficiently high. This attenuation can be provided by mechanical separation and/or an adaptive filter.

Electric or hybrid echo may occur, by way of example, as voice signals pass through different line impedances in a communication network (e.g. 4 wire to 2 wire). As voice signals pass from the four-wire to the two-wire portion of the network, for example, the higher energy level in the four-wire section reflects back on itself within the hybrid, creating echo. Real hybrid circuits cannot be 100% ideal because of the leakage, and the parasitic or parametric deviations. Therefore, part of the signal takes the wrong path from both the near end hybrid and the far end hybrid and thus becomes echo. In older telephone systems, the echo is 28 ms or less. In the more complex, modern telephone systems, the electric echo will be longer. For example, the electric echo in GSM systems could be up to 80 ms, and the electric echo in IP telephone could be up to 120 ms or even longer. If the total round-trip delay occurs within just a few milliseconds (i.e., within 28 ms), it generates a sense that the call is live by adding sidetone, which makes a positive contribution to the quality of the call. If the total network delay exceeds 36 ms, however, the positive benefits disappear, and intrusive echo results. Because of the non-stationary nature of various communication links, the hybrids cannot be perfectly tuned using hardware adjustments. Thus, a part of the received signal is reflected as echo, known as the electrical echo.

Acoustic echo is generated with handsets and other devices with a microphone and speaker/receiver. The speaker is referred to as a receiver for telephones and hearing instruments. Acoustic echo is produced by voice coupling between the earpiece and microphone. Sound from a speaker/receiver is heard by a listener, as intended. However, this same sound also is picked up by the microphone, both directly and indirectly, after bouncing off the roof, windows, and walls, for example. The result of this reflection is the creation of multi-path echo and multiple harmonics of echo, which, unless eliminated, are transmitted back to the distant end and are heard by the talker as echo. The echo from the undesired remote speech reflected from roof, windows, and walls, etc. could be as long as 200 ms.

Digital processing delays and speech-compression techniques further contribute to echo generation and degraded voice quality in wireless networks. Voice degradation is caused as voice compressing encoding/decoding devices (vocoders) process the voice paths within the handsets and in wireless networks. This results in returned echo signals with highly variable properties. When compounded with inherent digital transmission delays, call quality is greatly diminished for the wireline caller. Delays are encountered as signals are processed through various routes within the networks, including copper wire, fiber optic lines, microwave connections, international gateways, satellite transmission, and mixed technology digital networks where calls are processed across numerous network infrastructures. In digital wireless networks, voice paths are processed at two points in the network within the mobile handset and at the radio frequency (RF) interface of the network. As calls are processed through vocoders in the network, speech processing delays ranging from 80 ms to 100 ms are introduced, resulting in an unacceptable total end-to-end delay of 160 ms to 200 ms. As a result, echo cancellation devices are required within the wireless network to eliminate the hybrid and acoustic echoes in a digital wireless call. To further compound the delay problem, the real data must be allowed to “run ahead” to give the compression algorithm all the data it needs to perform the complex task of compression. Typically the audio is allowed to run ahead by approximate ⅛th of a second.

An echo canceller removes a caller's voice from the returning audio stream without removing the audio coming from the caller. Echo canceller technology is tuned to work correctly under the expected conditions. When echo cancellers don't work right they produce a variety of unwanted side effects (e.g. the caller hears his or her voice echoed or the caller's echo is canceled along with part of the voice of the caller).

The echo canceller employs a digital adaptive filter to set up a model or characterization of the voice signal and echo passing through the echo canceller. As a voice path passes back through the cancellation system, the echo canceller compares the signal and the model to cancel existing echo dynamically. This process may remove most (on the order of 80 to 90 percent or more) of the echo across the network. A non-linear processor (NLP) attenuates the signal of the remaining residual echo below the noise floor.

In echo cancellation, complex algorithmic procedures are used to compute speech models. This involves generating the sum from reflected echoes of the original speech, then subtracting this from any signal the microphone picks up. The result is the purified speech of the person talking. The format of this echo prediction must be learned by the echo canceller in a process known as adaptation. The parameters learned from the adaptation process generate the prediction of the echo signal.

Echo cancellers can be based on finite impulse response (FIR) filters. The echo canceller coefficients may be adapted using variants of the recursive least square error (RLS) or the least mean squared error (LMS) adaptation methods. One example of an algorithm used for adaptation of the coefficients of an echo canceller is the normalized least mean square error (NLMS) method.

