Pitch detection and intonation correction apparatus and method

• US 5,973,252 A
• Filed: 10/14/1998
• Issued: 10/26/1999
• Est. Priority Date: 10/27/1997
• Status: Expired due to Term

• Current Assignee: Antares Audio Technologies, LLC
• Original Assignee: Auburn Audio Technologies, Inc.
• Inventors: Hildebrand, Harold A.
• Primary Examiner(s): DONELS, JEFFREY
• Application Number:
  US09/172,978
• Time in Patent Office: 377 Days
• Field of Search: 84/603-605, 84/619, 84/645, 84/657
• US Class Current: 84/603
• CPC Class Codes: G10H 1/02 Means for controlling the tone frequencies, e.g. attack or decay; Means for producing special musical effects, e.g. vibratos or glissandos
• CPC Class Codes: G10H 1/366 with means for modifying or correcting the external signal, e.g. pitch correction, reverberation, changing a singer's voice
• CPC Class Codes: G10H 1/44 Tuning means
• CPC Class Codes: G10H 2210/066 for pitch analysis as part of wider processing for musical purposes, e.g. transcription, musical performance evaluation; Pitch recognition, e.g. in polyphonic sounds; Estimation or use of missing fundamental
• CPC Class Codes: G10H 2210/211 Pitch vibrato, i.e. repetitive and smooth variation in pitch, e.g. as obtainable with a whammy bar or tremolo arm on a guitar
• CPC Class Codes: G10H 2210/331 Note pitch correction, i.e. modifying a note pitch or replacing it by the closest one in a given scale
• CPC Class Codes: G10H 2240/311 MIDI transmission G10H2240/056 takes precedence
• CPC Class Codes: G10H 2250/135 Autocorrelation
• CPC Class Codes: G10H 2250/621 Waveform interpolation
• CPC Class Codes: G10H 2250/631 Waveform resampling, i.e. sample rate conversion or sample depth conversion
• CPC Class Codes: G10H 3/125 Extracting or recognising the pitch or fundamental frequency of the picked up signal
• CPC Class Codes: G10H 7/002 using a common processing for different operations or calculations, and a set of microinstructions (programme) to control the sequence thereof

Abstract

A device and method is disclosed to correct intonation errors and generate vibrato in solo instruments and vocal performances in real time. The device determines the pitch of a musical note produced by voice or instrument and shifts the pitch of that note to produce a very high quality, high fidelity output. The device includes a pitch detector that automatically recognizes the pitch of musical notes quickly. The detected pitch is then used as an input to a pitch corrector that converts the pitch of the input to an output with a desired pitch. The corrected musical note is then in tune with the pitch standard. The device and method employ a microprocessor that samples the signal from a musical instrument or voice at regular intervals using an analog-to-digital converter and then utilizes data derived from an auto-correlation function of the waveform to continuously determine the period of the waveform. The period of the waveform is then compared to a desired period or periods (such as found in a scale). The ratio of the waveform period and the desired period is computed to re-sample the waveform. This ratio is smoothed over time to remove instantaneous output pitch changes. The ratio is used to resample the input waveform. The resulting output waveform is processed through a digital-to-analog converter and output through audio interfaces.

Claims

1. A method for processing a music waveform comprising the steps of:

sampling said music waveform at intervals of time to produce a music waveform sequence of numerical representations of the waveform,

space="preserve" listing-type="equation">{x.sub.j } for j=0, . . . i, where

i represents the current sample of the waveform and 0, 1, 2 . . . represent previous time samples of the waveform,

determining the auto-correlation values of said sequence {x_j } at lag values zero and L, and

determining the smallest value of L which minimizes the difference between the auto-correlation at lag zero and the auto-correlation at lag L,

whereby said smallest value of L represents the measured period of said music waveform.

2. The method of claim 1 further comprising the step of

retuning said music waveform sequence by changing its period by an amount equal to the difference between said measured period of said music waveform and the period of a note from a musical scale that is closest to said measured period to produce a retuned music waveform sequence.

3. The method of claim 2 further comprising the step of

retuning said music waveform sequence by changing its period by an amount equal to the difference between said measured period of said music waveform and the period from a MIDI interface to produce a retuned music waveform sequence.

4. The method of claim 1 further comprising the step of

determining said auto-correlation value by E and H functions, ##EQU5## which are determined within two cycles of said waveform.

5. A method for processing a music waveform comprising the steps of,

sampling said music waveform at intervals of time to produce a sequence of musical representations at a sample rate of the waveform, {x_j } for j=0, 1, 2, . . . i, where i represents the current sample at a current sample rate of the waveform and j=1, j=2 . . . represent prior time samples of the waveform,

providing an estimate of lag value L_est of the period of the music waveform,

updating two functions representative of the accumulated energy of the waveform over two periods 2L of the waveform, using

View Dependent Claims (6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 17, 18, 19)

6. The method of claim 5 wherein

said value of L which minimizes the function V=E_i (L)-2H_i (L) is defined as L_min est, and said method further comprising the steps of,

determining three values, V₁, V₂, V₃ of said function about L_min est,

7. The method of claim 5 further comprising the step of

retuning said music waveform sequence by changing its period by an amount equal to the difference between said measured period of said music waveform and the period of a note from a musical scale that is closest to said measured period to produce a retuned music waveform sequence.

