Pitch Detection - Power Spectral Estimator to Tune Musical Instruments (Overtone Labs, Inc) - Issued Patent - PRIOR ART REQUEST

Question

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 Other apps have fearfully withdrawn from global app markets as a result of this patent.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.

• Publication number: US8502060 B2

• Publication type: Grant

• Application number: US 13/768,799

• Issued date: Aug 6, 2013

• Priority date: Nov 30, 2011

• Assignee: OVERTONE LABS, INC.

• Link to Google Prior Art Search - "Find Prior Art"

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.

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"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.

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Answer 1

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):

'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?

Answer 2

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

Answer 3

Unfortunately the 20100212475 by Toulson is listed on the face of the 8502060 patent. AFAIK that means that there is a presumption (albeit rebuttable) that this document was reviewed by the Patent Examiner and that the CLAIMS of the 8502060 application were nonetheless patentably novel and nonobvious over the Toulson reference.* In a normal 3rd party ex parte requested reexam, I don't believe that this reference can be used to raise the substantial new question of patentability required to actually open a reexam, IIRC.* HOWEVER, if the Director of Patents or her delegates Commissioner Focarino or Deputy Commissioner Hirshcfeld (or Office of Patent Quality Review) ((OR ANY EXAMINER)) were to sua sponte notice Claim 13 as prima facie invalid and under the Director's prerogative instigate a reexam, then nothing stops them from reconsidering the Toulson reference. BUT, at least with respect to Claim 13, it sure seems that performing a simple (Peak detection==loudest frequency band) of a Fourier transform (frequency spectrum) would anticipate if before 2011? or infringe if after Aug 2012??? Any highschool AP math students out there to confirm this? :) Sure seems that (aside from the computer processor in the claim -- oh right, nevermind) anyone subconsciously focusing on the characteristic timbre of a particular speaker in a cocktail party would infringe OR anticipate if anyone can find proof of a cocktail party conversation actually occurring before 2011?

Answer 4

The frequency selection bit of this all puzzles me in the sense of how they patented frequencies above a sound level.

Looking into several documents online, I discover on WikiPedia Pitch Detection Algorithm (PDA) which is the obvious: http://en.wikipedia.org/wiki/Pitch_detection_algorithm

A pitch detection algorithm (PDA) is an algorithm designed to estimate the pitch or fundamental frequency of a quasiperiodic or virtually periodic signal, usually a digital recording of speech or a musical note or tone. This can be done in the time domain or the frequency domain or both the two domains.

Reading into this even further, I see more about pitch detection in the references of this wiki article. This takes me to 2005 http://www.cs.otago.ac.nz/tartini/papers/A_Smarter_Way_to_Find_Pitch.pdf

A SMARTER WAY TO FIND PITCH

Philip McLeod, Geoff Wyvill

University of Otago Department of Computer Science

1. PEAK PICKING ALGORITHM The algorithm so far gives us correlation coefficients at integer τ. We will choose the first ’major’ peak as representing the pitch period. This is not always the maximum, which is considered the fundamental frequency. Firstly, find all of the useful local maxima. These are maxima with τ which potentially represent the period associated with the pitch.

So this proves that selecting a peak frequency is not novel or unique, and is obvious. Sorry, but this patent needs to be reexamined.

There's one more I like as well.

http://www.ime.usp.br/~mqz/Mitre_AESBR2006.pdf

Accurate and Efficient Fundamental Frequency

Determination from Precise Partial Estimates

There's several references in there about how most pitch detection is done in the frequency and time domains, and this document proposes methods for selecting the loudest partial and displaying it versus all of the partials. Same thing.

I think trying to patent these methods in the us8502060 is a bit far fetched considering techniques, common knowledge, and obviousness has been in existence since audio waves were discovered.

This is a bad method patent disguised as a Drum-Set Tuner? Do they realize the implications and overbroad nature of this patent?

