SYSTEMS AND METHODS FOR AUDIO PROCESSING

• US 20070061026A1
• Filed: 09/13/2006
• Published: 03/15/2007
• Est. Priority Date: 09/13/2005
• Status: Active Grant

• Current Assignee: DTS, Inc. (Adeia Inc.)
• Original Assignee: SRS Labs Incorporated (Adeia Inc.)
• Inventors: Wang, Wen
• Granted Patent: US 8,027,477 B2
• Time in Patent Office: Days
• US Class Current: 700/94
• CPC Class Codes: H04R 25/407 Circuits for combining signals of a plurality of transducers
• CPC Class Codes: H04S 1/002 Non-adaptive circuits, e.g. manually adjustable or static, for enhancing the sound image or the spatial distribution control circuits for electronic adaptation of the sound field H04S7/30
• CPC Class Codes: H04S 1/005 For headphones
• CPC Class Codes: H04S 1/007 in which the audio signals are in digital form data reduction aspects thereof based on psychoacoustics G10L19/02

Abstract

Systems and methods for audio signal processing are disclosed, where a discrete number of simple digital filters are generated for particular portions of an audio frequency range. Studies have shown that certain frequency ranges are particularly important for human ears'"'"' location-discriminating capability, while other ranges are generally ignored. Head-Related Transfer Functions (HRTFs) are examples response functions that characterize how ears perceive sound positioned at different locations. By selecting one or more “location-critical” portions of such response functions, one can construct simple filters that can be used to simulate hearing where location-discriminating capability is substantially maintained. Because the filters can be simple, they can be implemented in devices having limited computing power and resources to provide location-discrimination responses that form the basis for many desirable audio effects.

Claims

1. A method for processing digital audio signals, comprising:

receiving one or more digital signals, each of said one or more digital signals having information about spatial position of a sound source relative to a listener;

selecting one or more digital filters, each of said one or more digital filters being formed from a particular range of a hearing response function; and

applying said one or more filters to said one or more digital signals so as to yield corresponding one or more filtered signals, each of said one or more filtered signals having a simulated effect of said hearing response function applied to said sound source.

2. The method of claim 1, wherein said one or more digital signals comprise left and right digital signals to be output to left and right speakers.

3. The method of claim 2, wherein said left and right digital signals are adjusted for interaural time difference (ITD) based on said spatial position of said sound source relative to said listener.

4. The method of claim 3, wherein said ITD adjustment comprises:

receiving a mono input signal having information about said spatial position of said sound source;

determining a time difference value based on said spatial information; and

generating left and right signals by introducing said time difference value to said mono input signal.

5. The method of claim 7, wherein said time difference value comprises a quantity that is proportional to absolute value of sinθ

cosφ

, where θ

represents an azimuthal angle of said sound source relative to the front of said listener, and φ

represents an elevation angle of said sound source relative to a horizontal plane defined by said listener'"'"'s ears and the front direction.

6. The method of claim 4, wherein said determination of time difference value is performed when said spatial position of said sound source changes.

7. The method of claim 6, further comprising performing a crossfade transition of said time difference value between the previous value and the current value.

8. The method of claim 7, wherein said crossfade transition comprises changing the time difference value for use in said generation of left and right signals from the previous value to the current value during a plurality of processing cycles.

9. The method of claim 1, wherein said one or more filtered signals comprise left and right filtered signals to be output to left and right speakers.

10. The method of claim 9, further comprising adjusting each of said left and right filtered signals for interaural intensity difference (IID) to account for any intensity differences that may exist and not accounted for by said application of one or more filters.

11. The method of claim 10, wherein said adjustment of said left and right filtered signals for IID comprises:

determining whether said sound source is positioned at left or right relative to said listener;

assigning as a weaker signal the left or right filtered signal that is on the opposite side as the sound source;

assigning as a stronger signal the other of the left or right filtered signal;

adjusting said weaker signal by a first compensation; and

adjusting said stronger signal by a second compensation.

12. The method of claim 11, wherein said first compensation comprises a compensation value that is proportional to cosθ

, where θ

represents an azimuthal angle of said sound source relative to the front of said listener.

13. The method of claim 11, wherein said second compensation comprises a compensation value that is proportional to sinθ

, where θ

represents an azimuthal angle of said sound source relative to the front of said listener.

14. The method of claim 11, wherein said adjustment of said left and right filtered signals for IID is performed when new one or more digital filters are applied to said left and right filtered signals due to selected movements of said sound source.

15. The method of claim 14, further comprising performing a crossfade transition of said first and second compensation values between the previous values and the current values.

16. The method of claim 15, wherein said crossfade transition comprises changing the first and second compensation values during a plurality of processing cycles.

17. The method of claim 1, further comprising performing at least one of the following processing steps either before said receiving of said one or more digital signals or after said applying of said one or more filters:

sample rate conversion, Doppler adjustment for sound source velocity, distance adjustment to account for distance of said sound source to said listener, orientation adjustment to account for orientation of said listener'"'"'s head relative to said sound source, or reverberation adjustment.

18. The method of claim 1, wherein said application of said one or more digital filters to said one or more digital signals simulates an effect of motion of said sound source about said listener.

19. The method of claim 1, wherein said application of said one or more digital filters to said one or more digital signals simulates an effect of placing said sound source at a selected location about said listener.

20. The method of claim 19, further comprising simulating effects of one or more additional sound sources to simulate an effect of a plurality of sound sources at selected locations about said listener.

21. The method of claim 19, wherein said one or more digital signals comprise left and right digital signals to be output to left and right speakers and said plurality of sound sources comprise more than two sound sources such that effects of more than two sound sources are simulated with said left and right speakers.

