Patent Publication Number: US-10333482-B1

Title: Dynamic output level correction by monitoring speaker distortion to minimize distortion

Description:
BACKGROUND 
     Many amplifier and speaker systems provide reasonably good quality sound reproduction at volumes below a volume distortion threshold, while providing badly distorted sound at volumes significantly above the volume distortion threshold. 
     Distortion above the volume distortion threshold may be caused by amplifier clipping or saturation, as well as nonlinearities in the speakers themselves. For example, while many speakers have speaker-cone suspension systems that act like nearly-linear springs for small cone displacements, at large cone displacements return forces are nonlinear, being stronger than would be expected of a linear spring. Other nonlinearities result from changes in inductance as the voice coil moves through the magnetic field of the speaker, and changes in speaker compartment volume as the speaker cone moves. Yet more nonlinearities result from loose or failed glue, rubbing speaker parts, dirt, and other age-related effects. 
     Speaker housings, which often double as housings for other electronics from cell phones, televisions, computers, or intercoms, may also introduce nonlinearities at high volume as housing parts and housing contents vibrate, including vibrating against each other. 
     All these nonlinearities in amplifier and speaker result in amplifier-speaker system distortion. 
     Speaker distortion often includes harmonic distortion, where second and third harmonics of large signals appear. In addition to harmonics, speaker distortion may also include intermodulation products, where a first and second input frequency mix to generate sum and difference frequencies in speaker output. Many people can hear these harmonics and intermodulation products, finding them objectionable and finding they impair speech intelligibility. 
     Speaker system distortion at high volumes can be reduced by using larger, higher-power, speaker systems with higher distortion thresholds—this raises cost of systems. 
     Apparent speaker system distortion can also be reduced by use of dynamic compression, where loud signals are detected and volume automatically reduced. Such systems typically require careful engineering of compression parameters for each speaker system, amplifier, and application. 
     SUMMARY 
     In an embodiment, a speaker system featuring automatic volume adjustment to avoid excess distortion includes a first converter for converting an input signal to an input frequency domain signal; and a second converter for converting a microphone signal to an environmental frequency domain signal. A distortion analyzer receives output of the first and second converter and determines at least one distortion level and provides at least one gain adjustment signal used to control a corrector to adjust gain of a speaker driver. 
     In another embodiment, a speaker system performs a method of reducing distortion by automatically adjusting volume in each of a plurality of frequency bands including converting an input signal to a frequency domain input signal; converting a microphone signal to an environmental frequency domain signal; determining a distortion level; generating a gain adjustment signal; and adjusting gain of a signal provided to a speaker driver. 
    
    
     
       BRIEF DESCRIPTION OF THE FIGURES 
         FIG. 1  is a conceptual block diagram of an embodiment of a system for automatically detecting speaker system distortion and adjusting volume accordingly. 
         FIG. 2  is a conceptual block diagram of an embodiment of a system for automatically detecting speaker system distortion and adjusting volume in particular volume bands accordingly. 
         FIG. 3  is a conceptual block diagram of a distortion analyzer for use with multichannel volume correction, as illustrated in  FIG. 2   
         FIG. 4  is a flowchart illustrating operation of the system. 
     
    
    