Answer 5

Non-Cited Prior Art

US4823667 (Priority Date: Jun 22, 1987)

Title: Guitar controlled electronic musical instrument

Claim 1 - Predetermined Threshold

... a threshold detect unit whereby an on-signal is generated if said envelope of said string waveshape signal is greater than or equal to a prespecified threshold signal amplitude and whereby an off-signal is generated if said on-signal has been generated and said envelope of said string waveshape signal is less than said prespecified threshold signal amplitude, ...

The point of this is Claim 13 of '060 is invalid.

'060 Claim 13

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

Answer 6

Cited Prior Art - Useful for Reexamination

US Patent Application: US20100212475

Dr Rob Toulson, Priority Date July 13, 2007

This patent is being explored and compared here, for the possibility of reexamination.

MPEP 2216 says "The legal standard for ordering ex parte reexamination, as set forth in 35 U.S.C. 303(a), requires a substantial new question of patentability. The substantial new question of patentability may be based on art previously considered by the Office if the reference is presented in a new light or a different way that escaped review during earlier examination."

Since there is substantial new question of patentability already outlined on this site, this is a breakdown of Dr Rob Toulson's work. Toulson states that he abandoned the patent due to too much prior art.

US20100212475 (2007) Abstract

A device for tuning a percussion instrument or training a voice has a processor which is adapted to receive an electronic signal corresponding to an acoustic impulse produced by the percussion instrument or voice. The processor is further adapted to analyse the signal to generate one or more characteristics of the signal. The device also has a display connected to the processor which displays the generated characteristics. The one or more characteristics include any one or any combination of a time-domain characteristic of the signal, a plurality of fundamental and/or harmonic frequencies of the signal, and a frequency spectrum of the signal.

us8502060 (2011) 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 find it difficult to distinguish a difference. Toulson states:

A device for tuning a percussion instrument or training a voice has a processor which is adapted to receive an electronic signal corresponding to an acoustic impulse produced by the percussion instrument or voice.

And

The processor is further adapted to analyse the signal to generate one or more characteristics of the signal.

This could also be an over broad way of describing it, yet simple research into Toulson's work which was highly publicized in the audio field, journals, and even live presentations near Boston, MA (where the '060 patentee resides) would clearly indicate that Toulson used Peak Selection, Bandpass Filtering, and more exactly as later patented, produced, sold, and asserted on one app. Toulson is also the creator of iDrumTune app which has been out since 4/20/2012. A simple look at the '060 revision dates and you'll see that even his idea was later covered in this patent by the methods.

Dr Rob Toulson wrote a blog post on his thoughts in Nov, 2013. A few quotes:

From Quantitative Drum Tuning Blog Post - Nov, 2013

Wow - the world of drum tuning has changed very rapidly this year, but for me it's been a long time coming. Back in 2005 I started a research project to analyse drums with the intention of scientifically quantifying what was in and out of tune. To do this I wrote a piece of software which I called Percussionizer and started to demonstrate this software internationally in 2008. Percussionizer could do many novel and advanced things for analysing and tuning drums - it could record and display a drum hit and tell you the fundamental and most prominent overtone frequencies. It would tell you which musical pitch the drums were tuned to. It allowed you to set a target frequency which would bandpass filter a window around that target, removing noise and allowing the user to home in on their preferred sound. It had a mode for equalising or clearing the drumhead, which allowed you to take a number of bandpass filtered readings from around the edge of the drum and showed you the results.

