Take the 2-minute tour ×
Ask Patents is a question and answer site for people interested in improving and participating in the patent system. It's 100% free, no registration required.

AN OVERBROAD PATENT ON pitch detection - This issued patent from Overtone Labs seeks to patent the idea of...pitch detection for resonance tuning of a musical instrument. 10 minutes of your time can help narrow US patent applications before they become patents. Follow @askpatents on twitter to help.

A new method patent was granted in August of 2013 which is extremely dangerous. The patent is being asserted against a small veteran-owned company which released an app. Other apps have fearfully withdrawn from global app markets as a result of this patent.

QUESTION - Have you seen anything that was published before 11/30/2011 that discusses:

  • Using power spectral analysis to tune musical instruments.

Please review Patent US8502060 The app that is being sued is being forced into cease and desist and black out social media communities 60,000 musicians strong.

Claims 1 & 13 are particularly concerning.

1: A method for resonance tuning, comprising:

  • Receiving a signal in response to a resonance of a structure;
  • Determining a frequency or musical note related to an overtone from the signal;
  • Selecting the frequency or musical note related to the overtone as a filter mode reference frequency or musical note; and
  • Suppressing a display of frequencies or musical notes from a subsequent signal that deviate from the filter mode reference frequency or musical note by a predetermined threshold.

.

13: A method for pitch detection, comprising:

  • Providing one or more power spectrum frequency samples;

  • Selecting a frequency in a frequency band having a largest power spectrum magnitude from the one or more power spectrum frequency samples, the frequency band having an upper frequency limit and a lower frequency limit.


"A schematic view of an embodiment of the Pitch Estimator"

FIG. 16 cited by applicant in Supplemental Examination Support Doc as providing support for Claim 13. (annotation emphasis added)

What is good prior art? Please see our FAQ.

Want to help? Please vote or comment on submissions below. We welcome you to post your own request for prior art on other questionable US Patent Applications.


share|improve this question
3  
I'm not a subject matter expert here but isn't it strange that the applications was filed on 2/15/13 and allowed three months later (5/28/13)? No initial rejection, no changes to the claims. It's not plausible to me that, for example, claim 13 should have been allowed over prior art. I can see why OP posted this prior art request. Any indication in the file history how this could have happened? –  Micah Siegel Dec 11 '13 at 4:57
2  
It's an expedited prosecution where the Patent office relies on a prior art search by applicant where the applicant supplies the PTO with the best art they could find and says why their claims are patentable over the prior art. In this case, they were UNABLE TO FIND PRIOR ART FOR PEAK DETECTION which strains credulity!!! –  Frank-n'Grind Dec 11 '13 at 16:47
3  
Luckily, the prosecution history is public record and available at USPTO.gov under public pair. The most recent Examination Support Document (starting at page 5) is the most fun... tied by the Examiner's reasons for allowance in the Notice of Allowance. HOW IS PEAK DETECTION NOVEL, let alone not obvious??!? –  Frank-n'Grind Dec 11 '13 at 18:51
2  
It's inequitable that there's NO means to respond to a patent suit without a small business paying on the order of $30,000??? Patent attorneys + litigation counsel aren't cheap and neither is instigating a Reexam proceeding (another $6000; micro-entity doesn't apply for 3rd party requesters). –  Frank-n'Grind Dec 11 '13 at 18:55
2  
A change.org petition was started if anyone is interested. I find this a compelling story. tinyurl.com/kyh4gqx –  Robert Tesla III Dec 24 '13 at 19:44
show 15 more comments

44 Answers

up vote 7 down vote accepted

Non-Cited Prior Art Found

I find it amusing that Jesse Aronstein was mentioned via patent US4741242 in the asserted US8502060 patent, yet Aronstein's US4457203 was not. I know that all inventors get a bug in them and never stop inventing. I know if you've ever had a patent, you want more. Why would you cite someone in your patent and not do your full research on them? Even the patent attorney who prepared this patent should have done so.

Let's have a look at US4457203 and US8502060 side by side.

'203 Abstract (year 1982):

ABSTRACT A sound signal automatic detector used in a system with a micro computer and display for automatically detecting an input sound wave, computing from the detected sound wave the fundamental frequency of the sound and displaying its value in a number of different formats. The sound signal detector requires no attention on the part of a musician or other user while it is in operation and comprises a sound signal transducer supplying an amplifier having audio frequency bandpass characteristics compatible with the sound signal frequency spectrum over which sound signals to be analyzed extend. The bandpass characteristics of the amplifier preferably are defined by a high pass filter stage followed by an automatic gain control amplifier that in turn is followed by two stages of low pass filtering. The low pass filter stages supply their output to an alternate positive peak voltage and negative peak voltage detector circuit that functions to derive an output signal which is representative of the fundamental frequency of a input sound wave being analyzed. The output from the automatic detection circuit is supplied to a micro computer which then processes the signal and derives a number of different display formats for use by an instrumentalist, vocalist, or other musician or like person producing the sound for analysis and instruction purposes.

'060 Abstract (year 2013):

Abstract Provided are systems and methods for resonance tuning. A signal is received in response to a resonance of a structure. A frequency or musical note related to an overtone is determined from the signal. The frequency or musical note related to the overtone is selected as a filter mode reference frequency or musical note. A display of frequencies or musical notes from a subsequent signal that deviate from the filter mode reference frequency or musical note by a predetermined threshold is suppressed.

I'm going to assume, since it's done quiet often that it's ok to paraphrase things and call them your own.

'203 Claim 1 (year 1982):

  1. A sound pitch automatic detection circuit comprising:

(a) sound signal transducer means responsive to a sound signal in the form of a note being played or voiced for converting the sound signal to an electrical signal having corresponding audio frequency characteristics and a generally sinusoidally varying waveshape;

(b) amplifier means having audio frequency bandpass characteristics compatible with the sound signal frequency spectrum over which the sound signal extends and for amplifying the electrical signals derived by said transducer means;

(c) alternate positive polarity and negative polarity peak voltage detector means continuously responsive to the output from said amplifier means for detecting the first major positive going peak voltage and the first major negative going peak voltage which exceed respective positive and negative threshold voltage values and occurring in each fundamental period of the generally sinusoidally varying waveshape electric signal; and

(d) output circuit means responsive to the output from said alternate positive polarity and negative polarity peak voltage detector means for deriving an output electric signal representative of the fundamental frequency of the sound signal.

....

This is what I consider a well worded higher quality patent. It does not obscure or over broaden its claims.

'060 Claim 1 (year 2013):

  1. A method for resonance tuning, comprising: receiving a signal in response to a resonance of a structure; determining a frequency or musical note related to an overtone from the signal; selecting the frequency or musical note related to the overtone as a filter mode reference frequency or musical note; and suppressing a display of frequencies or musical notes from a subsequent signal that deviate from the filter mode reference frequency or musical note by a predetermined threshold.

  2. The method of claim 1, further comprising: displaying the frequency or musical note of the overtone as the filter mode reference frequency or musical note.

  3. The method of claim 1, wherein suppressing the frequencies or musical notes from a subsequent signal that deviate from the filter mode reference frequency or musical note by a predetermined threshold includes filtering the frequencies or musical notes of the subsequent signal to reject overtones or a fundamental tone.

  4. The method of claim 1, further comprising providing a bandpass filter centered at the filter mode reference frequency or musical note with a passband bandwidth extending from a predetermined frequency or frequency ratio below the filter mode reference frequency or musical note to a predetermined frequency or frequency ratio above the filter mode reference frequency or musical note, and suppressing frequencies that are outside of the passband bandwidth.

  5. The method of claim 1, wherein suppressing the frequencies from a subsequent signal that deviate from the filter mode reference frequency by a predetermined threshold includes limiting the start and stop frequencies of a frequency band used in a peak selection.

....

I believe Aronstein demonstrated this in his claims

Here's two diagrams from each patent side by side. What concerns me is the '060 diagram has a simple component to accomplish peak selection, and Aronstein actually drew out the operation of his simple component in expanded form. Again, I suppose it's ok to "paraphrase" or simplify by obscuring something and calling it your own?

'203 Figure 1 (Year 1982):

'203 Figure 1 (Year 1982)

'060 Figure 5 (Year 2011/2013):

'060 Figure 5 (Year 2011/2013)

US4457203 (1982) Summary

  • Musical Instrument Tuner - Reads Frequencies
  • Bandpass the target frequency to remove unwanted sound
  • When sound is above a level, a peak is detected
  • Sound above level is estimated
  • estimated pitch is displayed via several formats, to include difference from target.
  • All other pitches are suppressed

US8502060 (2011/2013) Summary

  • Musical Instrument Tuner - Reads Frequencies
  • Bandpass the target frequency to remove unwanted sound
  • When sound is above a level, a peak is detected
  • Sound above level is estimated
  • estimated pitch is displayed via several formats, to include difference from target.
  • All other pitches are suppressed

There's several claims in the US4457203 that were not cited as prior art to the US8502060 patent that will be easy to invalidate 1 if not all claims in the 060. I will revise this answer for more, yet I can't see the need other than for public consumption. I don't see anything novel with the 060 patent considering it's been done for years.

I still find it strikingly odd that Aronstein's '242 patent was cited on the '060 patent but his '203 was not. Coincidence?

share|improve this answer
    
BINGO! Great work! I had that one on my list. David G. Ellis, Steve A. Schoenberg (co-inventors with Jesse Aronstein) have some great patents too! –  Aron Stein Dec 15 '13 at 19:13
    
It's Peanut Butter Jelly Time! –  Tyler Ulrich Dec 15 '13 at 19:14
1  
The more I visualize this, the more I find it to be a silver bullet to the '060 patent. I'm curious to the motives behind not including it in the patent. I smell burnt toast here. Rule 11. –  Robert Tesla III Dec 17 '13 at 14:58
1  
OUCH!!! This is a done deal. –  Audio Sniper Dec 17 '13 at 20:11
    
This is interesting. I reviewed this patent again, and it's a deadlock to me. What is more fascinating is how many times patent examiners have cited this patent in relation to pitch detection devices, which could also be used as prior art to invalidate '060. There's a breeding ground of non-cited patents: google.com/patents/US4457203#forward-citations –  Robert Tesla III Dec 21 '13 at 20:02
add comment

(The point of this post is to prove obviousness and lack of novelty in the '060 Patent)

WikiPedia: Nyquist Frequency

The Nyquist frequency, named after electronic engineer Harry Nyquist, is ½ of the sampling rate of a discrete signal processing system. It is sometimes known as the folding frequency of a sampling system.