8. The method of claim 5 further comprising the step of

retuning said music waveform sequence by changing its period by an amount equal to the difference between said measured period of said music waveform and the period from a MIDI interface to produce a retuned music waveform sequence.

9. The method of claim 8 further comprising the step of

retuning said music waveform sequence by gradually changing said sample rate to a new sample rate which will convert the period L to said period of a note from a musical scale.

10. The method of claim 8 further comprising the step of,

retuning said music waveform sequence by gradually changing said sample rate to a new sample rate which will convert the period L to said period from said MIDI interface.

11. The method of claim 7 further comprising

the step of converting said returned waveform sequence into a retuned analog music signal.

12. The method of claim 8 further comprising

the step of converting said returned waveform sequence into a retuned analog music signal.

13. The method of claim 5 further comprising the step of,

first applying said sequence {x_j } for j=0, 1, 2 . . . i, to an 8 to 1 anti-aliasing filter, and

then downsampling said sequence.

14. The method of claim 5 wherein,

the variable L_est is computed from N to 1 downsampled data.

15. The method of claim 14 wherein,

N is the number 8.

16. The method of claim 5 wherein,

the variable L_est is computed from E_i (L) and H_i (L) over a wide range of the lag variable L using downsampled data.

17. The method of claim 5 wherein,

said sequence is downsampled by a rate of eight to one to produce a downsampled sequence,

providing said lag value estimate L_est of the period of said downsampled sequence, and

wherein periodicity of said waveform is determined by the step of,

varying the parameter L, and

selecting a particular value of L as the period of said waveform which minimizes the relationship,

18. The method of claim 17 further comprising the step of,

determining a second value of L which minimizes said function,

19. The method of claim 5 wherein,

said steps for processing a music waveform to determine the period of said waveform are performed in a programmed digital processor during background processing.

20. Apparatus for processing a music waveform comprising,

means for sampling said music waveform at intervals of time to produce a music waveform sequence of numerical representations of the waveform,

View Dependent Claims (21, 22, 23)

21. The apparatus of claim 20 further comprising,

means for retuning said music waveform sequence by changing its period by an amount equal to the difference between said measured period of said music waveform and the period of a note from a musical scale that is closest to said measured period to produce a returned music waveform sequence.

22. The apparatus of claim 21 further comprising,

means for retuning said music waveform sequence by changing its period by an amount equal to the difference between said measured period of said music waveform and the period from a MIDI interface to produce a retuned music waveform sequence.

23. The apparatus of claim 20 further comprising,

means for determining said auto-correlation value by E and H functions, ##EQU6## which are determined within two cycles of said waveform.

24. Apparatus for processing a music waveform comprising,

means for sampling said music waveform at intervals of time to produce a sequence of musical representations at a sample rate of the waveform, {x_j } for j=0, 1, 2, . . . i, where i represents the current sample at a current sample rate of the waveform and j=1, j=2 . . . represent prior time samples of the waveform,

means for providing an estimate of lag value L_est of the period of the music waveform,

means for updating two functions representative of the accumulated energy of the waveform over two periods 2L of the waveform, using

View Dependent Claims (25, 26, 27, 28, 29, 30, 31, 32, 33, 34, 35, 36, 37, 38)

25. The apparatus of claim 24 wherein,

said value of L which minimizes the function V=E_i (L)-2H_i (L) is defined as L_min est, and said apparatus further includes,

means for determining three values, V₁, V₂, V₃ of said function about L_min est,

26. The apparatus of claim 24 further comprising,

means for retuning said music waveform sequence by changing its period by an amount equal to the difference between said measured period of said music waveform and the period of a note from a musical scale that is closest to said measured period to produce a retuned music waveform sequence.

27. The apparatus of claim 24 further comprising,

means for retuning said music waveform sequence by changing its period by an amount equal to the difference between said measured period of said music waveform and the period from a MIDI interface to produce a retuned music waveform sequence.

28. The apparatus of claim 27 further comprising,

means for retuning said music waveform sequence by gradually changing said sample rate to a new sample rate which will convert the period L to said period of a note from a musical scale.

29. The apparatus of claim 27 further comprising,

means for retuning said music waveform sequence by gradually changing said sample rate to a new sample rate which will convert the period L to said period from said MIDI interface.

30. The apparatus of claim 26 further comprising,

means for converting said returned waveform sequence into a retuned analog music signal.

31. The apparatus of claim 27 further comprising,

means for converting said retuned waveform sequence into a retuned analog music signal.

32. The apparatus of claim 24 further comprising,

means for first applying said sequence {x} for j=0, 1, 2 . . . i, to an 8 to 1 anti-aliasing filter, and

means for subsequently downsampling said sequence.

33. The apparatus of claim 24 wherein, the variable L_est is computed from N to 1 downsampled data.

34. The apparatus of claim 33 wherein,

N is the number 8.

35. The apparatus of claim 24 wherein,

the variable L_est is computed from E_i (L) and H_i (L) over a wide range of the lag variable L using downsampled data.

36. The apparatus of claim 24 further comprising,

means for downsampling said sequence by a rate of eight to one to produce a downsampled sequence.

37. The apparatus of claim 36 further comprising,

means for determining a second value of L which minimizes said function,

38. The apparatus of claim 24 further comprising,

means for processing said music waveform to determine the period of said waveform in a programmed digital processor during background processing.