22. The method of claim 21, wherein said plurality of sound sources comprise five sound sources arranged in a manner similar to one of surround sound arrangements, and wherein said left and right speakers are positioned in a headphone, such that surround sound effects are simulated by said left and right filtered signals provided to said headphone.

23. A positional audio engine for processing digital signal representative of a sound from a sound source, comprising:

a filter selection component configured to select one or more digital filters, each of said one or more digital filters being formed from a particular range of a hearing response function, said selection based on spatial position of said sound source relative to a listener;

a filter application component configured to apply said one or more digital filters to one or more digital signals so as to yield corresponding one or more filtered signals, each of said one or more filtered signals having a simulated effect of said hearing response function applied to said sound from said sound source.

24. The audio engine of claim 23, wherein said hearing response function comprises a head-related transfer function (HRTF).

25. The audio engine of claim 24, wherein said particular range comprises a particular range of frequency within said HRTF.

26. The audio engine of claim 25, wherein said particular range of frequency is substantially within or overlaps with a range of frequency that provides a location-discriminating sensitivity to an average human'"'"'s hearing that is greater than an average sensitivity among an audible frequency.

27. The audio engine of claim 25, wherein said particular range of frequency includes or substantially overlaps with a peak structure in said HRTF.

28. The audio engine of claim 27, wherein said peak structure is substantially within or overlaps with a range of frequency between about 2.5 KHz and about 7.5 KHz.

29. The audio engine of claim 27, wherein said peak structure is substantially within or overlaps with a range of frequency between about 8.5 KHz and about 18 KHz.

30. The audio engine of claim 23, wherein said one or more digital filters comprise a plurality of digital filters.

31. The audio engine of claim 30, wherein each of said one or more digital signals is split into the same number of signals as the number of said plurality of digital filters such that said plurality of digital filters are applied in parallel to said plurality of split signals.

32. The audio engine of claim 31, wherein said each of one or more filtered signals is obtained by combining said plurality of split signals filtered by said plurality of digital filters.

33. The audio engine of claim 32, wherein said combining comprises summing of said plurality of split signals.

34. The audio engine of claim 30, wherein said plurality of digital filters comprise first and second digital filters.

35. The audio engine of claim 34, wherein each of said first and second digital filters comprises a filter that yields a response that is substantially maximally flat in a passband portion and rolls off towards substantially zero in a stopband portion of said hearing response function.

36. The audio engine of claim 35, wherein each of said first and second digital filters comprises a Butterworth filter.

37. The audio engine of claim 35, wherein said passband portion for one of said first and second digital filters is defined by a frequency range between about 2.5 KHz and about 7.5 KHz.

38. The audio engine of claim 35, wherein said passband portion for one of said first and second digital filters is defined by a frequency range between about 8.5 KHz and about 18 KHz.

39. The audio engine of claim 23, wherein said selection of said one or more digital filters is based on a finite number of geometric positions about said listener.

40. The audio engine of claim 39, wherein said geometric positions comprise a plurality of hemi-planes, each hemi-plane defined by an edge along a direction between the ears of said listener and by an elevation angle φ

relative to a horizontal plane defined by said ears and the front direction for said listener.

41. The audio engine of claim 40, wherein said plurality of hemi-planes are grouped into one or more front hemi-planes and one or more rear hemi-planes.

42. The audio engine of claim 41, wherein said front hemi-planes comprise hemi-planes at front of said listener and at elevation angles of approximately 0 and +/−

45 degrees, and said rear hemi-planes comprise hemi-planes at rear of said listener and at elevation angles of approximately 0 and +/−

45 degrees.

43. A system for processing digital audio signals, comprising:

an interaural time difference (ITD) component configured to receive a mono input signal and generate left and right ITD-adjusted signals to simulate an arrival time difference of sound arriving at left and right ears of a listener from a sound source, said mono input signal having information about spatial position of said sound source relative said listener;

a positional filter component configured to receive said left and right ITD-adjusted signals, apply one or more digital filters to each of said left and right ITD-adjusted signals to generate left and right filtered digital signals, each of said one or more digital filters based on a particular range of a hearing response function, such that said left and right filtered digital signals simulate said hearing response function; and

an interaural intensity difference (IID) component configured to receive said left and right filtered digital signals and generate left and right IID-adjusted signal to simulate an intensity difference of said sound arriving at said left and right ears.

44. The system of claim 43, further comprising at least one of the following:

a sample rate conversion component, a Doppler adjustment component configured to simulate sound source velocity, a distance adjustment component configured to account for distance of said sound source to said listener, an orientation adjustment component configured to account for orientation of said listener'"'"'s head relative to said sound source, or a reverberation adjustment component to simulate reverberation effect.

45. A system for processing digital audio signals, comprising:

a plurality of signal processing chains, each chain comprising;

an interaural time difference (ITD) component configured to receive a mono input signal and generate left and right ITD-adjusted signals to simulate an arrival time difference of sound arriving at left and right ears of a listener from a sound source, said mono input signal having information about spatial position of said sound source relative said listener;

a positional filter component configured to receive said left and right ITD-adjusted signals, apply one or more digital filters to each of said left and right ITD-adjusted signals to generate left and right filtered digital signals, each of said one or more digital filters based on a particular range of a hearing response function, such that said left and right filtered digital signals simulate said hearing response function; and

an interaural intensity difference (IID) component configured to receive said left and right filtered digital signals and generate left and right IID-adjusted signal to simulate an intensity difference of said sound arriving at said left and right ears.

46. An apparatus, comprising:

a means receiving one or more digital signals;

a means for selecting one or more digital filters based on information about spatial position of a sound source;

a means for applying said one or more filters to said one or more digital signals so as to yield corresponding one or more filtered signals that simulate an effect of a hearing response function.