Nyquist Frequency Aliasing

This describes common Nyquist anti-aliasing (filtering) which is also includes bandpass.

In a typical application of sampling, one first chooses the highest frequency to be preserved and recreated, based on the expected content (voice, music, etc.) and desired fidelity. Then one inserts an anti-aliasing filter ahead of the sampler. Its job is to attenuate the frequencies above that limit. Finally, based on the characteristics of the filter, one chooses a sample-rate (and corresponding Nyquist frequency) that will provide an acceptably small amount of aliasing.

Also read:

share|improve this answer
add comment

The more I look into this I find that it's obvious the patent covers basic human auditory senses.

Psychoacoustics:

http://en.wikipedia.org/wiki/Psychoacoustics

Hearing is not a purely mechanical phenomenon of wave propagation, but is also a sensory and perceptual event; in other words, when a person hears something, that something arrives at the ear as a mechanical sound wave traveling through the air, but within the ear it is transformed into neural action potentials. These nerve pulses then travel to the brain where they are perceived. Hence, in many problems in acoustics, such as for audio processing, it is advantageous to take into account not just the mechanics of the environment, but also the fact that both the ear and the brain are involved in a person’s listening experience.

The inner ear, for example, does significant signal processing in converting sound waveforms into neural stimuli, so certain differences between waveforms may be imperceptible.1 Data compression techniques, such as MP3, make use of this fact.2 In addition, the ear has a nonlinear response to sounds of different intensity levels; this nonlinear response is called loudness. Telephone networks and audio noise reduction systems make use of this fact by nonlinearly compressing data samples before transmission, and then expanding them for playback.3 Another effect of the ear's nonlinear response is that sounds that are close in frequency produce phantom beat notes, or intermodulation distortion products.4

Limits of perception

The human ear can nominally hear sounds in the range 20 Hz (0.02 kHz) to 20,000 Hz (20 kHz). The upper limit tends to decrease with age; most adults are unable to hear above 16 kHz. The lowest frequency that has been identified as a musical tone is 12 Hz under ideal laboratory conditions.5 Tones between 4 and 16 Hz can be perceived via the body's sense of touch. Frequency resolution of the ear is 3.6 Hz within the octave of 1000 – 2000 Hz. That is, changes in pitch larger than 3.6 Hz can be perceived in a clinical setting.5 However, even smaller pitch differences can be perceived through other means. For example, the interference of two pitches can often be heard as a (low-)frequency difference pitch. This effect of phase variance upon the resultant sound is known as beating.

The semitone scale used in Western musical notation is not a linear frequency scale but logarithmic. Other scales have been derived directly from experiments on human hearing perception, such as the mel scale and Bark scale (these are used in studying perception, but not usually in musical composition), and these are approximately logarithmic in frequency at the high-frequency end, but nearly linear at the low-frequency end.

The psychoacoustic model provides for high quality lossy signal compression by describing which parts of a given digital audio signal can be removed (or aggressively compressed) safely — that is, without significant losses in the (consciously) perceived quality of the sound.

It can explain how a sharp clap of the hands might seem painfully loud in a quiet library, but is hardly noticeable after a car backfires on a busy, urban street. This provides great benefit to the overall compression ratio, and psychoacoustic analysis routinely leads to compressed music files that are 1/10th to 1/12th the size of high quality masters, but with discernibly less proportional quality loss. Such compression is a feature of nearly all modern lossy audio compression formats. Some of these formats include Dolby Digital (AC-3), MP3, Ogg Vorbis, AAC, WMA, MPEG-1 Layer II (used for digital audio broadcasting in several countries) and ATRAC, the compression used in MiniDisc and some Walkman models.

Psychoacoustics is based heavily on human anatomy, especially the ear's limitations in perceiving sound as outlined previously. To summarize, these limitations are:

Psychoacoustics Model (from: http://en.wikipedia.org/wiki/File:Psychoacoustic_Model.svg)

Given that the ear will not be at peak perceptive capacity when dealing with these limitations, a compression algorithm can assign a lower priority to sounds outside the range of human hearing. By carefully shifting bits away from the unimportant components and toward the important ones, the algorithm ensures that the sounds a listener is most likely to perceive are of the highest quality.

How did they patent natural human instinct of auditory senses?

share|improve this answer
add comment

Non-Cited Prior Art

US6140568 (Priority Date: Nov 6, 1997)

This is a great example.

Title: System and method for automatically detecting a set of fundamental frequencies simultaneously present in an audio signal

ABSTRACT

A system and method for automatically detecting and identifying a plurality of frequencies simultaneously present in an audio signal, as well as the duration, amplitude, and phase of those frequencies, then filtering out harmonic components to determine which frequencies are fundamentals. The system includes a computer readable medium of instruction code that decomposes the signal into its component sine waves by computing and comparing correlations between the input signal and sine waves at various phase and amplitude combinations. The system also employs several optimization and error correction routines.

Claims

Claim 1

A method of identifying one or more fundamental frequencies simultaneously present in a complex signal, comprising the steps of:

receiving the complex signal;

decomposing the signal into sine wave components to determine all frequencies present in the signal;

setting and obtaining parameters used to detect fundamental frequencies;

and filtering out harmonic frequencies to determine the fundamental frequencies actually present in the signal.

Claim 2

The method of claim 1, wherein the step of receiving the complex signal includes dividing the signal into a series of sample windows which contain sample amplitudes.

Claim 3

The method of claim 2, further comprising an optimization step of ignoring sample windows in which the sample amplitudes do not meet a predetermined threshold.

Claim 4

The method of claim 2, wherein the step of decomposing includes obtaining best correlation scores by comparing reference frequencies to the complex signal, and comparing said best correlation scores to determine which reference frequencies are in the complex signal.

Claim 5

The method of claim 4, wherein the step of decomposing is carried out by comparing amplitudes resulting in the best correlation score at each reference frequency to amplitudes resulting in best correlation scores at adjacent reference frequencies to locate amplitude peaks.

Claim 6

The method of claim 4, wherein the step of decomposing includes the step of scaling the amplitudes of the detected frequencies according to said frequencies.

Claim 7

The method of claim 4, wherein the step of obtaining the best correlation scores comprises shifting the amplitude and phase of the reference frequencies until an amplitude/phase combination yielding the best correlation score out of all correlation scores for said combinations of each reference frequency is found.

Claim 8

The method of claim 4, further comprising the step of attenuating frequencies lower than current reference frequencies before correlation scores are obtained.

Claim 9

The method of claim 8, wherein said attenuation step comprises:

a high-pass filter having a cutoff frequency set to the current reference frequency; and

a low-pass filter having a cutoff frequency set to one half of the current reference frequency.

(High/Low Pass == BandPass)

Claim 10

The method of claim 7, further comprising an optimization step of ignoring amplitude/phase combinations which yield correlation scores that are worse than previously obtained correlation scores.

Claim 9 of US6140568 Proves that Claim 1 of '060 is not novel. Bandpass filtering around a target frequency is not novel, has been done, and this is prior art.

Claim 3 of US6140568 Proves that Claim 13 of '060 is not novel. This claims to select frequencies from above a predetermined threshold.

I feel that other claims in this patent as well would invalidate several claims within the '060 patent.

share|improve this answer
add comment

Non Cited Prior Art

PITCH DETECTION USING THE SHORT—TERM PHASE SPECTRUM

F.J. CHARPENTIER (Tokyo, 1986)

http://www.ee.columbia.edu/~dpwe/papers/Charp86-pitchphase.pdf

Abstract

A new frequency domain method for determining the fundamental frequency of speech is presented in this paper. This method uses the information contained in short-term phase spectrum whereas the previous methods were limited to the amplitude spectrum. The short-term spectrum is computed by DFT and is interpreted as the output of a bank of band-pass filters. Harmonic components are detected by searching for sets of three continuos filters having the same instantaneous frequency. The frequency of a detected harmonic is given by the instantaneous frequency itself. A conventional harmonic numbering algorithm is used to confer the set of detected harmonics to a value of the fundamental frequency. Preliminary results show the validity of the method.

This document goes into detail about detecting a frequency and automatically band-passing the signal to eliminate other harmonics. It was discusses it being used in a pitch detection application and eliminating other noise and computation below a threshold.

enter image description here enter image description here enter image description here

To further clarify, bandpass a frequency detected, and when applicable, reduce the onboard computation by only computing frequencies above a threshold. enter image description here

share|improve this answer
    
"...simultaneously using the information of the amplitude spectrum to restrict the calculations to the neighborhood of the spectral peaks" == 060: above a predetermined threshold with an upper frequency limit and a lower frequency limit. Seems Legit. –  Tyler Ulrich Dec 23 '13 at 14:56
add comment

Considering the substantial new question of patentability with prior art that was found in such a short time (5 days so far), one would be curious to how the patent was approved.

Information from USPTO Public Pair System

To obtain this patent information, go to the search, select Patent, and enter: 8502060 in the search box.

It will display the Application Data and audit trail (Transaction History).

USPTO Public Pair Application Data

This is the Transaction History, I particularly note no rejections to the claims, and a rapid approval process.

Transaction History Part 1

Transaction History Part 2

Next, under Image Wrapper File (public information) you see on 2/15/2013 when the application was filed, it had a Statement of Preexamination Search where the applicant cites a prior application and claims that all of the pre examination searches for prior art was completed.