FIG. 2illustrates an embodiment of a hearing instrument200adapted to operate in one mode212to cancel echoes in a communication channel and in another mode213to cancel acoustic feedback. In the acoustic mode213, the hearing instrument200uses a microphone201to convert a sound into an input electrical signal, processes the input electrical signal at a gain, outputs the processed electrical signal to the receiver202to produce a sound, and prevents or stops acoustic feedback using an adaptive filter. In the communication mode (e.g. RF mode212), the hearing instrument200uses a microphone201to convert a sound into an input electrical signal, transmits a wireless communication signal representative of the input signal, receives a return wireless communication signal, processes the received wireless communication signal at a gain, outputs the processed signal to the receiver202to produce a sound, and prevents or stops echo in the transmitted wireless communication signal using an adaptive filter.

FIG. 3illustrates an embodiment of an adaptive filter for filtering an input signal of the hearing aid. The illustrated adaptive filter314includes a filter estimator315and a filter316. The filter estimator315has input signals (signal A and signal B), and controls coefficients of the filter314at times based on an adaptation rate. The filter outputs a filter output signal, which is subtracted from the input signal to provide a filter input signal. According to an embodiment, the hearing instrument operating in the acoustic mode uses the filtered input signal and an output signal as the signal inputs, and the hearing aid operating in the RF mode uses an RF transmit signal and a processed RF receive signal as the signal inputs. The adaptation rate for the RF mode is faster than the adaption rate for the acoustic mode. In some embodiments, the adaptation rate for the RF mode is set approximately equal to the sampling rate of the digital sound signal in the RF transmission. The adaptive filter performs an anti-correlation function that prevents an undesired correlation between the filter input signals (signals A and B), which indicates that an undesired echo or undesired feedback is in one of the filter input signals.

FIG. 4illustrates signal flow for a hearing aid embodiment operating in the acoustic mode. The microphone401of the hearing instrument400converts a sound into an input analog signal417, and an analog-to-digital (A/D) converter418converts the input analog signal into a digital input signal419. A summer420receives the digital input signal419and subtracts a filter output signal421from the digital input signal419to provide a filtered input signal422. A digital signal processor (DSP) processes the filtered input signal422providing frequency selective gain and compression at423, and outputs a digital output signal424. A digital-to-analog (D/A) converter425converts the digital output signal424into an analog output signal426, which is received by the receiver402and converted into an acoustic signal by the receiver402. The adaptive filter414receives a first signal input, also referred to as a residue signal (e.g. signal A), from the filtered input signal422and receives a second signal input, also referred to as a training signal (e.g. signal B), from the digital output signal424. Using a slow adaptation rate427, also referred to as a feedback adaptation rate, along with the residue and training signal inputs, the adaptive filter414provides a filter output signal421that is subtracted from the digital input signal419. The adaptive filter414for the acoustic mode of operation provides an anti-correlation function for signals A and B, updating the filter output signal421at the slow adaptation rate427to prevent a correlation between the filtered input signal422and the digital output signal424.

FIG. 5illustrates signal flow for a hearing aid embodiment operating in the RF mode. The microphone501of the hearing instrument500converts a sound into an input analog signal517, and an A/D converter518converts the input analog signal into a digital input signal519. A summer520receives the digital input signal519and subtracts a filter output signal521from the digital input signal519to provide a filtered input signal522. The filtered input signal522is provided as signal528to a transceiver, which use an antenna529to transmit a wireless (e.g. RF) signal representative of the filtered input signal. The transceiver uses the antenna529to receive a wireless (RF) signal, and convert the received signal into a received electrical signal531. The illustrated transceiver provides functions of both a receiver and a transmitter. Some embodiments use an attenuator546for sidetone. The DSP processes the received electrical signal530at a gain at523, and outputs a digital output signal524. A D/A converter525converts the digital output signal524into an analog output signal526, which is received by the receiver502and converted into an acoustic signal by the receiver502. The adaptive filter514receives a first or residue signal input (e.g. signal A) from the filtered input signal522and receives a second or training signal input (e.g. signal B) from the digital output signal524. Using a fast echo adaptation rate527along with the signal inputs, the adaptive filter514provides a filter output signal521that is subtracted from the digital input signal519to provide the residue signal. The adaptive filter514for the RF mode of operation provides an anti-correlation function for signals A and B, updating the filter output signal521at the fast adaptation rate527to prevent a correlation between the filtered input signal522and the received electrical signal531. Thus, for a person speaking on phone107inFIG. 1, represented by the microphone544and receiver545connected via the network543and transceiver542by way of example and not limitation, the adaptive filter514prevents that person's voice that forms part of received electrical signal530from echoing back to the person within filtered input signal522through the acoustic pathway between the receiver502and microphone501. Some embodiments use voice activity detector (VAD)532to enable the adaptive filter514when a voice is detected. The VAD532may monitor signal524or another signal for the presence of a voice. Other embodiments do not include a VAD to enable the adaptive filter514.

