Abstract:
A method of acoustic transducer calibration ( 200, 400 ) using a band limited pseudo random noise source with an internal digital signal processor ( 209, 403 ) to tailor audio characteristics of an internal microphone  103  and internal speaker ( 301 ) within a communications device ( 101 ) to insure consistent amplitude and frequency characteristics of these microphone and speaker transducer devices. The method offers and advantage such that tuning of the amplitude and frequency response consistently converges to the desired filter response with a filter type offering operational stability.

Description:
TECHNICAL FIELD  
         [0001]    This invention relates in general to acoustic calibration and more specifically acoustic calibration for speaker and microphone anomalies as used in communications equipment.  
         BACKGROUND  
         [0002]    Many portable communications devices use some variety of transducer. A transducer can include such devices as a microphone to convert acoustic energy to electrical energy or a speaker to convert the electrical energy back to acoustic energy. Ideally, it is important to achieve some type of predetermined frequency response and gain from these devices in order for the communications device to operate most effectively. A transducer with a wide frequency response enables a compete spectrum of audio frequencies to be reproduced which are typically between 300 to 3000 Hertz (Hz). However, the acoustic responses of these transducer devices unfortunately are non-ideal, inconsistent and often have poor operational characteristics. This is due to such things as environmental factors, the mechanical placement of the transducer and/or variations in their manufacture.  
           [0003]    For example, a typical microphone used in a two-way radio device often can have a gain of +/−3 decibel (dB) as specified by most manufacturers. In the design and operation of two-way radio or cellular devices, this can make it difficult to electrically balance audio to the input circuitry of the device. This is due to wide variations in both microphone gain and frequency response. This same example is also applicable to the communications speaker output which often causes a user using numbers of similar types of communications equipment difficulty in maintaining a similar operating radio when comparing two devices. More often than not, this causes the user to falsely determine that a radio is defective when in-fact only slight acoustic variations in operation between either microphone or speaker cause each radio to sound differently to the user.  
           [0004]    Therefore, the need exists to provide a system for acoustic microphone and speaker calibration that will enable an electronic device to operate consistently regardless of slight operational dissimilarities between the microphone and speaker components. 
       
    
    
     BRIEF DESCRIPTION OF THE DRAWINGS  
       [0005]    [0005]FIG. 1 is a block diagram showing acoustic calibration of a microphone in a portable communications device.  
         [0006]    [0006]FIG. 2 is a block diagram showing the method of acoustic calibration of a microphone according to the preferred embodiment of the invention.  
         [0007]    [0007]FIG. 3 is a block diagram showing the acoustic calibration of an internal speaker in a portable communications device.  
         [0008]    [0008]FIG. 4 is a block diagram showing the method of acoustic calibration of an internal speaker according to the preferred embodiment of the invention. 
     