Image File Wrapper

Document ID 13768799 - Statement of Preexamination Search by Applicant

Statement of Preexamination Search

Other supporting documents in the Image File Wrapper, especially this one, include an OATH or Declaration provided by the applicant which allows the applicant to be punished under 18 U.S.C. 1001.

This oath was provided on 2/15/2013 along with all of the other required documents and barrage of paperwork to the patent examiner.

I hereby acknowledge that any willful false statement made in this declaration is punishable under 18 U.S.C. 1001 by fine or imprisonment of not more than (5) years, or both.

Oath Document File ID: 13768799

Later the applicant eludes to Rob Toulson's patent in their prior art submission claiming that Toulson's patent as prior art is invalid, which is false.

On 2/15/2013 when the application was submitted, another document titled:

Accelerated Examination Support Document

Submitted by the applicant, 3 prior art examples noted as "Closest Related Match" begin on page 5.

Accelerated Examination Support Document

Particularly of concern is relation to Toulson's patent on Page 9 - 12 where the applicant states Rob Toulson's patent and methods are not taught. Page 12 in regards to Claim 13 are particularly concerning.

Toulson's Patent Claims Dispute

The point here is this was an accelerated patent and the examiner relied on the applicant's prior-art search, oath, and good word. This is alarming.

Take a closer look at this rapid process. 02/15/2013 Filed, 05/28/2013 Approval

Can anyone elude to ever seeing a patent fly through the system this fast? The USPTO Expedited Examination Program http://www.uspto.gov/patents/process/file/accelerated/index.jsp

The USPTO has established procedures under which the examination of a patent application may be accelerated. Under one of these procedures, the USPTO will advance an application out of turn for examination if the applicant files a grantable petition to make special under the accelerated examination program. The USPTO is similarly revising the procedures for other petitions to make special, except those based on applicant’s health or age or the PPH pilot program. Other petitions to make special (i.e., based on: manufacture, infringement, environmental quality, energy, recombinant DNA, superconductivity materials, HIV/AIDS and cancer, countering terrorism, and biotechnology applications filed by small entities MPEP § 708.02 ) will be processed using the revised procedure for accelerated examination. Thus, petitions to make special, except those based on applicant’s health, age, or the PPH pilot program , will be required to comply with the requirements for petitions to make special under the accelerated examination program as set forth in this notice.

Next, you have the

02-15-2013 PET.SPRE.ACX Petition for 12-month Accelerated Exam

(Petition to make Special) yet this petition clearly states in item 1:

the claims must be directed to a single invention

Since Claim 1 and Claim 13 are method claims that are not bound to an embodiment, are they directed towards a single invention

Petition for 12-month Accelerated Exam

PET.DEC.TC - Petition decision routed to the Technology Center to act on the decision or continue prosecution

Filed on 3/18/2013 and the patent buzzes along nicely through the system without anyone rejecting claims.

Petition

On 5/28/2013 the Patent Examiner publishes his brief prior art search. And rubber stamps it Approved the same day.

Patent Examiner Prior Art Search

Confused? Where can I buy this kind of expedited processing?

share|improve this answer
2  
Thanks for the audit trail. 18 U.S.C. 1001 can be a real problem, considering the patent examiner will most likely point his finger at the applicant! –  Tyler Ulrich Dec 16 '13 at 21:44
1  
This is useful information! I would not want to be involved in the process of this patent getting granted. –  Aron Stein Dec 16 '13 at 21:55
    
Seriously? Those are very vague keywords for a utility patent. Zzzzzzzz.... presses submit. //Patent Examiner –  Audio Sniper Dec 17 '13 at 20:09
add comment

I find pitch refinement an interesting topic.

WO1999059138A2 (May 11, 1998)

Refinement of pitch detection

Abstract

Successive pitch periods/frequencies are accurately determined in an audio equivalent signal. Using a suitable conventional pitch detection technique, an initial value of the pitch frequency/period is determined for so-called pitch detection segments of the audio equivalent signal. Based on the determined initial value, a refined value of the pitch frequency/period is determined. To this end, the signal is divided into a sequence of pitch refinement segments. Each pitch refinement segment is associated with at least one of the pitch detection segments. The pitch refinement segments are filtered to extract a frequency component with a frequency substantially corresponding to an initially determined pitch frequency of an associated pitch detection segment. The successive pitch periods/frequencies are determined in the filtered signal

Description

.... In step 130, each pitch refinement segment is filtered to extract the fundamental frequency component (also referred to as the first harmonic) of that segment. The filtering may, for instance, be performed by using a band-pass filter around the first harmonic. It will be appreciated that if the first harmonic is not present in the signal (e.g. the signal is supplied via a telephone line and the lowest frequencies have been lost) a first higher harmonic which is present may be extracted and used to accurately detect this representation of the pitch. For many applications it is sufficient if one of the harmonics, preferably one of the lower harmonics, is accurately detected. It is not always required that the actually lowest harmonic is detected. Preferably, the filtering is performed by convolution of the input signal with a sine/cosine pair as will be described in more detail below.

I read each pitch refinement segment ('138) as subsequent signal ('060)

In step 140, a concatenation occurs of the filtered pitch refinement segments. The filtered pitch detection segments are concatenated by locating each segment at the original time instant and adding the segments together (the segments may overlap). The concatenation results in obtained a filtered signal. In step 150, an accurate value for the pitch period/frequency is determined from the filtered signal. In principle, the pitch period can be determined as the time interval between maximum and/or minimum amplitudes of the filtered signal. Advantageously, the pitch period is determined based on successive zero crossings of the filtered signal, since it is easier to determine the zero crossings. Normally, the filtered signal is formed by digital samples, sampled at, for instance, 8 or 16 Khz. Preferably, the accuracy of determining the moments at which a desired amplitude (e.g. the maximum amplitude or the zero-crossing) occurs in the signal is increased by interpolation. Any conventional interpolation technique may be used (such as a parabolic interpolation for determining the moment of maximum amplitude or a linear interpolation for determining the moment of zero crossing). In this way accuracy well above the sampling rate can be achieved.

(Extract from above:)

the pitch period is determined based on successive zero crossings of the filtered signal

To determine Zero Crossings, you need to have a predetermined threshold. For it to be successive, it would have been required to exceed a predetermined threshold.

In summary, if this method is employed on a microphone, guitar, keyboard, amplifier, and human voice, detects a pitch/fundamental/harmonics and applies a bandpass filter around them for suppressing further action it would infringe. This could include spectral analysis, visualizers, frequency readouts and any derivative thereof.

share|improve this answer
    
If I sing a note, or even speak in a certain tone, then this device concatenates over time when a specific tone is heard by way of amplitude and filters. Displaying a histogram of a repetitive action within a certain criteria (filtered range above a sound level). I suppose this would be the same as tuning a drum, guitar, piano, voice? Repeatedly producing a pitch within a pitch range.. –  Robert Tesla III Dec 21 '13 at 19:55
    
Pitch recognition is done by spectral analysis. Nice find. YEP. –  Fullsail SoundDesign Dec 21 '13 at 20:25
add comment

Non-Cited Prior Art

US4434697

Indicator apparatus for indicating notes emitted by means of a musical instrument

Claim 1

Apparatus for indicating the presence of musical notes and for identifying the musical notes detected comprising:

means for amplifying input signals corresponding to musical notes to be identified; filter means connected to said means for amplifying for eliminating harmonics from said input signals;

energy detecting means connected to said means for amplifying for detecting input signals exceeding a predetermined threshhold;

memory means including at least one memory for storing items of information representing a table of musical notes;

means for calculating the frequency and octive of said input signal, said means for calculating including microprocessor means and being connected to said filter means, said energy detecting means and said memory means, said means for calculating being responsive to input signals received from said filter means exceeding said predetermined threshhold determined by said energy detecting means to calculate the frequency thereof, said frequency calculated being employed to read from said memory for storing items of information representing a table of musical notes items of information representing the closest corresponding musical note for said frequency calculated; and

means for displaying, in alphanumeric form, said musical note read closest to each successive musical note in said input signals to be indentified and the octive in which said musical note resides.

Claim 2

The apparatus according to claim 1 wherein said means for calculating additionally comprises:

means for determining any difference between said frequency of said input signals calculated, and said closest corresponding musical note read;

means for providing indicia representative of any difference determined; and

means for supplying said indicia to said means for displaying to cause said indicia to be displayed.

Claim 3

The apparatus according to claim 2 wherein said indicia take the form of a plurality of signs, and selected ones of said plurality of signs respectively indicate that a difference between a calculated frequency of an input signal and a frequency of a displayed musical note is a positive value exceeding a predetermined limit, a negative value exceeding a selected limit and a value within a limit.

Claim 4

The apparatus according to any one of claims 1, 2 or 3 wherein said filter means comprises:

a plurality of low pass filter means for receiving said input signals representing said musical notes to be identified;

a plurality of threshhold detector means for indicating that input signals applied thereto exceed a predetermined threshhold, each of said plurality of threshhold detector means being connected to an associated one of said plurality of low pass filter means;

means for determining a one of said plurality of low pass filter means having the lowest cut-off frequency characteristic and at least a portion of said input signals having a predetermined threshhold level passing therethrough; and means for inhibiting outputs from remaining ones of said plurality of low pass filter means having cut-off frequency characteristics higher than that of said one of said plurality of low pass filter means.

share|improve this answer
    
"said means for calculating including microprocessor means and being connected to said filter means, said energy detecting means and said memory means, said means for calculating being responsive to input signals received from said filter means exceeding said predetermined threshhold determined by said energy detecting means to calculate the frequency thereof" == YEP –  Fullsail SoundDesign Dec 21 '13 at 20:24
add comment

Non Cited Prior Art

EP0285238B1 (1987)

Digital bandpass oscilloscope

This patent from 1987 discusses an Oscilloscope with bandpass filters, an accumulator that uses gates (predetermined threshold) and can easily be used to invalidate claims of the '060 patent.

