Apparatus and method for noise reduction for a full-duplex speakerphone or the like

• US 5,410,595 A
• Filed: 11/12/1992
• Issued: 04/25/1995
• Est. Priority Date: 11/12/1992
• Status: Expired due to Term

• Current Assignee: Apple Inc.
• Original Assignee: Motorola, Inc. (Motorola Solutions, Inc.)
• Inventors: Park, Sangil, Funderburk, Dion M.
• Primary Examiner(s): Dwyer, James L.
• Assistant Examiner(s): WEAVER, SCOTT LOUIS
• Application Number:
  US07/975,348
• Time in Patent Office: 894 Days
• Field of Search: 379/388, 379/389, 379/390, 379/406, 379/410, 379/411, 379/202, 379/203, 379/204, 379/205, 379/206, 379/412, 370/110.1, 370/62
• US Class Current: 379/406.08
• CPC Class Codes: H04M 9/08 Two-way loud-speaking telephone systems with means for conditioning the signal, e.g. for suppressing echoes for one or both directions of traffic
• CPC Class Codes: H04M 9/082 using echo cancellers echo cancellers per se H04B3/23

Abstract

The characteristics of a room in which a speakerphone (20) is located are measured by determining a time between a test signal and its first attack, and a number of sample periods between the first attack and a time when the average power in the echo falls below a threshold. The first-attack time determines a pre-filter delay and the number of sample periods determines a tap length for an adaptive echo-canceling filter (62). In a teleconferencing environment, an annoying initialization sequence is avoided by initializing filter coefficients for each microphone (140), and saving the initial filter coefficients generated thereby in a corresponding nonvolatile memory (104). In response to an off-hook signal, the coefficients are retrieved from the nonvolatile memory (104). During operation, the coefficients are constantly updated. If another microphone (141) is enabled, the stored coefficients corresponding to that microphone (141) are dynamically substituted for the present coefficients.

Claims

1. An apparatus for noise reduction for full-duplex communication, comprising:

a summing device having a positive input for receiving a microphone input signal, a negative input, and an output for providing a first output signal;

a delay buffer having an input terminal for receiving a digital receive path signal, and an output terminal for providing a second output signal; and

an adaptive finite impulse response (AFIR) filter having a signal input terminal coupled to said output terminal of said delay buffer, a feedback input terminal coupled to said output terminal of said summing device, and an output terminal coupled to said negative input of said summing device for providing a third output signal;

said delay buffer delaying said digital receive path signal by a first predetermined number of sample periods, said first predetermined number of sample periods equal to a number of sample periods from a test loudspeaker output signal to a first time when a power of said microphone input signal exceeds a first predetermined threshold;

said AFIR filter having a number of coefficients equal to a number of sample periods between said first time and a second time at which an expected power of said microphone input signal drops below a second predetermined threshold;

said AFIR filter having a plurality of contiguous memory locations for storing successive samples of said second output signal, said AFIR filter multiplying said number of coefficients by corresponding ones of said successive samples of said second output signal to obtain said third output signal.

2. The apparatus of claim 1 wherein said number of coefficients of said AFIR filter are updated using a least-mean-squares algorithm.

3. The apparatus of claim 1 wherein said first predetermined threshold equals said second predetermined threshold.

4. An apparatus for noise reduction for full-duplex communication, comprising:

a communications bus;

a nonvolatile memory coupled to said communications bus;

an adaptive finite impulse response (AFIR) filter having an input coupled to said communications bus for receiving initial values of a predetermined number of AFIR filter coefficients, said AFIR filter applying said predetermined number of AFIR filter coefficients to a corresponding predetermined number of samples of an input signal to provide an echo-cancelled output signal, said AFIR filter continuously updating said predetermined number of AFIR filter coefficients while said AFIR filter applies said predetermined number of AFIR filter coefficients using a least-mean-squares algorithm to provide updated values of said predetermined number of AFIR filter coefficients; and

a data processor coupled to said communications bus, for transferring said initial values of said predetermined number of AFIR filter coefficients from said nonvolatile memory to said AFIR filter in response to an off-hook input signal, and for transferring said updated values of said AFIR filter coefficients from said AFIR filter to said nonvolatile memory in response to a deactivation of said off-hook input signal.

5. The apparatus of claim 4 wherein said data processor further provides an off-hook output signal to a telephone transmission line coupler to trigger a telephone line relay after transferring said initial values of said predetermined number of AFIR filter coefficients from said nonvolatile memory to said AFIR filter.