    
     DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENT  
       [0009]    Referring now to FIG. 1, a portable two-way communications device  101  such as a two-way radio or cellular telephone includes an internal speaker and internal microphone  103 . In the preferred embodiment of the invention, during the acoustic calibration of a microphone  103 , a characterized external speaker  105  is attached to the communications device  101  that is used to produce audible pseudo random noise generated by an internal digital signal processor (DSP). The pseudo random noise is directed toward the microphone  103 . As is well known in the art, acoustic band limited pseudo random noise is often referred to as “pink noise” and is audio generated over the audible frequency range of 300 Hz to 3 KHz.  
         [0010]    [0010]FIG. 2 depicts a block diagram showing the method of acoustic calibration of the microphone  103  according to the preferred embodiment of the invention. Pseudo random noise  201  is generated and supplied to a filter  203 . The pseudo random noise can be generated either internally from the communications device or from an external source. The filter  203  acts to tailor the frequency response of the external speaker  105  in order to provide optimized frequency and gain characteristics for microphone calibration where “h” is the frequency response of the speaker and “1/h speaker” is the inverse frequency response. 1/h speaker is used to denote the combination of frequency responses to produce a “flat” frequency response. Thus, filter  203  effectively normalizes the frequency and gain response of the speaker  105  used for calibration of the microphone  103 . DSP  209 , as discussed hereinafter, is the actual device the optimizes the characteristics of microphone  103 .  
         [0011]    The amplitude of the pseudo random noise coming from speaker  105  is sufficient enough such that it is supplied to the input of microphone  103 . Although microphone  103  is shown as an internal microphone, it will be evident to those skilled in the art the an external speaker microphone, such as a speaker microphone, could be calibrated using this method as well. The output of the microphone  103  is directed to a digital signal processor (DSP) type audio filter  209 . As is well known in the art, the DSP  209  acts to transform the analog microphone input and convert it to a digital signal where it can be easily processed and manipulated to add, remove or alter its signal characteristics. These signal characteristics include but are not limited to amplitude or frequency components.  
         [0012]    In order to control the DSP filter  209 , a comparison  211  is made between the output of the pseudo noise signal which represents a “desired” signal (d) and an output of the DSP filter  209  (y). A delay  213  is provided to the pseudo random noise generator so as to allow proper synchronization between noise signals as each travels by separate paths though the audio chain. As seen in FIG. 2, this chain is comprised of speaker  10 , microphone  103  and DSP filter  209  An error signal (e) is produced at the output of the comparator  211  that is directed to the DSP filter  209 . The error signal works to control a plurality of signal coefficients in various DSP algorithms used to process the analog signal from microphone  103 . The filter coefficients are changed to provide an optimized microphone output to enable the two-way communications device to operate by having consistent gain and frequency components from the output of the its microphone  103 . It will be evident to those skilled in the art that after the calibration of the microphone  103  the DSP filter  209  will continue to use the same calculated frequency coefficients in order to provide optimized audio to the communications device  101  from microphone  103 . It is important to note that FIG. 2 represents a unique system identification adaptive microphone filter structure which converges directly to the inverse filter in a fixed input response (FIR) structure which has no stability issues.  
         [0013]    [0013]FIG. 3 illustrates a block diagram showing the acoustic calibration of an internal speaker  301  in a portable communications device according to the preferred embodiment of the invention. FIG. 3 shows the portable communications device  101  with internal speaker  301  that is typically located within the device. As will be evident to those skilled in the art, although the discussion herein will be directed to an internal microphone, calibration of an external microphone or speaker such as a handheld public safety microphone would also be possible using this method.  
         [0014]    In order to calibrate the internal speaker  301 , pseudo random noise is delivered from the speaker  301  at an amplitude such that it can be detected either by the calibrated internal microphone  103  or an external microphone  303 . Moreover, as shown by the block diagram in FIG. 4, the pseudo random noise may be generated either by the internal DSP or an external source. After detection by the external microphone  303 , the detected audio is then filtered by filter  406  in order to obtain the desired amplitude and frequency response from the microphone  303 . As noted previously, “h” denotes the frequency response and “1/h mic” is the inverse frequency response of the microphone. Both the h response and 1/h response are combined to produce a “flat” response.  
         [0015]    Filter  203  effectively normalizes the frequency and gain response of the speaker  105  used for calibration of the microphone  103 . DSP  209  is the actual device the optimizes the characteristics of microphone  103 . Preferably the external microphone  303  has already been previously calibrated according to the methods as defined herein. The output (y) of the filter  401  is then compared  405  with the pseudo noise generator  201  (d).  
         [0016]    The output of the pseudo noise generator  201  is delayed  407  before comparison in order to insure the timing and synchronization is correct between both noise signals as they travel though the audio chain of the portable communications device. Based on this comparison, an error signal (e) is produced at the output of the comparator  405  that is directed to the DSP filter  403 . As with the microphone calibration, the error signal works to control a plurality of signal coefficients in the DSP algorithms used to process the analog signal before entering speaker  301 .  
         [0017]    The filter coefficients are then changed to provide an optimized speaker input to enable the internal speaker  301  in the two-way communications device to operate by having consistent gain and frequency components from the output of the its speaker  301 . It will be evident to those skilled in the art that after the calibration of the speaker  301  the DSP filter  209  will continue to use the same calculated frequency coefficients in order to provide optimized audio to the communications device  101  from speaker  301 . It is important to note that FIG. 4 represents a unique system identification adaptive speaker filter structure which converges directly to the inverse filter in a fixed input response (FIR) structure which has no stability issues.  
         [0018]    While the preferred embodiments of the invention have been illustrated and described, it will be clear that the invention is not so limited. Numerous modifications, changes, variations, substitutions and equivalents will occur to those skilled in the art without departing from the spirit and scope of the present invention as defined by the appended claims.