This patent talks about using Time and converting to Frequency Domains, analyzing a signal, applying a bandpass filter to it (high/low frequency limits) and deriving subsequent signals from bandpass above a predetermined threshold (gates). Look under [0047] in the description for gates used in the embondiment.

Background of the Invention

[0001] The present invention relates to digital storage oscilloscopes in general and in particular to a digital oscilloscope for displaying waveforms representing component signals of frequency within selectable passbands of an input signal.

[0002] The behavior of component signals of frequency within selected frequency bands of a wideband analog signal is often of interest, and spectrum analyzers provide researchers with frequency domain plots of signal amplitudes within a band. However, sometimes researchers wish to view a frequency band of interest as a time domain plot. Oscilloscopes plot signal magnitudes as function of time, but when a waveform representing an input signal is displayed by a conventional oscilloscope, signal components having frequencies within a particular frequency band of interest are often difficult to observe due to the presence of higher or lower frequency components. A analog bandpass filter is sometimes utilized to remove the higher and lower frequency components from the analog signal before it is applied to the input of an oscilloscope, but many different bandpass filters would be needed in order to separately view a wide range of selectable passbands.

[0047] Accumulator 132 includes an adder 134 and a random access memory (RAM) 136. Adder 134 is adapted to add each output term produced by multiplier 130 to an accumulated sum R stored in RAM 136. The sum produced by adder 134 may then be stored in RAM 136, thereby replacing the accumulated sum R with the result of the addition. Addressing of RAM 136 is controlled by an address signal (ADDR) provided by state machine 118 of FIG. 8. Data output terminals of RAM 136 are coupled to an input of adder 134 through a set of AND gates 138, each having another input controlled by a signal NADD supplied by state machine 118 of FIG. 8. When NADD is low, a 0 value, rather than the currently addressed data in RAM 136, is passed to adder 134. The NADD signal may be driven low when the output of multiplier 130 is the first term of a sum to be accumulated in RAM 136 so that adder 134 merely added a 0 to that term and forwards it for storage in RAM 136. The output of adder 134 is coupled to data input terminals of RAM 136 through another set of AND gates 140. A signal NLOAD produced by state machine 118 is applied to an additional input of each AND gate 140 and is driven low when RAM 136 is to store a 0 value rather than the output of adder 134. The NLOAD signal allows the contents of any storage location in RAM 136 to be initialized to 0 when necessary.

(going through a serious of noise gates to determine if the signal is worthy of displaying above a threshold / level).

EP0285238B1

share|improve this answer
add comment

Non-Cited Prior Art

US 3812432 A (Priority Jan 5, 1973)

https://www.google.com/patents/US3812432

Tone detector

Description (Abstracted)

More specifically, a tone detector, in accordance with the invention, includes a threshold detector for generating a substantially constant amplitude pulsating signal representative of intervals between prescribed levels of the instantaneous amplitude of an applied signal, for example, positive and negative peak amplitudes. No pulsating signal is generated during intervals in which the amplitude of the applied signal is below the prescribed level. The presence of frequency components of interest in the applied signal is determined by supplying the pulsating signal to appropriate filters. The peak value of the output from each filter is detected and compared with a predetermined reference signal to determine, in accordance with the invention, whether the particular filter output represents the fundamental frequency of the applied signal.

An additional aspect of the instant invention is concerned with eliminating possible detection errors caused by noise signals. Such errors are substantially eliminated, in accordance with the invention, by inhibiting the operation of the threshold detector until the is disabled until the average amplitude of the applied signal exceeds a predetermined value.

  • FIG. 1 depicts a tone detector circuit illustrating the invention;
  • FIG. 2 shows in greater detail a threshold detector which may be utilized in the tone detector of FIG. 1;
  • FIG. 3 illustrates details of an average detector which may be employed in the circuit of FIG. 1;
  • FIG. 4 depicts details of a frequency component detector which may be used in the tone detector of FIG. 1; and

  • FIGS. 5A, 5B and 5C each show a sequence of waveforms useful in describing operational modes of the tone detector of FIG. 1.

Claim 1

A tone detector circuit which comprises:

means responsive to an applied signal for generating a pulsating signal having a substantially constant amplitude and being representative of intervals between prescribed amplitude levels of said applied signal, said pulsating signal generating means being selectively disabled in response to a predetermined signal being supplied thereto;

means in circuit relationship with said pulsating signal generating means and being responsive to at least one predetermined frequency component of said pulsating signal for generating a pulse signal representative of intervals when the amplitude of said at least one frequency component exceeds a predetermined reference level;

means for detecting a predetermined amplitude characteristic of said applied signal; and

means in circuit relationship with said amplitude detecting means and said pulsating signal generating means and being responsive to the output from said amplitude detecting means for generating a signal to disable said pulsating signal generating means during intervals when the output from said amplitude detecting means is below a predetermined level.

Suppressing below a predetermined threshold is the same as displaying above a predetermined threshold

Claim 2

A tone detector as defined in claim 1 wherein said amplitude detecting means includes means for generating a signal representative of the average amplitude of said applied signal.

Claim 3

A tone detector as defined in claim 1 wherein said pulsating signal generating means includes a threshold detector for generating a pulsating signal having a substantially constant amplitude and being representative of intervals between prescribed values of the instantaneous amplitude of said applied signal.

Claim 4

A tone detector as defined in claim 3 wherein said pulse signal generating means includes filter means in circuit relationship with said threshold detector for passing only said at least one frequency component of said pulsating signal and level detector means in circuit with said filter means for generating said pulse signal during intervals in which the amplitude of said at least one frequency component exceeds a predetermined level.

Claim 5

A tone detector as defined in claim 4 wherein said level detector means includes peak detector means in circuit with said filter means for generating a unipolarity signal having an amplitude proportional to the peak amplitude value of said at least one frequency component output from said filter means and comparator means having first and second inputs and an output, a reference signal source being connected in circuit with said first input, said peak detector means being connected in circuit with said second input and said comparator means being responsive to a reference signal and to said unipolarity signal for generating said pulse signal at said comparator means output during intervals in which the amplitude of said unipolarity signal exceeds the amplitude of said reference signal.

In summary, 3812432 is a method of detecting tone above a predetermined threshold, limited by filter means to identify the tone. '060 Claim 13 and quite possibly Claim 1 invalid. By displaying the output of any calculations, equated to digits represented by frequencies, this is effectively the same as '060 in my opinion.

3812432 Figure of Filter and Threshold Detector

share|improve this answer
add comment

Non Cited Prior Art

US 4227437 A (1977)

Frequency detecting apparatus

Note: a stringed instrument is also a resonant structure, in fact everything in the world resonates and therefore is a resonant structure.

Abstract

A detecting apparatus for tuning musical instruments receives an input signal from a sound transducer and removes second harmonic content therefrom by means of a filter responsive to a note and octave selection. A phase lock pulse generator receives the filtered output and generates a signal in step with the input sound signal. The generated signal is counted by a note counter during a gating period derived by counting a predetermined number of output cycles of an oscillator started in step with the input signal. Three outputs indicating "on-frequency", "sharp" or "flat" are supplied in the alternative at the end of the gating period in accordance with the count in the note counter at that time

Claim 1

  1. Frequency detection apparatus for tuning a stringed instrument or the like, comprising:

a transducer for receiving sound information from a vibrating string on said instrument and for converting the same into an electrical signal of like frequency, selection means for selecting a note corresponding to a correct frequency to be detected,

a filter for receiving said electrical signal and responsive to said selection means for filtering second harmonic information from said electrical signal,

a first oscillator phase locked to the signal provided by said filter for generating a phase locked signal,

a second oscillator also responsive to the signal provided by said filter to produce a standard frequency output signal initiated in synchronous time relation with the signal provided by said filter,

a control counter coupled to the output of said second oscillator for counting the output cycles of said standard frequency output signal, and a gating circuit responsive to a count of said control counter for gating said phase locked signal during a gating period as determined by a predetermined counted number of cycles of said standard frequency output signal,

note counter means coupled for receiving the phase locked signal as gated by said gating circuit for detecting whether the number of output cycles of the gated phase locked signal received during said gating period reaches a predetermined count corresponding to the note selected by said selection means, and display means for producing first, second and third indications for respectively indicating substantial identity in frequency between the electrical signal and the note selected by said selection means, or whether the electrical signal is above or below said note, said display means receiving the output of said note counter means substantially at the end of said gating period produced by said gating circuit to determine whether said note counter means has made a full count corresponding to the selected note substantially at the end of said gating period.

Claim 2

The apparatus according to claim 1 wherein said note counter means includes a note counter for counting said phase locked signal as gated by said gating circuit, and a plurality of note gates, each responsive to a note selection by said selection means for recognizing a particular count output from said note counter, said note gates supplying a common output, said display means being responsive to said gating circuit at the end of the gating period for testing said common output from said note gates to determine whether said note counter means has made said full count corresponding to the selected note substantially at the end of said gating period.

Claim 3

The apparatus according to claim 1 wherein said note counter means includes a note counter for counting said phase locked signal as gated by said gating circuit, and a plurality of note gates, each responsive to a note selection by said selection means for recognizing a particular count output from said note counter, said note gates supplying a common output,

said display means being responsive to said gating circuit at the end of the gating period for testing said common output from said note gates to determine whether said note counter means has made said full count corresponding to the selected note substantially at the end of said gating period.