     DETAILED DESCRIPTION OF THE EMBODIMENTS 
     Single Channel Correction 
     A single-channel speaker system  100  with automatic detection and compensation for speaker-system distortion is illustrated in  FIG. 1 . In this system, an input signal  102  is fed to a frequency-banding and amplitude-detection subsystem, implemented with either a digital time-domain to frequency-domain conversion, or in some embodiments an analog filter bank  104 . In particular embodiments, the digital time-domain to frequency-domain conversion is performed by a fast Fourier transform (FFT) unit, a discrete Fourier transform (DFT) unit, or by a discrete cosine transform (DCT), each of which provide a frequency-domain representation of an input signal, A microphone  106  is located to observe sound  107  generated by loudspeaker  108 , and coupled to another frequency-banding and amplitude-detection subsystem, also implemented either with a digital FFT, DFT, or DCT unit or analog filter bank  110 . Each digital FFT, DFT, or DCT unit or filter bank  104 ,  110 , generates a frequency domain output  112 ,  114  that includes an amplitude versus frequency distribution map for its respective input signal, the input signal  102  or the microphone  106 . The amplitude versus frequency map, or frequency domain, signals  112 ,  114  are input to a distortion analyzer  116  that determines whether sound  107  generated by loudspeaker  108  has unacceptable distortion, if distortion is unacceptable, distortion analyzer  116  provides an adjustment signal  118  to a corrector  120  that adjusts volume of input signal  102  before amplification in a final amplifier and speaker drive circuit  122  that in turn drives speaker  108 . 
     Distortion analyzer  116  is based on the fact that nonlinear devices, such as amplifiers driven into saturation, and overloaded loudspeakers, produce second, third, and higher harmonics, as well as intermodulation products, all of which may appear in frequency bands where the device input may be silent. A basic distortion analyzer  116  may operate by examining amplitude in frequency bands of both the input amplitude-frequency map  112  and microphone amplitude-frequency map  114 . Distortion products can be recognized as present in a frequency band of A frequency when frequency band A of input amplitude-frequency map  112  has negligible or no signal, but the same frequency bands A of output amplitude-frequency map  114  show a significant signal, while one or more of frequency band A/2 at one-half, A/3 at one-third, and/or A/4 at one quarter of frequency A have significant signals. When distortion products in frequency band A are detected, distortion analyzer  116  concludes that whichever or all high volume signals present in the frequency bands A/2, A/3, or A/4 are being distorted. 
     When distortion analyzer  116  determines that distortion above a threshold is present, distortion analyzer  116  alters adjustment signal  118  to reduce gain in corrector  120 , thereby reducing input to speaker driver  122  and reducing amplitude at loudspeaker  108  to reduce distortion. 
     Multiband Correction 
     Large signals in one or a few frequency bands can result in considerable distortion across many frequencies, creating objectionable nose and impairing intelligibility of voice signals. 
     Intelligibility of speech is often maintained despite reduction of volume in some frequency bands. For example, speech having reduced volume at low frequencies, such as below 300 Hz, while retaining full volume at higher frequencies, retains much intelligibility. We propose a loudspeaker system  200  ( FIG. 2 ) that identifies particular frequency bands of speech having high volumes and which cause distortion, then trims volume in those frequency bands while leaving other frequency bands unaltered. 
     In this system, an input signal  202  is fed to an analog or time domain to frequency domain converter  204  for conversion to a frequency domain input  206 , converter  204  performs frequency-banding and amplitude-detection and is typically implemented with an analog-to-digital converter if the signal is analog, and a digital fast Fourier transform (FFT) unit  204 , although an analog filter bank may substitute for the FFT in some embodiments. The frequency domain input  206  is fed to a multiband corrector  208  adapted to individually control gain of each frequency band of the frequency-domain input. In an embodiment using an FFT with bandwidth from 80 to 8000 Hz using twelve millisecond FFT timeslices and 24 khz sampling, there may be 288 such frequency bands; in most systems there will be at least 100 frequency bands in the frequency-domain input signal. 
     Corrector  208  has output  210 ; corrector output  210  represents the time domain input  206  with gain adjustments applied to some, but not all, frequency bands. Corrector output  210  is fed to a frequency domain to analog domain converter  212 , typically incorporating a digital inverse-FFT unit and a digital-to-analog converter, to provide an analog signal to a speaker driver  214  that in turn drives a speaker  216 ; speaker  216  is a loudspeaker as known in the audio systems art. 
     A microphone  218  senses audio  220  emitted by speaker  216  into the environment, providing an environmental audio signal  222 , that is provided to another analog or time domain to frequency domain converter  224  for conversion to a frequency domain environmental audio signal  226 . Both the frequency domain environmental audio signal  226  and corrector output  210  are input to distortion analyzer  228 , which provides individual gain adjustment signals  230  to multiband corrector  208 . 
     In this embodiment, it is not sufficient to merely identify that distortion is taking place, it is necessary to identify particular frequency bands of the frequency-domain input signal having large volume signals that that provoke the distortion so those bands can be adjusted, while leaving other bands unaltered. 
     In order to analyze distortion and determine appropriate corrections, in our distortion analyzer  228 ,  300  ( FIG. 3 ), we construct, in the same digital signal processor that performs the FFT and inverse FFT functions, a model of the speaker driver  214 , speaker  216 , and path for audio  220  to microphone  218 , using an adaptive filter  304 . The corrector output  210 ,  302  is input to adaptive filter  304 , and adaptive filter output  306  for each timeslice is compared in a least-squares difference detector  308  to the frequency domain environmental audio signal  310 ,  226 . Least squares output of the difference detector  308  is fed to a coefficient adapter  312  that adjusts coefficients of adaptive filter  304  to minimize the least squares output of the difference detector  308  in each timeslice and thereby fit the adaptive filter&#39;s coefficients so that it correctly models distortion of the speaker driver  214 , speaker  216 , and path for audio  220  to microphone  218 . 
     The adaptive filter includes nonlinear terms and includes a multiplicity of multipliers each of which multiplies a coefficient of the adaptive filter by a term derived from an input or delayed input of the adaptive filter. 
     In a particular embodiment, the adaptive filter implements
 
