SPEECH DISTRIBUTION SYSTEM

• US 20080021706A1
• Filed: 08/06/2007
• Published: 01/24/2008
• Est. Priority Date: 12/09/1999
• Status: Abandoned Application

• Current Assignee: Azoteq Pty Ltd.
• Original Assignee: Azoteq Pty Ltd.
• Inventors: Bruwer, Frederick
• Application Number:
  US11/834,195
• Publication Number:
  US 20080021706A1
• Time in Patent Office: Days
• US Class Current: 704/233
• CPC Class Codes: H04R 27/00 Public address systems circuits for preventing acoustic reaction H04R3/02; circuits for distributing signals to loudspeakers H04R3/12; monitoring or testing arrangements for public address systems H04R29/007; amplifiers H03F

Abstract

A method of distributing speech which includes the steps of, at a given location, receiving an audio signal, extracting from the audio signal a signal representing speech originating from or near the location, and distributing an electric signal which is mixed with the extracted speech signal via an audio system to be played over at least one loudspeaker.

Claims

1. A method of distributing speech in the presence of an audio system and using a single microphone (18.1) which includes the steps of:

(a) at a first location receiving a first audio signal (S1a) through a single microphone (18.1), the first audio signal including a second signal related to an output of the audio system, a voice signal representing speech originating from or near the first location, and noise (AS1m),

(b) extracting from the first audio signal a third signal (S1ae¹), representing the voice signal and the noise, by subtracting from the first audio signal an estimation of the second signal, determined by using an electrical output signal (Ase) of the audio system on its own as a reference signal for an adaptive filter, (52), the third signal being extracted directly from the first audio signal and without going through a loudspeaker and microphone first,

(c) amplifying the third signal and then mixing the third signal (S1ae¹) with the electrical output signal (Ase) to produce a combined output signal,

(d) distributing the combined output signal through the audio system to at least one loudspeaker (12.2 to 12.4), said method further including at least one of the following steps;

(i) automatically adjusting the level of amplification of the third signal in step (c) based on at least the level of background noise;

(ii) individually adjusting the level of amplification of the third signal in step (c) for each loudspeaker;

(iii) reducing the third signal in step (c) if it is determined that no speech is present in the first audio signal (S1a), and

(iv) using the combined output(s) as reference input(s) to an adaptive echo cancellation mechanism (54) to reduce acoustic feedback.

2. A method according to claim 1 further comprising the step of directing the amplified third signal to an electrical input of a mobile phone in order to reduce the influence of the sounds from the audio system in the speech signal going to the phone.

3. A method according to claim 1 wherein step (d) comprises the step of distributing the second output signal to a plurality of locations within a motor vehicle, the plurality of locations corresponding to seating positions inside the vehicle.

4. A method according to claim 1 further comprising the step of using the third signal in (c) for speech recognition processing to control at least one of the following:

signal strength of the distributed second output signal;

audio system volume;

CD selection;

track selection;

mobile phone functions;

radio station selection;

wiper functions;

lights;

climatic control;

electronic guidance control; and

entertainment system control;

where said speech recognition processing is initiated with an activation button.

5. A method according to claim 1 further comprising the steps of calculating a transfer function during a set up procedure and storing coefficients of digital filter to be used to extract the third signal in step (b), and then loading the stored coefficients into the filter when at least the filter is activated.

6. A method according to claim 1 whereby a delay is introduced into the extracted voice signal before it is combined with the output from the audio system, based on the physical position of the loudspeaker, to reduce the effect of the difference in the speed at which sound travels and the speed of electric signals.

7. A method according to claim 6 further comprising the steps of recognizing a substantial change in the transfer function and storing a new set of coefficients for future use.

8. A method according to claim 1 further comprising the steps of introducing a delay in the third signal, said delay having a relationship to the distance between the loudspeaker and the microphone, for each individual loudspeaker.

9. Apparatus for distributing speech in the presence of a first audio signal (As) produced by an audio system (10), the apparatus comprising

a receiving device (18.1) for receiving an acoustic signal (S1a, As1m) from at least one of a plurality of locations, the acoustic signal including at least a second signal related to an output signal of the audio system, a voice signal (S1a) representing speech originating at least from or near the one location and noise,

a module (1.1) for extracting from the received acoustic signal a third signal which represents the voice signal and the noise, by subtracting from the received acoustic signal an estimation of the second signal which is calculated using an electrical output signal of the audio system without mixing it with other signals as reference signal to an adaptive filter and the received acoustic signal, the third signal being extracted directly from the first audio signal and without going through a loudspeaker and microphone first, and the apparatus further comprising at least one of the following configurations

(i) a distribution unit (14, 16.2 to 16.4) for distributing a combined output signal, which includes the extracted voice signal and the first audio signal, to at least some of the locations, the third signal being amplified to a level that is at least influenced by the level of background noise,

(ii) a distribution unit (14, 16.2 to 16.4) for distributing a combined output signal, which includes the extracted voice signal and the first audio signal, to an input to a mobile phone,

(iii) at least one adaptive filter, with coefficients for the adaptive filter(s) determined during an installation or setup procedure and then stored in memory for use when the filter is activated,

(iv) an adaptive echo cancellation filter (54) receiving the electrical signals going to the loudspeakers for reducing acoustic feedback.