This patent claims (paraphrased):

  • Plucking a string ( a resonant structure per '060 )
  • Identifying the note via the tuner means
  • Applying a BandPass Filter around the detected note (target frequency).
  • Providing a gating mechanism (for sounds above a predetermined threshold).
  • Deriving subsequent signals above a predetermined threshold with a high/low filter bounds determined by the previous signal (as per any filter has, that is what a filter is.).
share|improve this answer
add comment

Programming Electronic Music in Pd Johannes Kreidler 27-01-2009 http://www.pd-tutorial.com/english/ch03s08.html This seems a good primer for understanding what's going on with FFT, selection, filtering, displaying, and the like**

3.8.1.1 Analyzing partials

Let's return to a basic concept of additive synthesis: a sound comprises partials. If you want to find out what the component parts of a sound are, you could employ a set of band-pass filters for every partial: ... 3.8.2.1 Filters ... 3.8.3.2 Tuner

Here's one way to build a tuner: patches/3-8-3-2-tuner.pd

share|improve this answer
4  
3.8.2.1 Filters What's useful about FFT, of course, is that the values it determines can be changed before you resynthesize the components into a sounding result. For example, you could set certain bins to be louder or quieter; you could build filters like high-pass, low-pass, etc., or 'draw' one yourself. patches/3-8-2-1-fft-filter.pd –  Robert Tesla III Dec 20 '13 at 15:10
add comment

MOTU has been shipping their CueMix FXsoftware with their mixer units for quite some time.

MOTU Tuner Plugin

Instrument tuner (Published: 2010-10-14 15:51:06 )

Just open the Tuner window, play a note, and use the large graphic display to get in tune with an accuracy of one 10th of a cent (one 1,000th of a semi-tone). Being in tune has never been easier.

The Tuner displays the detected note by frequency (in Hertz), note name and octave, with an adjustable reference frequency for A4 between 400 and 480 Hz.

The large meter gives you a clear indication of how high or low you are from the detected pitch.

Large red arrows direct you up or down as needed to zero in on the correct pitch.

You can even tune phase-coherent stereo signals.

The CueMix FX Tuner is as advanced and accurate as any dedicated hardware tuner out there.

There is also an FFT Plugin:

FFT Plugin

FFT Display

CueMix FX provides an optional real-time FFT display super-imposed on top of the graphic EQ curve so you can see as well as hear the effect of your equalization adjustments.

This software ships with MOTU Audio Interfaces which all have Noise Gates, Bandpass filters, and more built right into the plugins and physical input boards.

http://www.motu.com/products/cuemix-fx/tuner-analysis.html

http://www.motu.com/products/cuemix-fx/new-in-cuemix

share|improve this answer
    
The FFT Display is affected by the EQ curve. An EQ is built of filters and bands designed to suppress or attenuate signals in/out of those bands. Several EQ, visual ones more so, adjust band ranges / filters. The fact this displays pitch analysis while allowing it to be affected by the filter ranges of the EQ is an interesting example. bhphotovideo.com/find/newsLetter/Understanding-EQ.jsp –  Robert Tesla III Dec 20 '13 at 17:34
    
One would also acknowledge that the input gain on a physical motu device where this plugin receives it's sound source would be considered a "Predetermined Threshold" and it's a manual requirement to set for any procedure for recording with a microphone or other instrument. –  Robert Tesla III Dec 20 '13 at 17:36
add comment

This is a Lab at College of Engineering Montana State University Spring of 2006

EE477 Digital Signal Processing Spring 2006

Lab 11

http://www.coe.montana.edu/ee/rmaher/ee477_SP06/ee477_fftlab_sp06.pdf

Short-time Fourier transform (STFT)#

....

The DFT is a frequency-sampled version of the Fourier transform, so multiplying the DFT by a filter function in the frequency domain is actually the equivalent of circular convolution, not linear convolution. This means that the resulting time domain signal may have “time domain aliasing”if the effects of the circular overlap are not accounted for. Refer to an authoritative DSP textbook for the details of this issue.

....

Exercise C: STFT for signal processing

Now that the basic pass program is working, you can consider some more interesting STFT- based processing. In this part you will do some signal-dependent processing. Since the DFT gives a complex view of the input signal’s short-time spectrum, we can take advantage of the spectral analysis to do some signal enhancement.

It is common to have an input signal that is contaminated with unwanted broadband noise. One way to reduce the undesired noise is to use a spectral threshold. The algorithm compares the spectral magnitude in each FFT bin to a threshold value. If the magnitude is above the threshold, it is assumed to be “signal”and gets passed unaltered. On the other hand, if the measured magnitude in an FFT bin is below the threshold, it is assumed to be noise and the bin is set to zero. If the threshold is chosen carefully, the output signal will have less audible noise than the input signal. This process is known as a “de-hisser”or a spectral “noise gate.”

In order to do the threshold test you will need to add another parameter to the block_fft() function: xx=block_fft(in,fftlen,thresh), and use the threshold and the FFT magnitude to take out the FFT bins that are below the threshold. You will also need to experiment with various threshold values, and try different types of input signals to verify that your de-hisser is working. Demonstrate your code by choosing an appropriate threshold value for the noisy speech signal on the course web site (instructor check off C).

share|improve this answer
    
lol a school project? So now every engineering lab could get busted? –  Aron Stein Dec 20 '13 at 4:46
add comment

I would like to offer my insight. I have asked my professor to present this in class as we are learning about FFT and DSP signal analysis. Perhaps my classmates can join in.

Part of audio training is learning about Apple's Audio Units. Audio Units have been the cornerstone in apple's audio development for years and is utilized by developers that all would infringe upon this patent.

Audio Units on Mac OS

https://developer.apple.com/library/mac/documentation/musicaudio/Conceptual/AudioUnitProgrammingGuide/TheAudioUnit/TheAudioUnit.html

Under Listing 2-10

The audio unit you build in “Tutorial: Building a Simple Effect Unit with a Generic View” makes use of all three of these methods: GetParameterInfo, GetParameter, and SetParameter.

An audio unit sometimes needs to invoke a value change for one of its parameters. It might do this in response to a change (invoked by a view or host) in another parameter. When an audio unit on its own initiative changes a parameter value, it should post an event notification.

For example, in a bandpass filter audio unit, a user might lower the upper corner frequency to a value below the current setting of the frequency band’s lower limit. The audio unit could respond by lowering the lower corner frequency appropriately. In such a case, the audio unit is responsible for posting an event notification about the self-invoked change. The notification informs the view and the host of the lower corner frequency parameter’s new value. To post the notification, the audio unit follows a call to the SetParameter method with a call to the AUParameterListenerNotify method.

Apple Audio Units MAC OSX 10.6 (August, 2009)

AudioUnitFrequencyResponseBin An audio unit’s audio level at a particular frequency.

typedef struct AudioUnitFrequencyResponseBin { Float64 mFrequency Float64 mMagnitude; } AudioUnitFrequencyResponseBin;

Fields

mFrequency

mMagnitude

Discussion An array of AudioUnitFrequencyResponseBin are passed in to kAudioUnitProperty_FrequencyResponse with the mFrequency field filled in. The array is returned with the mMagnitude fields filled in. If fewer than kNumberOfResponseFrequencies are needed, then the first unused bin should be marked with a negative frequency.

Availability Available in OS X v10.6 and later. (Aug, 2009) Declared In AudioUnitProperties.h

All apple audio software, to include itunes (think visualizer) and any audio/input output driver on Mac OS uses AudioUnits.

Audio Units in IOS Since at Least 2008

  • The '060 patent now has priority from Nov 30,2011 over any and all apps that respond to voice recognition, instrument tuning, spectral analysis, and several other applications of various sorts.

https://developer.apple.com/library/iOS/documentation/AudioUnit/Reference/AudioUnitParametersReference/Reference/reference.html

Since IOS 2.0 (July 11, 2008 saw the public release of iPhone OS 2.0, with upgrades through version 2.2.1 made available.)

Constants

kMultiChannelMixerParam_Volume

Sets the audio gain for a mixer input or the output. Range is from 0 (for silence) through 1 (for unity gain). Available in iOS 2.0 and later. Declared in AudioUnitParameters.h.

kMultiChannelMixerParam_PreAveragePower

Indicates the average “pre” power in decibels (dB). Read only. Available in iOS 2.0 and later. Declared in AudioUnitParameters.h.

kMultiChannelMixerParam_PrePeakHoldLevel

Indicates the “pre” peak hold level in decibels (dB). Read only. Available in iOS 2.0 and later. Declared in AudioUnitParameters.h.

kMultiChannelMixerParam_PostAveragePower

Indicates the average “post” power in decibels (dB). Read only. Available in iOS 2.0 and later. Declared in AudioUnitParameters.h.

kMultiChannelMixerParam_PostPeakHoldLevel

Indicates the “post” peak hold level in decibels (dB). Read only. Available in iOS 2.0 and later. Declared in AudioUnitParameters.h.

AS OF IOS 5.0 (Release Date: October 12, 2011)

Important Note: Developer Preview of all Apple Code is published in June, 2011. All documents relating to default BandPass filtering were provided public online. Any app updated to IOS 5 that used spectrum/frequency display of any time would be employing these default methods

Bandpass Unit Parameters Parameters for the Bandpass unit.

enum { kBandpassParam_CenterFrequency = 0, kBandpassParam_Bandwidth = 1 };

Constants

kBandpassParam_CenterFrequency

Range is from 20 Hz to less than the Nyquist frequency (half the sample rate). Default value is 5,000 Hz. Used on the Global scope. Available in iOS 5.0 and later. Declared in AudioUnitParameters.h.

kBandpassParam_Bandwidth

Range is from 100 through 12000 cents. Default value is 600 cents. Used on the Global scope. Available in iOS 5.0 and later. Declared in AudioUnitParameters.h.

Highpass Unit Parameters Parameters for the Highpass unit.

enum { kHipassParam_CutoffFrequency = 0, kHipassParam_Resonance = 1 };

Constants

kHipassParam_CutoffFrequency

Range is from 10 Hz to less than the Nyquist frequency (half the sample rate). Default value is 6900 Hz. Used on the Global scope. Available in iOS 5.0 and later. Declared in AudioUnitParameters.h.

kHipassParam_Resonance

Range is from –20 through +40 dB. Default value is 0 dB. Used on the Global scope. Available in iOS 5.0 and later. Declared in AudioUnitParameters.h.