6. An apparatus for noise reduction for full-duplex communication, comprising:

a plurality of microphones;

a plurality of analog-to-digital converter means coupled to corresponding ones of said plurality of microphones and to a communications bus, for sampling audio input signals from said corresponding ones of said plurality of microphones;

a nonvolatile memory coupled to said communications bus including a plurality of nonvolatile memory portions corresponding to each of said plurality of microphones;

an adaptive finite impulse response (AFIR) filter having an input coupled to said communications bus for receiving initial values of a predetermined number of AFIR filter coefficients, said AFIR filter applying said predetermined number of AFIR filter coefficients to a corresponding predetermined number of samples of an input signal to provide an echo-cancelled output signal, said AFIR filter continuously updating said predetermined number of AFIR filter coefficients while said AFIR filter applies said predetermined number of AFIR filter coefficients using a least-mean-squares algorithm to provide updated values of said predetermined number of AFIR filter coefficients; and

a data processor coupled to said communications bus, for transferring said initial values of said predetermined number of AFIR filter coefficients, from a portion of said nonvolatile memory to said AFIR filter in response to an off-hook input signal, wherein said data processor further continuously determines an active one of said plurality of microphones and transfers said initial values of said predetermined number of AFIR filter coefficients from a corresponding one of said plurality of nonvolatile memory portions to said AFIR filter in response to said data processor determining a different active one of said plurality of microphones.

7. A data processing system for full-duplex communication comprising:

a first microphone;

a first analog-to-digital converter (ADC) having an input coupled to said first microphone, and an output coupled to a communication bus;

a second microphone;

a second ADC having an input coupled to said second microphone, and an output coupled to said communication bus;

a first nonvolatile memory coupled to said communication bus and corresponding to said first microphone;

a second nonvolatile memory coupled to said communication bus and corresponding to said second microphone;

an acoustic echo canceller including an adaptive finite impulse response (AFIR) filter, said AFIR filter having an input coupled to said communication bus for receiving initial values of a plurality of coefficients therefrom, said AFIR filter applying said plurality of coefficients to a corresponding number of a plurality of samples of an input signal received from a selected one of said first and second microphones to provide an echo-cancelled output signal, said AFIR filter continuously updating said plurality of coefficients while said selected one of said first and second microphones is active, using a least-mean-squares algorithm to provide updated values of said plurality of coefficients; and

a data processor coupled to said communication bus, for selecting one of said first and second microphones, and for transferring said initial values of said plurality of coefficients from one of said first and second nonvolatile memories corresponding to said selected one of said first and second microphones, to said AFIR filter, in response to said data processor selecting one of said first and second microphones;

wherein said data processor further transfers said updated values of said plurality of coefficients from said AFIR filter to one of said first and second nonvolatile memories corresponding to a deselected one of said first and second microphones.

8. The data processing system of claim 7 wherein each of said first and second nonvolatile memories comprises an electrically erasable programmable read only memory (EEPROM).

9. A method for noise reduction for full-duplex communication, comprising the steps of:

detecting an off-hook indication from a full-duplex speakerphone;

transferring initial values of a plurality of adaptive finite impulse response (AFIR) filter coefficients from a nonvolatile memory to an AFIR filter;

providing an off-hook indication to a telephone line;

filtering an acoustic input signal, received from a microphone of said full-duplex speakerphone, in said AFIR filter;

continuously updating the values of said plurality of AFIR filter coefficients to provide updated values of said plurality of AFIR filter coefficients while performing said step of filtering;

detecting an on-hook indication from the full-duplex speakerphone;

transferring said updated values of said plurality of AFIR filter coefficients from said AFIR filter to said nonvolatile memory; and

providing an on-hook indication to said telephone line.

10. The method of claim 9 wherein the step of updating the values of said plurality of AFIR filter coefficients is performed according to a predetermined algorithm.

11. The method of claim 10 wherein said step of updating the values of said plurality of AFIR filter coefficients further comprises the step of updating said plurality of AFIR filter coefficients according to a least-mean-squares (LMS) algorithm during said step of filtering.

12. A method for noise reduction for full-duplex communication, comprising the steps of:

estimating a power of both a first acoustic input signal received from a first microphone and a second acoustic input signal received from a second microphone;

selecting one of said first and second microphones with a greater of the two estimated powers and deselecting another one of said first and second microphones with a lower of the two estimated powers;

transferring current values of a plurality of adaptive finite impulse response (AFIR) filter coefficients from an AFIR filter to one of first and second nonvolatile memories corresponding to said deselected one of said first and second microphones;

transferring initial values of said plurality of AFIR filter coefficients from one of said first and second nonvolatile memories corresponding to said selected one of said first and second microphones, to said AFIR filter;

filtering the acoustic input signal from said selected one of said first and second microphones in said AFIR filter; and

continuously updating the initial values of said plurality of AFIR filter coefficients according to a predetermined algorithm to provide updated values of the initial values of said plurality of AFIR filter coefficients while said selected one of said first and second microphones remains selected.