Multi-channel spatialization system for audio signals

• US 5,438,623 A
• Filed: 10/04/1993
• Issued: 08/01/1995
• Est. Priority Date: 10/04/1993
• Status: Expired due to Fees

• Current Assignee: Administrator of The National Aeronautics and Space Administration
• Original Assignee: United States Administrator of National Aeronautics and Space Administration
• Inventors: Begault, Durand R.
• Primary Examiner(s): BRINICH, STEPHEN M
• Application Number:
  US08/130,948
• Time in Patent Office: 666 Days
• Field of Search: 381/1, 381/17, 381/25, 381/24, 381/26, 381/63
• US Class Current: 381/17
• CPC Class Codes: H04S 1/005 For headphones
• CPC Class Codes: H04S 2420/01 Enhancing the perception of the sound image or of the spatial distribution using head related transfer functions [HRTF's] or equivalents thereof, e.g. interaural time difference [ITD] or interaural level difference [ILD]

Abstract

Synthetic head related transfer functions (HRTFs) for imposing reprogrammable spatial cues to a plurality of audio input signals included, for example, in multiple narrow-band audio communications signals received simultaneously are generated and stored in interchangeable programmable read only memories (PROMs) which store both head related transfer function impulse response data and source positional information for a plurality of desired virtual source locations. The analog inputs of the audio signals are filtered and converted to digital signals from which synthetic head related transfer functions are generated in the form of linear phase finite impulse response filters. The outputs of the impulse response filters are subsequently reconverted to analog signals, filtered, mixed and fed to a pair of headphones.

Claims

1. A three dimensional audio display system for imposing selectively changeable spatial cues to a plurality of audio signals, comprising:

a respective plurality of parallel audio signal paths for translating said plurality of audio signals and wherein each signal path includes,

first filter means having a predetermined filter characteristic and being responsive to one audio signal of said plurality of audio signals,

means coupled to said first filter means for converting said one audio signal to a digital audio signal,

selectively changeable digital storage means coupled to said converting means and generating first and second digital audio signals in two discrete signal channels from said digital audio signal, each said channel further including means for storing time delay data and means for storing a set of filter coefficients derived from an arbitrary head related transfer function and implementing a synthetic head related transfer function in the form of a linear phase finite impulse response filter which operates to impose spatial cues to said first and second digital audio signals for a predetermined spatial location relative to a listener,

means coupled to said digital storage means for converting said first and second digital audio signals to first and second analog audio signals,

second filter means having a predetermined filter characteristic coupled to said converting means for filtering said first and second analog audio signals;

first and second circuit means coupled to said second filter means for combining respective first and second analog audio signals and generating therefrom first and second composite first and second audio signals; and

transducer means coupled to said first and second composite audio signals for generating a plurality of audio output signals which appear to emanate from selectively predetermined different spatial locations.

2. An apparatus according to claim 1 wherein said storage means comprises an interchangeable programmable read only memory programmed with time delay difference information regarding the difference in time delays for sound to reach the left and right ears of said listener for a preselected spatial location and a set of filter coefficients used to implement finite impulse response filtering over a predetermined audio frequency range.

3. A system according to claim 2 and additionally including a digital signal processing chip coupled to said memory for accessing said interchangeable programmable read only memory.

4. A system according to claim 1 wherein said first and second filter means comprise lowpass filter means having predetermined stopband frequencies.

5. A system according to claim 2 wherein said filter characteristic comprises a lowpass filter characteristic having a stopband frequency set to a predetermined maximum usable frequency.

6. A system according to claim 5 wherein the stopband frequency is set substantially at or below one half the Nyquist rate.

7. A system according to claim 1 wherein said set of filter coefficients result from a filter design procedure for reducing the number of coefficients from an original set of coefficients and where a filter error is placed in a region below the Nyquist rate F_c N but above a predetermined maximum frequency of interest F_c J.

8. A system according to claim 7 wherein said set of filter coefficients have a maximum weighting value for a predetermined low frequency range, an intermediate weighting value lower than said maximum value for a predetermined intermediate frequency range extending up to F_c J and a minimum weighting value for said predetermined upper frequency range extending up to F_c N.

9. A system according to claim 1 wherein said audio signals comprise relatively narrow band audio signals.

10. A system according to claim 1 wherein both said first and second circuit means for combining respective first and second analog audio signals comprise left and right summing networks.

11. A system according to claim 8 and additionally including amplifier means coupled to said left and right summing networks.

12. A system according to claim 9 and wherein said transducer means comprises a pair of headphones.

13. A method for producing a three dimensional audio display imposing selectively changeable spatial cues to a plurality of audio signals, comprising the steps of:

feeding a plurality of analog audio signals outputted from a respective plurality of relatively narrow band audio signals coupled to a respective plurality of parallel signal paths;

lowpass filtering said plurality of analog audio signals;

converting said plurality of analog audio signals to digital audio signals;

converting each of said digital audio signals to first and second digital audio channel signals;

selectively delaying and filtering said first and second digital channel signals by feeding said digital audio channel signals to respective interchangeable circuit means, said circuit means implementing a predetermined time delay and a linear phase finite impulse filter response derived from a synthetic head related transfer function, thereby imposing spatial cues to said first and second digital audio channel signals for a desired spatial location relative to a listener;

converting said digital audio channel signals to first and second analog audio channel signals;

lowpass filtering said first and second analog audio channel signals;

combining respective first and second analog audio channel signals and generating first and second composite first and second audio signals; and

coupling said first and second composite second audio signals to transducer means, said transducer means reproducing a plurality of analog audio output signals which appear to emanate from different selectively changeable spatial locations.

14. A method according to claim 13 wherein said interchangeable circuit means comprises a PROM that addresses a digital signal processing chip.

15. A method according to claim 13 wherein said spatial locations include at least 60°

left, 150°

left, 150°

right, and 60°

right of the listener and at 0°

elevation.

16. A method according to claim 13 wherein said step of delaying comprises delaying one of said digital channel signals by a delay corresponding to time difference for a sound emanating from a predetermined spatial position to reach the left and right ears of the listener.

17. A method according to claim 13 wherein said step of filtering comprises applying a set of stored filter coefficients implementing a finite impulse response over a predetermined audio frequency range to each digital channel signal.

18. A method according to claim 17 wherein said filter coefficients are generated by the further steps of:

storing measured head related transfer functions for a left and a right ear of a listener for each predetermined spatial position required as separate files and computer apparatus;

performing a Fast Fourier Transform on each of said files providing an analysis of the magnitude of the head related transfer functions;

supplying a weighting value to each frequency and magnitude derived from the Fast Fourier Transform;

utilizing the weighting values and designing a finite impulse response linear phase filter to generate a reduced number of coefficients where a filter error is placed in a region below a Nyquist rate F_c N but above a predetermined maximum frequency of interest F_c J.

19. A method according to claim 17 wherein said set of filter coefficients have a maximum weighting value for a predetermined to low frequency range, an intermediate weighting value lower than said maximum value for a predetermined intermediate frequency range extending up to F_c J and a minimum weighting value for a predetermined upper frequency range extending up to F_c N.

20. A method according to claim 13 wherein said audio signals comprise audio signals included in an analog output of a plurality of band limited radio communications signals received on mutually different carrier frequencies.