Lowpass Unit Parameters

Parameters for the Lowpass unit.

enum { kLowPassParam_CutoffFrequency = 0, kLowPassParam_Resonance = 1 };

Constants

kLowPassParam_CutoffFrequency

Range is from 10 Hz to less than the Nyquist frequency (half the sample rate). Default value is 6900 Hz. Used on the Global scope. Available in iOS 5.0 and later. Declared in AudioUnitParameters.h.

kLowPassParam_Resonance

Range is from –20 through +40 dB. Default value is 0 dB. Used on the Global scope. Available in iOS 5.0 and later. Declared in AudioUnitParameters.h.

High Shelf Filter Unit Parameters Parameters for the High Shelf Filter unit.

enum { kHighShelfParam_CutOffFrequency = 0, kHighShelfParam_Gain = 1 };

Constants

kHighShelfParam_CutOffFrequency

Range is from 10000 Hz to less than the Nyquist frequency (half the sample rate). Default value is 10000 Hz. Used on the Global scope. Available in iOS 5.0 and later. Declared in AudioUnitParameters.h.

kHighShelfParam_Gain

Range is from –40 through +40 dB. Default value is 0 dB. Used on the Global scope. Available in iOS 5.0 and later. Declared in AudioUnitParameters.h.

Low Shelf Filter Unit Parameters

Parameters for the Low Shelf Filter unit.

enum { kAULowShelfParam_CutoffFrequency = 0, kAULowShelfParam_Gain = 1 };

Constants

kAULowShelfParam_CutoffFrequency

Range is from 10 through 200 Hz. Default value is 80 Hz. Used on the Global scope. Available in iOS 5.0 and later. Declared in AudioUnitParameters.h.

kAULowShelfParam_Gain Range is from –40 through +40 dB. Default value is 0 dB. Used on the Global scope. Available in iOS 5.0 and later. Declared in AudioUnitParameters.h.

The next set of Constants (Defaults) are in relation to the Dynamics Processing, which is the pre-processing of all signals.

Dynamics Processor Unit Parameters

Parameters for the Dynamics Processor unit.

enum { kDynamicsProcessorParam_Threshold = 0, kDynamicsProcessorParam_HeadRoom = 1, kDynamicsProcessorParam_ExpansionRatio = 2, kDynamicsProcessorParam_ExpansionThreshold = 3, kDynamicsProcessorParam_AttackTime = 4, kDynamicsProcessorParam_ReleaseTime = 5, kDynamicsProcessorParam_MasterGain = 6, kDynamicsProcessorParam_CompressionAmount = 1000, kDynamicsProcessorParam_InputAmplitude = 2000, kDynamicsProcessorParam_OutputAmplitude = 3000 };

Constants

kDynamicsProcessorParam_Threshold

Range is from –40 through +20 dB. Default value is –20 dB. Used on the Global scope. Available in iOS 5.0 and later. Declared in AudioUnitParameters.h.

kDynamicsProcessorParam_HeadRoom

Range is from 0.1 through 40 dB. Default value is 5 dB. Used on the Global scope. Available in iOS 5.0 and later.

https://developer.apple.com/library/iOS/documentation/AudioUnit/Reference/AudioUnitParametersReference/Reference/reference.html

AudioUnits specifically has default settings for peak selection, filtering, cuttoff (high/low pass filters == bandpass filter) and more. Since these DEFAULTS are on every Apple Audio Unit, anything displaying a spectrum of frequencies, a calculated musical note, or possibly a visualizer would infringe upon '060.

All audio units, especially in the last 5 years on mobile devices has had built-in bandpass and high/low pass filtering. Likewise for amplitude selection.

These filters have been built into Apple's IOS Audio Units for quite sometime. It was often considered that processing every possible signal on a mobile device is not efficient, and reduces battery lifetime considerably. The same goes for any device that runs on batteries and employs a signal display; especially with a physical motor/servo driven needle. Utilizing frequencies above "predetermined threshold" while suppressing the other signals is Audio 101 for portable devices.

It is my belief that:

  • Any Mac OS Software which detects a frequency through a microphone and displays it such as Logic, Garage Band, and several 3rd Party software will utilize a predetermined threshold and a high/low (bandpass)filter or an upper and lower frequency bounds
    • Any IOS app which will utilize a predetermined threshold and a high/low (bandpass)filter or an upper and lower frequency bounds
    • Thousands of applications and software plugins by several vendors prove prior art over this and that it is not a unique concept, yet it's the standard way of performing a signal processing method, at least on Apple.
share|improve this answer
    
Are you saying that any spectrum analyzer on any mac device employs high/low frequency bounds and a predetermined threshold even if it wasn't specifically in it's design? google.com/search?q=app+visualizer+apps google.com/search?q=ios+tuner+apps –  Tyler Ulrich Dec 20 '13 at 0:30
    
I believe that it is implied that filters, thresholds, and all audio units and anything built with them would infringe Claim 13 by just the defaults alone. Claim –  Patented Dec 20 '13 at 1:55
    
Yes any apple based (and other common think: OpenGL visualizers) software that can recognize pitch, and even better do it in real time via a microphone (to include Google Maps responding to your words) and more will infringe upon these methods. It confuses me that this patent was filed in 2011 and revised in 2013. Who are these geniuses who didn't realize what they patented? Where was the examiner? –  Fullsail SoundDesign Dec 20 '13 at 2:10
2  
This is a very long answer that demonstrates many of the building blocks were known. It does not address the key things that got the patent allowed. In finding claims 1-12 allowable, the examiner wrote: " . . . claims 1-12 are considered allowable over the prior art, as the closest prior art, Chiba, Richardson et aI., and the Toulson references, do not explicitly teach suppressing a display of frequencies or musical notes from a subsequent signal that deviate from the filter mode reference frequency or musical note by a predetermined threshold as recited." –  George White Dec 20 '13 at 5:40
1  
@GeorgeWhite this is a good point. We should be focusing our search for a reference that says:"suppressing a display of frequencies or musical notes from a subsequent signal that deviate from the filter mode reference frequency or musical note by a predetermined threshold" sure seems trivial though. There's getting to be a lot of noise on here that might detract from on-point answers. :( –  Frank-n'Grind Dec 20 '13 at 14:15
show 1 more comment

Drum tuning as compared to tuning most musical instruments does not involve detecting simple series of harmonically related notes (same note in different octaves), but different unrelated notes generated by the sundry different length physical paths that different head resonances travel over.

060 claim 1 with several means combined describes applying a bandpass filter to a conventional note sniffer measurement string to preferentially isolate one resonance pitch and reject other resonances. This may have some utility in the context of parsing the multiple pitches drums can make when struck at different places on the head. If novel(?) this seems like a narrow improvement patent that should not control the basic measurement technology of generic note sniffers not specifically using a BPF (but I am not a patent lawyer).

060 claim 13 seems like a paraphrased version of claim one in abstract terms where instead of a BP filter "means", the frequency band has an upper and lower limit? This equivalent claims language is unclear (to me) whether they are describing computer software or something else?

I have a patent related to drum tuning (US 6,925,880), while I do not use simple note sniffing or FFT (yet). I am familiar with the instrument and technology.

US 6,925,880 Claim 18

A method for measuring the acoustic properties of a drumhead of a drum, comprising:

applying energy to a surface of the drumhead to cause the drumhead to emit acoustic energy therefrom; receiving acoustic energy emitted from the drumhead using moveable acoustic sensor positioned above the surface of the drumhead and in proximity to a tightening bolt of the drum without substantial interference from other tightening bolts of the drum;

converting the acoustic energy received by the acoustic sensor into a signal corresponding to an acoustic property of the drumhead; and adjusting all of the tightening bolts of the drum to a single resonant frequency

share|improve this answer
    
This is a method claim that is not bound to any specific embodiment. Even though this patent is labeled Drum-Set Tuner, it is still a method patent. Therefore, any claim can be independent and cover a method of performing an action, such as picking up a phone and dialing it. Which, older phones used BandPass Filters, Tone Recognition and suppressed any unrecognized tones (outside their filters) from the display as you dialed. This method effectively covers that now, even though phones like this have een around forever. –  Aron Stein Dec 17 '13 at 20:15
2  
I am reviewing your patent Mr Roberts. Very fascinating work. That is a true invention worthy of patenting. Do you feel like any of your claims were possibly used in this patent? I see that you are listed on the face of the '060 as Cited Art. I would be curious to see the arguments. –  Robert Tesla III Dec 17 '13 at 20:33
    
@ Robert Tesla III my answer was too long for the window... short answer is no I did not claim using a bandpass filter with FFT, but yes I discriminate between fundamental and overtone for drum tuning. I have described this in my literature and on my website for several years. –  John Roberts Jan 2 at 20:16
add comment

Frequency based criterion for distinguishing tonal and noisy spectral components http://www.cscjournals.org/csc/manuscript/Journals/SPIJ/volume4/Issue1/SPIJ-56.pdf International Journal of Computer Science and Security, Volume (4): Issue (1) by M. Kulesza No copyright notice in casual glance, however, no references cited beyond 2009.

(1)receives a sound, Fourier Transforms it to Magnitude/Power Spectrum and selects peak magnitude

share|improve this answer
    
@GeorgeWhite any thoughts on this regarding Claim 13? –  Frank-n'Grind Dec 20 '13 at 14:17
add comment

http://www.technick.net/public/code/cp_dpage.php?aiocp_dp=guide_dft_db_display "Music 320 Background Reader" by Julius O. Smith III, (Course Background Reader, Music 320). Copyright © 2001-01-02 by Julius O. Smith III. - Center for Computer Research in Music and Acoustics (CCRMA), Department of Electrical Engineering, Stanford University. This is a modified HTML version reproduced by permission. Copyright © 1997-2011 Nicola Asuni - Tecnick.com LTD - All Rights Reserved. ...