He later states in this blog post:

I first demonstrated Percussionizer internationally at The Art of Record Production conference in Lowell (near Boston, Massachusetts) in 2008 and my article was published subsequently in their Journal in 2009 - "The perception and importance of drum tuning in live performance and music production". In 2009 I then showcased the Percussionizer software at the world's largest Audio Engineering Society Conference in New York. Here I gave a 90 minute tutorial to a huge audience of people interested in drum tuning and percussion acoustics (see here). During the tutorial I discussed the importance of drum tuning, described the fundamental vibration science and demonstrated the Percussionizer software in its entirety. It was a fantastic event and the audience were excited to see my novel approach to tuning and analysis. This tutorial was audio recorded and can be purchased from the AES website here. I also published my tutorial slides on my own website here. At the end of the tutorial I explained that my next step was to develop an iPhone app, which would bring this software to a mass marketplace. More recently I've presented my drum and drum tuning research in the UK in 2010 "Fine tuning percussion – a new educational approach" and in the highest regarded acoustics journal in the world, The Journal of the Acoustical Society of America "Analysis and manipulation of the modal ratios of cylindrical drums". A number of my publications have been in collaboration with Dr Phill Richardson, of Cambridge University, who's PhD I supervised in exactly this field of quantitative drum tuning - he's an international expert in all this too!

Prior to demonstrating my software I applied for a worldwide patent on the new drum tuning approach - method, software and apparatus. The patent was first applied for in 2007 and went fully public in 2010. Here, you can read my patent - Tuning or Training Device - which covers all the aspects of the Percussionizer software and hardware design.

Toulson released his iDrumTune App on 4/20/2012 pre-dating the iDrumTech App (2/8/2013) and pre-dating the revisions of the '060 patent on 2/15/2013.

How Was This Prior Art Allowed?

Answer 7

Non-Cited Prior Art Found

ACOUSTIC ANALYSIS AND TUNING OF CYLINDRICAL MEMBRANOPHONES

Authors: PHILLIP, GILES, MACGILLIVRAY, RICHARDSON

Date: Aug-2010

Anglia Ruskin University

This document is over 100 pages and contains several references to Toulson, and others. It also depicts the peak selection (predetermined threshold in '060) and filtering means. This document was not cited, and belief is that it can easily invalidate '060 Claims 1&13 easily.

Page 59,60

However, existing electronic tuning devices only allow a single fundamental frequency to be detected and evaluated as in tune, such as the device described by Miesak (1984). This data is sufficient for tuning instruments such as guitar and piano which have a strong fundamental frequency. However, more complex sounds such as those from percussion require details of the fundamental frequency, higher partials, relative harmonic strengths and the time-domain envelope in order to tune the drum.

A novel percussion analysis and tuning tool has been developed and patented by Rob Toulson (Toulson, 2007). The new analysis system is being utilised in this research and, as such, becomes the first ever scientific tuning analysis of percussion instru- ments by such a method.

Page 86

The onset of the response is determined by finding the first instance of a threshold exceedance of half the amplitude of the maximum data reading.

Figure 4.11 (partial)

Since the '060 Patent talks about Zero Padding in it's description:

Page 90

“When we zero-pad a signal in the time-domain, we get a smoother look- ing frequency resolution...so adding zeroes means that there will be more frequency domain points. The analysis frequencies, as a result, become finer since they are closer together.”

Weeks (2006, p.214)

In regards to filtering:

Page 91

In addition, and prior to spectral analysis, the captured waveform data is high-pass filtered to remove any unwanted rumble, DC offset or other unwanted low-frequency components. In the current research this is performed using a 5th-order Butterworth filter with a 30-Hz cut-off frequency, and is implemented by the Matlab Butterworth filter function (MathWorks, 2010d). This processing method has been used in analysis of musical instruments previously, for example by Taguti and Tohnai (2001) who used a 4th-order Butterworth filter with a cut-off frequency of 80 Hz.

http://angliaruskin.openrepository.com/arro/handle/10540/189010 2010. Too much for me to start editing and commenting on each detail, but a very thorough chronology and bibliography of related prior art developments none-the-less.

Answer 8

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

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.

Answer 9

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));"

Answer 10

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.

Answer 11

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.

Answer 12

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.).

Answer 13

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.

Answer 14

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).

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

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.