10. Apparatus according to claim 9 wherein the distribution unit includes means for distributing the combined output signal(s) to each of the plurality of locations but with a reduced output level to the at least one location from which the acoustic signal was received.

11. Apparatus according to claim 9 wherein the receiving device comprises a microphone which is one of a plurality of microphones each of which is associated with one of a related plurality of locations.

12. Apparatus according to claim 9 wherein the module comprises at least a second adaptive filter for echo cancellation of the amplified signals using the electrical signal(s) going to the loudspeaker(s) as reference inputs.

13. Apparatus according to claim 12 further comprising a central unit coupled to the first filter that receives the acoustic signal over conductive lines which are used for distributing said first audio signal.

14. Apparatus according to claim 12 wherein the module further comprises a white noise source from which white noise is added to the audio system before being output to loudspeakers, the first filter being responsive to the white noise to build a desired transfer function.

15. Apparatus according to claim 12 wherein the first filter comprises a digital filter and the module comprises a memory to store calculated coefficients of the digital filter, the coefficients being loaded into the first filter at start up.

16. Apparatus according to claim 9 further comprising a processor for controlling the signal strength of the extracted signal in a manner which is dependent on the level of the audio system output and background noise.

17. Apparatus according to claim 9 further comprising a control unit located at least at one of said plurality of locations to control the strength of the combined output signal distributed to that location.

18. Apparatus according to claim 9 further comprising a motor vehicle audio system which is integrated therein.

19. Apparatus according to claim 9 further comprising a central unit for adjusting the volume of the extracted voice signal that is mixed with the audio system output for distribution to the various locations.

20. Apparatus according to claim 9 further comprising delay means for compensating for differences in transmissions times of electrically distributed signals and acoustically propagated signals to the plurality of locations.

21. Apparatus according to claim 9 further comprising a single microphone for mainly distributing enhanced voice from a front to a rear of a vehicle.

22. Apparatus according to claim 9 further comprising a duplication of some filters to better estimate the signal to be removed in a stereo system by feeding each stereo signal without being mixed with other signals, as a reference signal into the adaptive filters.

23. Apparatus according to claim 9 wherein the combined output signal is varied in strength per output location, thereby broadcasting the combined output signal at different amplitudes to the various output locations.

24. Apparatus according to claim 9 whereby only signals representing voice and noise from a specific location are used as an input to a phone.

25. Apparatus according to claim 24 whereby phone communication is enhanced by using adaptive filter(s) that effectively reduce the contribution of the audio system output to the signal output of the microphone feeding into the phone, said adaptive filter(s) use as reference the electrical output of the audio system.

26. A method of distributing speech in the presence of an audio system which comprises the steps of:

(a) receiving at least one first location a first audio signal (S1a) through a microphone (18.1, 18.2, 18.3, 18.4), the first audio signal including a second signal related to an output of said audio system, a voice signal representing speech origination from or near at least one of the first locations and background noise (AS1m),

(b) extracting from the first audio signal (S1a) a third signal (S1ae¹), representing the voice signal and the noise, by subtracting from the first audio signal an estimation of the second signal, determined by using an electrical output signal (Ase) of the audio system as a reference signal for an adaptive filter, (52) the third signal being extracted directly from the first audio signal and without going through a loudspeaker and microphone first,

(c) mixing the third signal (S1ae¹) with the electrical output signal (Ase) to produce a combined output signal,

(d) distributing the combined output signal through the audio system to at least one loudspeaker (12.2 to 12.4), said method further including at least one of the following steps;

(i) monitoring the background noise and automatically adjusting the amplification of the speech signal to reflect variations in the level of the background noise,

(ii) applying delays, corresponding to the distance between the microphone(s) and the loudspeaker(s) based on the speed of sound, to the distribution of the third signal,

(iii) routing the third signal extracted from a first audio signal associated with a specific location to a vehicle based mobile phone input and routing the output from the mobile phone to the loudspeaker of the same specific location, and

(iv) reducing acoustic feedback through adaptive echo cancellation (54) based on the electrical signals going to the loudspeakers being used as an input to an adaptive echo canceller.

27. A method according to claim 26 further comprising the steps of receiving a respective first audio signal from each of the locations, and distributing respective combined output signals which correspond to each third signal extracted for the various locations, respectively to multiple loudspeakers.

28. A method according to claim 26 further comprising the steps of filtering each respective first audio signal received from each of the locations to form a filtered signal, and shifting the filtered signal in frequency to form a shifted signal that can be transmitted to a central unit on conductive lines which are also used for the transmission of audio signals to said at least one loudspeaker.

29. A method according to claim 26 wherein the third signal from a specific location can be used as input to a system for recognizing voice commands.

30. A method according to claim 27 further comprising the step of adjusting the strength of the third signal, for each of the locations before it is mixed into the audio system output in step (c), individually at each location.

31. A method according to claim 26 further comprising the steps of calculating the transfer function during a set up procedure and storing the coefficients of a digital filter which is used to extract the third signal in step (b), and then loading the stored coefficients into the filter when at least the filter is activated.

32. A method according to claim 26 further comprising the step of analyzing at least one signal received over the microphone to determine the presence of speech and squelching the third signal of step (c) when speech is absent.

33. A method according to claim 26 whereby a delay is introduced into the extracted voice signal before it is combined with the output of the audio system, based on the physical position of the loudspeaker, to reduce the effect of the difference in the speed at which sound travels and the speed of electric signals.