Code in Matlab: sound(xw,fs); % Might as well listen to it xzp = [xw,zeros(1,N-L)];% Zero-padded FFT input buffer X = fft(xzp); % Spectrum of xw, interpolated by factor ZP

Xmag = abs(X); % Spectral magnitude Xdb = 20*log10(Xmag); % Spectral magnitude in dB

XdbMax = max(Xdb); % Peak dB magnitude Xdbn = Xdb - XdbMax; % Normalize to 0dB peak

dBmin = -100; % Don't show anything lower than this Xdbp = max(Xdbn,dBmin); % Normalized, clipped, dB magnitude spectrum fmaxp = 2*f; % Upper frequency limit of plot, in Hz kmaxp = fmaxp*N/fs; % Upper frequency limit of plot, in bins fp = fs*[0:kmaxp]/N; % Frequency axis in Hz

% Ok, plot it already!

subplot(2,1,1); plot(1000*t,xw); xlabel('Time (ms)'); ylabel('Amplitude'); title(sprintf('a) %d Periods of a %3.0f Hz Sinusoid, Kaiser Windowed',nper,f));"

share|improve this answer
add comment

http://pi.physik.uni-bonn.de/~dieckman/DFT/DFT.html Amplitude and Phase of a discrete Fourier Spectrum Last change on: Wed 7 Sep 2011 [not sure when initially published]

"This tutorial describes the calculation of the amplitude and the phase from DFT spectra with finite sampling...

Finding Amplitude and Phase for select frequencies The next statement collects all bin numbers of the spectrum with bin values [magnitudes] over threshold into a list s. The threshold is used to suppress insignificant bins...[pseudo/real code follows enabling]"

== 13. A method ... comprising: providing one or more power spectrum frequency samples; selecting a frequency ... having a largest power spectrum magnitude ....

share|improve this answer
    
I would assume list s is for estimating with pseudo code over a period of time in the frequency domain, or as the '060 diagram calls it "pitch estimator" –  Aron Stein Dec 20 '13 at 0:45
    
I interpret the above "collecting ... into a list" based on "amplitude" value analogous to Claim 13's "selecting" based on "magnitude" in which both are operating on frequency power spectra. –  Frank-n'Grind Dec 20 '13 at 14:22
add comment

http://www.plogue.com/bidule/help/ch04s11.html Bidule user manual (C) 2001

"FFT:Time Domain to Spectral Domain Conversion ... Spectral To Loudest Freq/Amp

Takes the frequency+magnitude in input and outputs the loudest frequency with its amplitude"

== 13. ... comprising: providing one or more power spectrum frequency samples; selecting a frequency in a frequency band having a largest power spectrum magnitude from the one or more power spectrum frequency samples...

share|improve this answer
    
FFT provides power spectrum frequency sample which is then plugged and chugged to "output the loudest freq" which to me is analogous to Claim 13's "selecting" the freq based on "magnitude." –  Frank-n'Grind Dec 20 '13 at 14:23
add comment

US PG Pub 2010/0128897 with priority to Japanese Application 2007-092067 (Mar. 30, 2007), PCT/JP08/55757, then US filing of 371 on Sep. 30 2009. This seems good 102(b) and 102(e) art:

"...mixed signals for one frame which are converted into those in a frequency region [fourier transformed == "power spectrum frequency sample"] ... A noise signal selection unit selects a noise signal for each frequency bin on the basis of the amount of noise measured." (Abstract).

See Fig. 10. Microphones 15 accept sounds, Fourier Transform (FFT) 17 to a power spectrum frequency sample, and noise amount measuring unit 140(240) selects the frequency bin with the loudest or highest magnitude noise. Q.E.D. :^)

8502060 13. A method for pitch detection, comprising: providing one or more power spectrum frequency samples; selecting a frequency in a frequency band having a largest power spectrum magnitude from the one or more power spectrum frequency samples, the frequency band having an upper frequency limit and a lower frequency limit.

P.S. This reference also discloses "masking" == "suppressing" == "removing" frequency bands as recited in Claim 1 and may be useful in 103 analyses. I do not see display in this case and this case is directed to noise removal not necessarily tuning - but that hardly matters.

share|improve this answer
add comment

The Secret is BandPass is No Secret!!

FREQUENCY FILTERING in practice

version 1.0 released 29/1/99 http://www.xsgeo.com/course/filt.htm

Since this patent focuses around peak selection and filtering, and has no merit otherwise, it's important to realize how common bandpass filtering is and it's wide use among many fields of frequency analysis (medical, television, search for extra terrestrial and of course musical instruments and sound manipulation).

In this 1999 article, it clearly states that bandpass is very common

Introduction

The commonest form of filtering is to remove unwanted frequency components from the data by bandpass frequency filtering. This may be to remove frequencies above the Nyquist before re-sampling or to remove noise types e.g. low frequency swell noise from the data. Most commonly bandpass filters are applied post-migration to improve the clarity of the display. While filters can be applied in several domains they are usually designed in the frequency domain for clarity. They may be applied in the time domain by convolution or in the frequency domain by multiplication, however this is usually transparent to the user. The user has to decide whether to apply a minimum phase or zero-phase filter and must input sufficient parameters to specify the pass or reject bandwidth.

POST-STACK APPLICATIONS

Once the data have been migrated they are usually bandpass filtered to improve the clarity of the display by increasing the signal-to-noise ratio. This is required since the higher frequencies are usually lost in the deeper sections due to various attenuation mechanisms and without filtering the clarity of the deeper section is reduced by the presence of high-frequency noise. It is normal to pass higher frequencies in the shallow part of the section and filter high frequencies in the deeper part.

If the migrated data have been converted to zero-phase then zero-phase filters may be used, otherwise minimum phase filters should be used.

Written in 1999 Describes methods of '060 and it's description section near perfect

Common Filters to include BandPass

share|improve this answer
add comment

NON-CITED Prior Art (1999)

ELECTRONIC GUITAR TUNER

by Kathleen Wettstein and Adam Wunderlich May 4, 1999

http://courses.engr.illinois.edu/ece445/projects/.../project45_final_paper.doc

I found this old 1999 document on 3 examples of basic guitar tuners. This document also has considerable mention of peak detection and only allowing portions of the signal through above a predetermined threshold

The FFT was implemented as a radix 2, decimation in time algorithm. The FFT code used was that made available to students in ECE 320. Since the FFT was a decimation in time algorithm, the complex output was in bit-reversed order. Consequently, the program reordered the output in the correct order, and placed the magnitude squared of the FFT data in an array. Next, the FFT data array was searched for the fundamental frequency. For reasons mentioned earlier, the search algorithm began at the ninth sample in the array. The search was executed by first comparing each data value to an FFT threshold value. *The threshold was chosen so that only peaks in the FFT surpassed it. If the output sample was above the threshold, the program executed a subroutine to identify if the data sample was indeed a peak. A point was considered a peak if the next value in the array was less. *

If a point was identified as a peak, it was considered to be the fundamental. The index of the fundamental in the FFT output array was used to reference two look-up tables. The first look-up table gave the pitch, and second gave the cents out of tune.

It searches for a peak in sound (above a predetermined threshold) and then outputs a display of figures based on frequencies calculated through FFT.

Claim 13 '060 Invalidated?

Figure 5. Block Diagram for DSP program

Figure 5.  Block Diagram for DSP program

Page 8: Pitch Detection

Motorola’s DSP5630x family of processors was chosen because of its availability and familiarity of the designers with the product. Also, this family of DSPs is widely used because of their versatility and capacity for operations such as filtering, and Fast Fourier Transforms (FFTs).

Discusses filtering like it's just another part of audio signal processing.

share|improve this answer
1  
"Is magnitude above threshold?" == "Suppressing a display of frequencies or musical notes from a subsequent signal that deviate from the filter mode reference frequency or musical note by a predetermined threshold." How is this novel if using a bandpass filter? Or anything with a high/low limit which would ultimately have a center (target frequency)? –  Aron Stein Dec 19 '13 at 0:19
add comment

Can You Patent Human Auditory Senses?

I believe they just patented basic human auditory senses. The Claims in '060 discuss selecting frequencies above a predetermined threshold, and suppressing the rest. Likewise, Claim 1 discusses bandpass filtering around a target frequency. This is exactly what the natural auditory senses of the human ear and brain do.

Auditory Masking - Wikipedia

http://en.wikipedia.org/wiki/Auditory_masking

"Off frequency listening is when a listener chooses a filter just lower than the signal frequency to improve their auditory performance. This “off frequency” filter reduces the level of the masker more than the signal at the output level of the filter, which means they can hear the signal more clearly hence causing an improvement of auditory performance."

Am I to assume that anyone recognizing a pitch would be infringing?

share|improve this answer
add comment

Non-Cited Prior Art

Visualization of Musical Pitch

Philip McLeod and Geoff Wyvill.

In Computer Graphics International 2003, 2003

http://miracle.otago.ac.nz/tartini/papers/Visualization_of_Musical_Pitch.pdf

2.5 Choosing the Fundamental

After the frequencies of all the local maxima have been found, their amplitudes are calculated. The strongest frequency is selected and assumed to be one of the harmonics from the note being played. The fundamental is not always the strongest frequency in a note. Figure 4 shows an example of a note where the fifth harmonic is the most powerful.

The pitch of the note is related to the “dominant” frequency. In this case, dominant means what a musician perceives to be the main frequency. This is not necessarily the lowest or the one with the largest amplitude. The following process appears to agree with our subjective observation al- most all of the time. Firstly we pick out the frequency, f, with the largest am- plitude. This is almost certainly a true harmonic of the fundamental we are seeking. In other words, the fundamental frequency is F = f/n where n is an integer. For each value of n from one to ten we examine the spectrum to see how many frequencies are potential harmonics of F. A frequency is a potential harmonic if it is close the ideal frequency of the harmonic. We actually calculate a score for each F = f/n which is a weighted sum of closeness measures. The more peaks that are separated by F the better the fit.

Figure 4

This example simply demonstrates that selecting a peak frequency for display and suppressing the other frequencies is common practice. Claim 13 is not novel, and very obvious to anyone in spectral analysis.

share|improve this answer
    
This looks like the idrumtune app picture re: toulson! –  Tyler Ulrich Dec 16 '13 at 21:50
    
This certainly proves that the '060 patent lacks novel, when nearly every subject matter on FFT, pitch detection, etc talks about selecting a peak frequency and uses filtering means to remove DC offset, outside noise and limit the display of pitch to only the useful values. –  Aron Stein Dec 17 '13 at 14:41
add comment

Non-Cited Prior Art

The Scientist and Engineer's Guide to Digital Signal Processing

DSPGuide.com

1997-2011

This is an online, paper, hard copy publication for Digital Signal Processing. It's know as one of the defacto sources of information in relation to signal processing.