Document ID 13768799 - Statement of Preexamination Search by Applicant

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.

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.

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.

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

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.

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

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

Answer 15

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.

Answer 16

The US4028985 Patent that I find useful from Filed on Feb 17, 1976 and Published Date: Jun 14, 1977 is very interesting indeed. It's abstract states:

An improved system for the determination of pitch or perceived frequency of signals generated by musical instruments or voices and for the display of the results in a format easily understood and used by musicians. An incoming signal from a microphone is peak detected after amplification, then further processed by rejecting spurious peaks through the use of an inhibit circuit which sends onward acceptable pulses and inhibits those pulses which arrive during a period immediately following each accepted pulse, the period being approximately 3/8 of the time between acceptable pulses. These pulses drive a period measuring circuit which provides a signal proportional to the time between input pulses, the signal being fed back to the inhibit circuit for its proper operation and being fed forward to the readout display where the results are translated into convenient musical notation, like C.sub.+.sub.1 .music-sharp.. The display further subdivides the standard 6% musical pitch increments in 2% increments so that slightly sharp or flat indications can be provided for the use of the musician.

US4028985 Claims:

1. An improved pitch determining and display system comprising, in combination:

a. signal generating means for providing a signal which corresponds to pitch produced by a musician said signal generating means comprising a microphone for picking up the sounds produced by a musician and translating the sounds into corresponding electronic signals;

b. signal processing means for determining the pitch produced said means being connected to the signal generating means and driven by the electronic signals generated by said microphone, said signal processing means comprising:

i. amplifier means connected to the microphone for amplifying the signal produced by the microphone:

ii. peak detector means connected to the amplifier means for providing an output pulse when an input peak pulse from the amplifier means exceeds a previous input pulse less an amount proportional to the time that has elapsed since the previous input pulse;

.....

c. display means connected to the signal processing means for translating the determined pitch into convenient musical notation, said display means being adapted to receive the signal from said period measuring means.

1. The pitch determining and display system of claim 1, wherein the output of the period measuring means is fed back to the peak detection means to provide the most appropriate constant of proportionality for the period of the signal entering the signal processing means.

2. The pitch determining and display system of claim 2, wherein the display means is a digital readout display for providing pitch information in convenient musical notation.

And more in Claim 9:

1. Improved electronic signal processing circuitry comprising:

a. mplifier means for amplifying an electronic signal supplied to the input thereof;

b. peak detector means connected to the amplifier means for providing an output pulse when an input peak pulse from the amplifier means exceeds a previous input pulse less an amount proportional to the time that has elapsed since the previous input pulse;

** I believe that invalidates Claim 13 of the US8502060 Patent**

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.

Answer 17

My understanding of guitar tuners from Zen-on Justina Quartz Guitar Tuners and many other tuners utilize a filter means when you select which guitar string you are tuning to. This is a chromatic tuner, but it still use frequencies to translate to musical notes which I believe the 060 patent being asserted uses the same methods.

There's a lot to be said on the WikiPedia Page for Guitar Tuners - Especially about Peterson Strobosoft Tuners which have been patented by US6580024

In Peterson's patent US6580024, they discuss in the Figures description the following, which even they admit is blatantly obvious:

Concurrently, computing engine 22 analyzes the conditioned and digitized input signal or waveform to detect a unique phase or portion of the fundamental period segments of said signal or waveform, typically a portion corresponding to the original fundamental period peaks of the input signal or waveform. There are many methods and algorithms current in the art which implement this function, the details of which are understood by those of skill in the art and are not a subject of this invention. Upon detection of said phase or portion of each fundamental period segment, or upon detection of a selected number of said phases, computing engine 22 sends commands to visual display 25 to make visible one or a plurality of image objects or patterns at positions determined by the most recently calculated displacement value.