 ŝ   r ( n,qk )= h   qk   H ( n,k ) s   o ( n,k ),  (1)
 
     where q=2, 3, . . . represents the q-th harmonics, q=1 represents each fundamental frequency band
 
 h   qk ( n,k )=[ h   qk ( n,k ), h   qk ( n− 1, k ), . . .  h   qk ( n−P,k )] T   (2)
 
     is a vector containing the coefficients of the adaptive filter of order P. 
     s o (n,k)=[s o (n,k)s o (n−1,k), . . . s o (n−P,k)] T  is history of the output audio, and h H  denotes the conjugate transpose of vector h. 
     Once fitting is complete for each timeslice, we extract the nonlinear coefficients from the adaptive filter, then sum, in nonlinear coefficient summer  320 , the power of the coefficients of the adaptive filter for all the harmonic distortion components for the each k-th frequency band: 
     
       
         
           
             
               
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     For calculating harmonic distortion, the fundamental frequency band is ignored, so that i starts from 2. Typically, Q ranges from 2 to 5 and the harmonic distortion from 6 onwards are usually rather small and negligible. 
     Once fit, the distortion terms for each frequency band k are compared in comparator  324  to limits T k    326  to identify frequency bands k responsible for distortion in the speaker system. A prior timeslice gain setting g k (n)  328  in a timeslice gain register  328  for each band k is decreased in gain adjuster  330  to provide a next-timeslice gain setting g k (n+1)  332 ,  230 , whenever the corresponding distortion term D k  exceeds the allowable distortion threshold for that band T k.326 . This gain register, after adjustment, is used to control gain of corresponding bands in multiband corrector  208  for the next timeslice. In one particular embodiment the gain setting g(k) is reduced by a constant for each timeslice in which distortion term D k  exceeded the allowable distortion threshold for that band T k , in another dynamic-gain adjustment embodiment the gain setting g(k) is reduced by an amount 
     
       
         
           
             
               
                 