Chapter 26: Target Selection

Discusses several aspects and reasons why you would choose values over a predetermined threshold as Claim 13 of '060 claims rights to. In this example, a device for reading cancer in a patient is discussed:

This method of converting the output value into a probability can be useful for understanding the problem, but it is not the main way that target detection is accomplished. Most applications require a yes/no decision on ...

In short, we need a machine that can carry out a multi-parameter space division, according to examples of target and nontarget signals. This ideal target detection system is remarkably close to the main topic of this chapter, the neural network.

Chapter 14:

http://www.dspguide.com/ch14/5.htm

(Just for clarification on Bandpass filters, what they are, how they are comprised, and how common they are in audio.

High-pass, band-pass and band-reject filters are designed by starting with a low-pass filter, and then converting it into the desired response. For this reason, most discussions on filter design only give examples of low-pass filters. There are two methods for the low-pass to high-pass conversion: spectral inversion and spectral reversal. Both are equally useful.

...

BandPass Filter Construction

Lastly, Figs. 14-8 and 14-9 show how low-pass and high-pass filter kernels can be combined to form band-pass and band-reject filters. In short, adding the filter kernels produces a band-reject filter, while convolving the filter kernels produces a band-pass filter. These are based on the way cascaded and parallel systems are be combined, as discussed in Chapter 7. Multiple combination of these techniques can also be used. For instance, a band-pass filter can be designed by adding the two filter kernels to form a band-pass filter, and then use spectral inversion or spectral reversal as previously described. All these techniques work very well with few surprises.

Chapter 17: Custom Filters

http://www.dspguide.com/ch17.htm

Most filters have one of the four standard frequency responses: low-pass, high-pass, band-pass or band-reject. This chapter presents a general method of designing digital filters with an arbitrary frequency response, tailored to the needs of your particular application. DSP excels in this area, solving problems that are far above the capabilities of analog electronics. Two important uses of custom filters are discussed in this chapter: deconvolution, a way of restoring signals that have undergone an unwanted convolution, and optimal filtering, the problem of separating signals with overlapping frequency spectra. This is DSP at its best.

This NON-CITED Prior Art online book has several examples, common solutions to all of the claims in the '060 patent. Reexamine.

share|improve this answer
add comment

This could be considered prior art, but it's an application called Drum-Tuner on both iTunes and Google Play. The iphone version was published Jul 25, 2012

enter image description here

The proof here is that the '060 is not novel, and obvious.

The '060 Application filing date is 02/15/2013, published Aug 6, 2013 The '060 patent was an extension of Application US20130139672 Published Jun 6, 2013

My thoughts are leaning towards how is an idea novel and unobvious if the application or patent was not published yet so many very close ideas such as Drum-Tuner and apparently a few other software apps like Toulson's iDrumTune came out which now technically infringe upon 060 due to its broad claims.

Anyone?

share|improve this answer
    
I wouldn't consider this prior art, but it states obviousness. –  Aron Stein Dec 17 '13 at 21:16
add comment

I am curious to how this is NOVEL or even UNOBVIOUS. I am a recording engineer and I use common software such as Pro Tools, Logic Pro and others. Virtually all Digital Audio Workstations and recording software comes equipped with BandPass, Tuners, and Noise Gates. An end user could easily stumble and infringe upon this patent easily.

Allow me to demonstrate:

Claim 13 Infringement

  1. Open Logic Pro (or even more distributed Garage Band) with Live or Recorded Samples
  2. Select a Noise Gate on a Channel Strip
  3. Add the Tuner Plugin under Metering > Tuner > Tuner

You have now infringed upon Claim 13 of '060 Patent.

Also note if you use a recording interface that uses BandPass Filters or Noise Gates, and you simply activate the Tuner plugin on a channel strip, you have effectively infringed on this brand new patent.

Claim 13 Infringement

Claim 1 Example:

  1. Open Logic Pro (or even more distributed Garage Band) with Live or Recorded Samples
  2. Select a Noise Gate on a Channel Strip
  3. Add a BandPass Filter: Audio Units > Bandpass Filter (Adjust to range of a drum or instrument) after you determine the pitch the tuner is displaying, adjust it to that range. To enable / disable this "mode" just bypass the bandpass filter.
  4. Add the Tuner Plugin under Metering > Tuner > Tuner

Claim 1 Infringement: BandPass Filter and a Tuner

This is a very ingredient oriented piece of software. I attribute this patent to pulling out bread, ham, cheese and and assembling it and patenting that method.

share|improve this answer
1  
Seems like mixing the cake ingredients and then putting them in the oven to bake. Cake Mix! –  Aron Stein Dec 17 '13 at 20:16
2  
Very common practice to apply plugins on an audio chain. Very common to use spectral analysis to adjust EQ and filters. Very common applications all have these plugins. –  Robert Tesla III Dec 17 '13 at 20:32
2  
A very common set of audio plugins indeed. –  Tyler Ulrich Dec 17 '13 at 20:41
2  
I am not a fan of the examination process this patent was (not) subjected to, but how hard something is to do once someone tells you what to do is not relevant to novelly or obviousness. Everything is obvious in hindsight.That is why examiners look to find specific documents, published before the application, that disclose every element in a claim. –  George White Dec 18 '13 at 0:41
add comment

Non-Cited Prior Art

US4227437

Priority Date: Oct 11, 1977

Another fine example of selecting frequencies above a predetermined threshold is US4227437 Published Oct 14, 1980

Frequency detecting apparatus

  1. Frequency detection apparatus for tuning a stringed instrument or the like, comprising:

a transducer for receiving sound information from a vibrating string on said instrument and for converting the same into an electrical signal of like frequency, selection means for selecting a note corresponding to a correct frequency to be detected,

Transducer == microphone

a filter for receiving said electrical signal and responsive to said selection means for filtering second harmonic information from said electrical signal,

a first oscillator phase locked to the signal provided by said filter for generating a phase locked signal,

a second oscillator also responsive to the signal provided by said filter to produce a standard frequency output signal initiated in synchronous time relation with the signal provided by said filter,

a control counter coupled to the output of said second oscillator for counting the output cycles of said standard frequency output signal, and a gating circuit responsive to a count of said control counter for gating said phase locked signal during a gating period as determined by a predetermined counted number of cycles of said standard frequency output signal, note counter means coupled for receiving the phase locked signal as gated by said gating circuit for detecting whether the number of output cycles of the gated phase locked signal received during said gating period reaches a predetermined count corresponding to the note selected by said selection means,

and display means for producing first, second and third indications for respectively indicating substantial identity in frequency between the electrical signal and the note selected by said selection means, or whether the electrical signal is above or below said note, said display means receiving the output of said note counter means substantially at the end of said gating period produced by said gating circuit to determine whether said note counter means has made a full count corresponding to the selected note substantially at the end of said gating period.

Sure seems like claim 1 & 13 of US8502060 when you realize what a gate is.

share|improve this answer
    
Agreed, this is a very straight forward concept for anyone who is an audiophile. –  Aron Stein Dec 15 '13 at 7:26
1  
I concur, straightforward answer. –  Audio Sniper Dec 17 '13 at 20:08
add comment

Non-Cited Prior Art

4,608,993

Priority Date: Jul 31, 1984

After looking at this, it is my personal belief that there is a very strong possibility that every claim of the '060 Patent can be invalidated by an attorney. The easiest to invalidate is Claim 13. US Patent 4,608,993 is directed to a blood flow measurement device and method. On the face of it, how does a medical device relate to the '060 Patent? The '993 Patent states in the Abstract,

"The systems employ electronic techniques for providing accurate tracking of portions of the frequency spectra of Doppler shift signals to determine peak and means velocity and acceleration.”

The basic element of the system of the ‘993 patent determines a peak ultrasonic acoustic signal of a power spectra within a band and correlates this to a peak velocity. The system filters out the sound of unwanted sources, such as the heart, and detects the peak volume within a set band. The '993 Patent utilizes frequency detection or in other words pitch detection by initialization a filtering means and determines a peak value of a power spectra found by using a Fast Fourier Transform (FFT).

I am very confident that I can find references that could be used by an attorney to invalidate ALL of the claims of the '060 Patent, not just the independent claims.

To further clarify, Doppler, or Doppler Effect is essentially sound or light waves.

Doppler effect n. A change in the observed frequency of a wave, as of sound or light, occurring when the source and observer are in motion relative to each other, with the frequency increasing when the source and observer approach each other and decreasing when they move apart. The motion of the source causes a real shift in frequency of the wave, while the motion of the observer produces only an apparent shift in frequency. Also called Doppler shift.

share|improve this answer
    
Nice answer! I like that.. FFT covers just about every measurement known to man. It's Public Domain too! –  Tyler Ulrich Dec 13 '13 at 4:39
    
FFT is a specific algorithm for computing a Fourier transform. Fourier proved that any repeating waveform in time could be equally expressed in the frequency domain. I'm sure there are thousands of specific inventions that make use of the FFT. –  George White Dec 13 '13 at 5:17
1  
This probably has claims that should never have been allowed, particular claim 13, but just saying "FFT" doesn't help. –  George White Dec 13 '13 at 5:19
1  
That's a very heartfelt example. The heart is a percussive instrument, and a resonant structure. –  Aron Stein Dec 15 '13 at 7:31
add comment

protected by Community Dec 20 '13 at 15:16

Thank you for your interest in this question. Because it has attracted low-quality answers, posting an answer now requires 10 reputation on this site.

Would you like to answer one of these unanswered questions instead?

Not the answer you're looking for? Browse other questions tagged or ask your own question.