Peterson's tuners use a method of frequency translation into musical objects, as well as musical notes which the 060 Patent claims to have invented. Adjusting the input level (Peak Detection) is not novel, nor is using a standard Noise Gate, or Gate

A noise gate or gate is an electronic device or software that is used to control the volume of an audio signal. Gating is the use of a gate. Comparable to a compressor, which attenuate signals above a threshold, noise gates attenuate signals that register below the threshold.1 However, noise gates attenuate signals by a fixed amount, known as the range. In its most simple form, a noise gate allows a signal to pass through only when it is above a set threshold: the gate is 'open'. If the signal falls below the threshold no signal is allowed to pass (or the signal is substantially attenuated): the gate is 'closed'.2 A noise gate is used when the level of the 'signal' is above the level of the 'noise'. The threshold is set above the level of the 'noise' and so when there is no 'signal' the gate is closed. A noise gate does not remove noise from the signal. When the gate is open both the signal and the noise will pass through. Gates typically feature 'attack', 'release', and 'hold' settings and may feature a 'look-ahead' function.

I'm curious to how employing a Noise Gate on a signal is novel per Claim 1 of the 060 Patent:

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

Answer 18

Non-Cited Prior Art

US5777248 (Priority Date Jul 22, 1996)

Title: Tuning indicator for musical instruments

Abstract:

A musical instrument tuning aid. The display provides the user tuning information in a rapidly interpreted form. The input note, or tuning device setting, is displayed in a clock face format. A sharp/flat indicator provides a course display for gross tuning. A display means for electronically producing a stroboscopic display between the input tone and the internally generated reference frequency provides for very accurate fine tuning with instantaneous response to pitch changes. Signal conditioning to control signal level enhances the strobe display. Tracking low pass filtering is provided to enhance strobe display for high harmonic content signals. An adjustable band pass filtering mode is provided to allow analysis of individual harmonics. A crystal timebase is used to generate a high accuracy reference frequency with fine calibration adjustment. A control means is provided to measure the input signal fundamental, select the nearest chromatic scale note, and set tuning device accordingly, allowing hands off operation. In automatic mode the note display tracks the input note providing an easy to read display indicating the note played, as well as tuning of that note.

While this tuner uses a strobe form of display, it also displays a frequency, note and octave.

Claims

1. A tuning aid for musical instruments comprising:

(a) a note/octave display comprising indication elements arranged in a pattern around two concentric circles to resemble a clock face having an inside pattern corresponding to an hour hand designating an octave and an outside pattern corresponding to a minute hand designating the twelve notes of a chromatic scale;

(b) a strobe display array consisting of a plurality of luminous elements arranged in a circular pattern;

(c) a reference frequency generating means of generating a reference frequency;

(d) a control means of controlling the reference frequency to a desired note to be tuned and controlling the note/octave display accordingly;

(e) a means of enabling each element of said strobe display sequentially at a rate such that said strobe display cycles once for each two periods of said reference frequency, each enabled element having an intensity;

(f) a means of inputting a signal having a peak level for analysis;

(g) a level control means of automatically controlling the peak level of said signal to a predetermined level and producing a level controlled output;

(h) a filtering means of receiving the level controlled output and producing a filtered output;

(i) a positive rectifying means of positive rectification of the filtered output of said filtering means and producing a positive rectified output having an instantaneous magnitude;

(j) a means of controlling the intensity of the enabled element of said strobe display according to the instantaneous magnitude of the positive rectified output of said positive rectifying means.

2. A tuning aid as in claim 1 wherein said filtering means is configured in low pass mode and is adjusted by said control means to attenuate signal harmonics higher than said reference frequency.

3. A tuning aid as in claim 1 wherein said filtering means is configured in band pass mode and is controlled by said control means to pass only a selected signal harmonic.

4. A tuning aid as in claim 1 in which the filtering means is connected to a fundamental period detection means which is connected to said control means controlling a sharp/flat indication means accordingly.