FIG. 6illustrates an embodiment of a hearing instrument capable of selectively operating in both the acoustic mode and the RF or echo mode. In the illustrated embodiment, both the acoustic and RF modes use the same hearing instrument components, including the microphone601, the A/D converter618, the summer620, the antenna629, the transceiver, the DSP gain623, the D/A converter625, the receiver602and the adaptive filter614. The illustrated embodiment includes a mode specific adjuster633for making adjustments within the hearing instrument according to the selected mode634(acoustic mode or RF mode). The modes may be switched in a number of ways. For example, some embodiments default to operating in the acoustic mode, and automatically switch to the RF mode if a good RF signal is received by the device. Some embodiments switch only if the RF signal contains a code that matches a code stored in the hearing instrument. Some provide a switch for a wearer of the hearing instrument to manually switch between modes, and some embodiments allow the user to switch modes through another means such as a magnetic switch or a wireless signal from a device, by way of example but not limitation.

In the acoustic mode, sound entering microphone601feeds forward to output processed signal representative of the sound from the receiver602. The microphone601converts a sound into an input analog signal617, and an analog-to-digital (A/D) converter618converts the input analog signal into a digital input signal619. A summer620receives the digital input signal619and subtracts a filter output signal621from the digital input signal619to provide a filtered input signal622. The mode specific adjuster633adjusts the hearing instrument to pass the filtered input signal622to the DSP623. The DSP processes the filtered input signal622at a gain, and outputs a digital output signal624. The D/A converter625converts the digital output signal624into an analog output signal626, which is received by the receiver602and converted into an acoustic signal by the receiver602. The mode specific adjuster633provides the filtered input signal622as a residue signal input (e.g. signal A) to the adaptive filter614, and provides the digital output signal624as a training signal input (e.g. signal B) to the adaptive filter614. The mode specific adjuster633also controls the adaptation rate627for the adaptive filter, providing a slow adaptation rate for the acoustic mode. Using a slow adaptation rate627along with the signal inputs (signals A and B), the adaptive filter614provides a filter output signal621that is subtracted from the digital input signal619. The adaptive filter614for the acoustic mode of operation provides an anti-correlation function for signals A and B, updating the filter output signal621at the slow adaptation rate627to prevent a correlation between the filtered input signal622and the digital output signal624.

In the RF or echo mode, sound representative of a signal received by RF receiver630is output from the receiver602, and the RF transmitter628transmits an RF signal representative of the sound detected by the microphone602. The microphone601converts a sound into an input analog signal617, and an analog-to-digital (A/D) converter618converts the input analog signal into a digital input signal619. A summer620receives the digital input signal619and subtracts a filter output signal621from the digital input signal619to provide a filtered input signal622. The mode specific adjuster633adjusts the hearing instrument to pass the filtered input signal622to the transmitter628, and to pass received signals from the RF receiver630to the DSP. The DSP processes the received signals from the RF receiver630at a gain, and outputs a digital output signal624. The D/A converter625converts the digital output signal624into an analog output signal626, which is received by the receiver602and converted into an acoustic signal by the receiver602. The mode specific adjuster633provides the filtered input signal622as a residue signal input (e.g. signal A) to the adaptive filter614, and provides digital output signal624as the training signal input (e.g. signal B) to the adaptive filter614. The mode specific adjuster633also controls the adaptation rate627for the adaptive filter, providing a fast adaptation rate for the RF mode. Using a fast adaptation rate627along with the residue and training signal inputs (signals A and B), the adaptive filter614provides a filter output signal621that is subtracted from the digital input signal619. The adaptive filter614for the RF mode of operation provides an anti-correlation function for signals A and B, updating the filter output signal621at the fast adaptation rate627to prevent a correlation between the filtered input signal622presented to the RF transmitter628and the received signal from the RF receiver630, which effectively cancels an echo at a far end of a communication channel (e.g. at phone107inFIG. 1).

FIG. 7illustrates an embodiment of a hearing instrument capable of selectively operating in both the acoustic mode and the RF mode.FIG. 7is similar toFIG. 6, illustrating the mode specific adjuster733and the adaptive filter714in more detail. The illustrated mode specific adjuster733includes a signal pathway adjuster735for adjusting or selecting signal pathways appropriate for the acoustic mode or the RF mode. The mode specific adjuster733includes filter input signals selector to control which signals are delivered to the adaptive filter714as the residue and training input signals (signals A and B) for the filter in the acoustic mode and the RF mode. The illustrated mode specific adjuster733includes an adaptation rate selector737which controls the adaptation rate of the filter714to provide a fast adaptation rate for the RF mode and a slow adaptation rate for the acoustic mode. The illustrated adaptive filter714includes a filter estimator738that has an update algorithm that receives the filter input signals A and B, and provides an updated filter output signal721at the adaption rate727.