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     where P k  (n) is the instantaneous power of the k-th input audio band. The gain of this embodiment curbs down signals with large power as P k (n) increases. Similarly, it also decreases as D k  (n) gets larger, meaning high nonlinearities. The constant C k  is a predefined constant. It is used to scale the product of D k  (n) and P k (n) to a comparable level to 1 when the k-th input audio band gets closer to the maximum value of the dynamic range. 
     In both the constant-gain-decrease on a frequency band causing distortion greater than the threshold, and the dynamic gain adjustment embodiment, gain adjuster  330  adds a small return factor to each gain of g(k) less than a standard gain G s  to bleed the adjusted gains back to standard gain over a restoration time, restoration time is typically at least several seconds to minutes. In an alternative faster-responding embodiment, a smoothing factor is applied to Dk(n) instead of using the return factor to bleed gain back to standard gain. 
     The system described with reference to  FIG. 2  applies the method  400  illustrated in  FIG. 4 . 
     At system startup, gain settings for each channel are initialized  402 , in an embodiment to initialization constants such as unity, and in another embodiment to a last gain used prior to system shutdown. An input signal  202  is converted  404  from time or analog domain to frequency domain, and the gain settings are applied  406  in corrector  208  to provide a corrected audio  210  in frequency domain. The corrected audio  210  is converted  408  to analog domain and used to drive a speaker  216 . 
     Audio from speaker  216  is received by a feedback microphone  218  and converted  410  to frequency domain. The corrected audio  210  and frequency domain microphone audio and analyzed  412  to determine distortion levels, these distortion levels are used to determine  414  new gain settings for the next timeslice. The method repeats with converting  404  more input from time or analog domain to frequency domain. 
     Combinations 
     Features of the automatic distortion-reducing speaker system described herein may be combined in multiple ways as described in the claims that follow and in the following proposed combinations: 
     A speaker system designated A featuring automatic volume adjustment to avoid excess distortion includes a first converter adapted to convert an input signal to an input frequency domain signal; a second converter adapted to convert a microphone signal to an environmental frequency domain signal; a distortion analyzer coupled to receive the input frequency domain signal and the environmental frequency domain signal, the distortion analyzer configured to determine at least one distortion level and to provide at least one gain adjustment signal; and a corrector adapted to control gain of a signal provided to a speaker driver coupled to drive a speaker. 
     A speaker system designated AA including the speaker system designated A wherein the at least one distortion level is a plurality of distortion levels, each distortion level corresponding to distortion in a different frequency band of a plurality of frequency bands, and the at least one gain adjustment signal being a plurality of gain adjustment signals each coupled to control gain of a frequency band of the signal provided to the speaker driver. 
     A speaker system designated AB including the speaker system designated A or AA wherein the first and second converter perform a fast Fourier transform (FFT), an output of the first converter is coupled as an input to the corrector, and the corrector is coupled to the speaker driver through an inverse-Fourier transform (I-FFT). 
     A speaker system designated AC including the speaker system designated A, AA or AB wherein the distortion analyzer comprises an adaptive filter coupled to receive an output of the corrector, the adaptive filter being configured by coefficients, the adaptive filter coefficients repeatedly fit to provide adaptive filter output close to the environmental frequency domain signal. 
     A speaker system designated AD including the speaker system designated A, AA, AB, or AC, wherein the at least one distortion level is determined from the adaptive filter coefficients after the adaptive filter coefficients are fit to the environmental frequency domain signal, and wherein the gain adjustment signals are adjusted according to the at least one distortion level. 
     A method designated B of reducing speaker system distortion by automatically adjusting volume in each of a plurality of frequency bands includes converting an input signal to a frequency domain input signal; converting a microphone signal to an environmental frequency domain signal; determining at least one distortion level; generating at least one gain adjustment signal; and controlling gain of a signal provided to a speaker driver coupled to drive a speaker. 
     A method designated BA including the method designated B; wherein the at least one distortion level is a plurality of distortion levels, each distortion level of the plurality of distortion levels corresponding to distortion in a different frequency band of a plurality of frequency bands, and the at least one gain adjustment signal being a plurality of gain adjustment signals each coupled to control gain of a frequency band of the signal provided to the speaker driver. 
     A method designated BB including the method designated B or BA wherein the converting an input signal to a frequency domain input signal and the converting a microphone signal to an environmental frequency domain signal are performed by performing a fast Fourier transform (FFT). 
     A method designated BC including the method designated B, BA, or BB wherein the step of determining at least one distortion level comprises fitting coefficients of an adaptive filter repeatedly to provide adaptive filter output close to the environmental frequency domain signal. 
     A method designated BD including the method designated B, BA, BB, or BC wherein determining at least one distortion level further comprises determining the at least one distortion level from the adaptive filter coefficients, and wherein at least one gain is adjusted according to the at least one distortion level. 
     Changes may be made in the above methods and systems without departing from the scope hereof. It should thus be noted that the matter contained in the above description or shown in the accompanying drawings should be interpreted as illustrative and not in a limiting sense. The following claims are intended to cover all generic and specific features described herein, as well as all statements of the scope of the present method and system, which, as a matter of language, might be said to fall therebetween.