5. A tuning aid as in claim 1 in which the filtering means is connected to a fundamental period detection means which determines the input signal's fundamental frequency and is connected to said control means which automatically selects the note from the chromatic scale closest to the input signal fundamental frequency and sets said reference frequency and said note/octave display accordingly.

The first few sections of this diagram from US5777248 are the foundation of the '060 patent. The rest of this illustration is omitted on '060 patent diagrams and one can guess what the "Pitch Estimator" and other obscured items actually do. My guess is the items illustrated below.

This should be enough to invalidate claims 1 & 13 of US8502060:

US8502060 Claims 1 & 13

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.

Answer 19

Non-Cited Prior Art

Title: Digital audio signal processor employing multiple filter fundamental acquisition circuitry

US 5056398 A

Priority Date (Sep 20, 1988)

ABSTRACT

A digital signal processing apparatus is provided for identifying the octave, note and cent of a musical sound. The apparatus includes a transducer for converting the musical sound into an electrical signal, a plurality of digital detection units, each dedicated to a particular sub-scanning interval within an overall frequency interval to be scanned, receiving the electrical signal from the transducer, for determining the octave, note and cent of the musical sound by detecting a fundamental frequency of the electrical signal; and a display unit, responsive to the detection unit, for displaying the note as an alphanumeric character and the cent as a positive or negative decimal integral number from -49 to +50 with zero cents representing perfect concert pitch.

This should easily invalidate Claim 1 in regards to bandpass filtering a target frequency.

Claim 1 of '398

1. An apparatus for identifying the octave, note and degree of sharpness or flatness of a musical sound, comprising:

(a) a transducer means for converting said sound into an electrical signal;

(b) a filter means, responsive to said electrical signal provided by said transducer means, for passing a filter output signal having a frequency corresponding to a frequency of said electrical signal;

(c)** a microprocessor means**;

(d) a fundamental detection means, cooperating with said microprocessor means, for analyzing said filter output signal to identify when said frequency of said filter output signal corresponds to the fundamental frequency of said electrical signal and for providing a fundamental detection signal indicating such correspondence;

(e) control means for causing said filter means, responsive to said fundamental detection signal, to pass signals at a particular scanning frequency at which said frequency of said filter output signal corresponds to said fundamental frequency;

(f) a square wave generator means, responsive to said frequency of said filter output signal corresponding to said fundamental frequency, for providing a square wave output signal having a same period as said filter output signal;

(g) a logic circuit means, responsive to said square wave output signal, for providing an output indicating a period of said square wave output signal;

(h) said microprocessor means comprising means for comparing said output from said logic means with a look-up table including a plurality of previously calculated periods to provide an output indicating the octave, note and degree of sharpness or flatness or said musical sound; and

(i) a display means, responsive to said output of said microprocessor means, for displaying said octave, note and degree of sharpness or flatness of said musical sound, said note being displayed as an alphanumeric character and said degree of sharpness or flatness being displayed as a positive or negative number on a scale including a zero value representing perfect concert pitch and a plurality of positive or negative values on each side of zero.

2.An apparatus as in claim 1, wherein said filter means comprises a plurality of switched capacitor filters respectively provided to scan a plurality of sub-intervals of an overall scanning interval and wherein said control means comprises a plurality of dedicated center frequency filter clocks respectively provided for said plurality of switched capacitor filters to cause said switched capacitor filters respectively to sweep said plurality of sub-intervals.

I consider the above method above proves that Claim 1 of '060 is not novel. Selecting a target frequency for a bandpass filter for filtering means of the musical instrument you're tuning seems logical, obvious and not novel.

'060 Claim 1

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.

This is the same method, just broadened in description.

Answer 20

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.

Answer 21

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.

Answer 22

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 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

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.

Answer 23

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.

...

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.

Answer 24

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.

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.

Answer 25

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

Answer 26

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.

Answer 27

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...

Answer 28

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 ....

Answer 29

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).

Answer 30

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

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 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