FIG. 8illustrates signal flow for an embodiment of a hearing instrument capable of selectively operating in both the acoustic mode and the RF mode. The illustrated switches are not intended to identify a specific type of switch, but rather are intended to provide an illustration of the signal pathway, filter input signals, and adaptation rate can be changed when switching between the acoustic and RF modes of operation.

The mode selector834controls the mode in which the hearing instrument operates. The modes may be switched in a number of ways. For example, some embodiments default to operating in the acoustic mode, and automatically switch to the RF mode if a good RF signal is received by the device. Some embodiments switch only if the RF signal contains a code that matches a code stored in the hearing instrument. Some provide a switch for a wearer of the hearing instrument to manually switch between modes, and some embodiments allow the user to switch modes through another means such as a magnetic switch or a wireless signal from a device, by way of example but not limitation. In the embodiment illustrated inFIG. 8, the mode selector834controls switches to provide appropriate signal pathways, filter input signals, and adaptation rate for each of the acoustic and RF modes. The switches SW1, SW2and SW3are in position1for the acoustic mode and are in position2for the RF mode.

In the acoustic mode, sound entering microphone801feeds forward to output processed signal representative of the sound from the receiver802. The microphone801converts a sound into an input analog signal817, and an A/D converter818converts the input analog signal into a digital input signal819. A summer820receives the digital input signal819and subtracts a filter output signal821from the digital input signal819to provide a filtered input signal822. Switches SW1and SW2pass the filtered input signal822, which the DSP processes at a gain at823, and outputs a digital output signal824. The D/A converter825converts the digital output signal824into an analog output signal826, which is received by the receiver802and converted into an acoustic signal by the receiver802. The filtered input signal822is provided as a residue signal input to the adaptive filter814, and the digital output signal824is provided as a training signal input to the adaptive filter814. Switch SW3selects to use the acoustic adaptation coefficient (e.g. slow rate)839for the adaptation rate used in the adaptive filter814. Using the slow adaptation rate839received through control827, along with the residue and training signal inputs (signals A and B), the algorithm within the adaptive filter814provides a filter output signal821that is subtracted from the digital input signal819. The adaptive filter814for the acoustic mode of operation provides an anti-correlation function for signals A and B, updating the filter output signal821at the slow adaptation rate to prevent a correlation between the filtered input signal822and the digital output signal824.

In the RF mode, sound representative of a signal received by the transceiver from the antenna829is output from the hearing instrument receiver802, and the transceiver transmits an RF signal representative of the sound detected by the microphone802. The microphone801converts a sound into an input analog signal817, and an A/D converter818converts the input analog signal into a digital input signal819. A summer820receives the digital input signal819and subtracts a filter output signal821from the digital input signal819to provide a filtered input signal822. The filtered input signal822passes through switch SW1as the transmit signal828and a received signal830from the transceiver are passed through SW2. Some embodiments provide an attenuator846for sidetone. The DSP processes the received signals from the RF receiver830at a gain823, and outputs a digital output signal824. The D/A converter825converts the digital output signal824into an analog output signal826, which is received by the receiver802and converted into an acoustic signal by the receiver802. The filtered input signal822is provided as a residue signal input to the adaptive filter814, and the digital output signal824is provided as a training signal input to the adaptive filter. Switch SW3selects to use the wireless adaptation coefficient or RF adaptation coefficient (e.g. fast rate)840for the adaptation rate used in the adaptive filter814. Using the fast adaptation rate840along with the residue and training signal inputs (signals A and B), the algorithm within the adaptive filter814provides a filter output signal821that is subtracted from the digital input signal819. The adaptive filter814for the RF mode of operation provides an anti-correlation function for signals A and B, updating the filter output signal821at the fast adaptation rate to prevent a correlation between the filtered input signal822and the received signal, which effectively cancels an echo at a far end of a communication channel (e.g. at phone107inFIG. 1). Thus, for a person speaking on phone107inFIG. 1, represented by the microphone844and receiver845connected via the network843and transceiver842by way of example and not limitation, the adaptive filter814prevents that person's voice that forms part of received electrical signal830from echoing back to the person within filtered input signal832through the acoustic pathway between the receiver802and microphone801.

Some embodiment use VAD832to enable the adaptive filter814when a voice is detected. The VAD832may monitor signal831or another signal (e.g. signal824) for the presence of a voice. Other embodiments do not include a VAD to enable the adaptive filter814within the RF mode.

The above detailed description is intended to be illustrative, and not restrictive. The scope of the invention should, therefore, be determined with reference to the appended claims, along with the full scope of equivalents to which such claims are legally entitled.