Abstract:
The present invention is directed most particularly to wireless communication systems for interconnection with telephone systems, and more particularly to methods and apparatus to suppress unwanted and annoying audio spikes or bursts that infiltrate the wireless system from ultimately reaching the user&#39;s earpiece. An embodiment of the invention is directed to a telephone headset amplifier system with a noise blanking or squelching capability, which attenuates the incoming audio signal when the incoming signal exceeds a predetermined threshold.

Description:
FIELD OF THE INVENTION  
       [0001]     The present invention is directed generally to communication systems for interconnection with telephone systems including VoIP, and more particularly to methods and apparatus to suppress unwanted and annoying primarily audio spikes or bursts that infiltrate the system from ultimately reaching the user&#39;s earpiece or other transducer.  
       BACKGROUND  
       [0002]     Communication systems for interconnection of a headset or other receiver with telephone systems are well known. Wireless system adds a new dimension to the use of specialized and report receivers interconnected into telephone (including Voice over IP VoIP networks). Such systems include a fixed transceiver which is connected to a telephone line with associated hardware to relay signals to and from a remote device. Wireless systems allow substantial freedom of movement during a telephone conversation since the user is not anchored to a telephone cord line. However, wireless systems are inherently more susceptible to static or other forms of interference. More particularly, background noise and electromagnetic interference are significant sources of poor audio quality. In addition, wireless systems are susceptible to annoying loud short-lived bursts or audio spikes.  
       SUMMARY OF THE INVENTION  
       [0003]     Given the situation described above there is a need for an improved approach to eliminate or effectively suppress annoying noise spikes or other audible audio bursts before they reach the user&#39;s earpiece, particularly, in wireless communications systems.  
         [0004]     One embodiment of the invention is directed to a telephone amplifier noise suppression device comprising a first and second electronic channel. An input signal is split equally amongst the two channels and a predetermined time delay is imposed on the signal in the second channel. If the signal in the first channel is found to exceed a predetermined threshold, then the signal in the second channel is attenuated.  
         [0005]     Another embodiment of the invention is directed to a telephone amplifier noise suppression device comprising a first and second electronic channel. The first electronic channel prefilters the audio input and attenuates its output, which is inputted to the second channel, if its input exceeds a predetermined threshold, and passes its output signal unattenuated to the input of the second channel when its input does not exceed the predetermined threshold.  
         [0006]     Another embodiment of the invention is directed to a method of suppressing noise pulses in a telephone set. An input signal is split equally and synchronously amongst two channels A and B. Channel A has a faster transit time, input to output, than channel B and the output of channel is A compared against a predetermined threshold value prior to channel B&#39;s signal arriving at its output. If channel A&#39;s signal exceeds the predetermined threshold value, then a control signal is initiated to blank the output of channel A. Blanking of signal A is continued until the output of channel B drops below the predetermined threshold value.  
         [0007]     Another embodiment of the invention is directed to a method of suppressing noise pulses in a telephone set. An input signal is split equally and synchronously amongst two channels A and B. A predetermined time delay is incorporated in channel B such that the output of channel B is not immediately available to the listener. Signal A is tested against the predetermined threshold value, if signal A exceeds the predetermined threshold a fixed time delay is initiated prior to retesting signal A against the predetermined threshold value. If after the second test, and still sooner than the time delay incorporated in channel B, signal A still exceeds the predetermined threshold value, then signal B is blanked. Signal B is continuously blanked for a predetermined time wherein signal A is then tested for a third time against the predetermined threshold value. If signal A still exceeds the predetermined threshold, then the listener is warned that the audio signal has been blanked. The blanking of signal B continues until signal A drops below the predetermined threshold value.  
         [0008]     The above summary of the present invention is not intended to describe each illustrated embodiment or every implementation of the present invention. The figures and the detailed description which follow more particularly exemplify these embodiments. 
     
    
     BRIEF DESCRIPTION OF THE DRAWINGS  
       [0009]     The invention may be more completely understood in consideration of the following detailed description of various embodiments of the invention in connection with the accompanying drawings, in which:  
         [0010]      FIG. 1  depicts a functional block diagram of the circuitry to implement a noise suppression and automatic signal level control system.  
         [0011]      FIG. 1A  illustrates a flow chart according to the present invention which discloses the steps which comprise the preferred embodiment of the invention.  
         [0012]      FIG. 1B  is a continuation of  FIG. 1A  which illustrates a flow chart according to the present invention which discloses the steps which comprise the preferred embodiment of the invention.  
         [0013]      FIG. 2  shows time domain data of the present invention, consistent with the architecture outlined in  FIG. 1 , employing the blanking network described.  
         [0014]      FIG. 3  shows time domain data of the present invention, consistent with the architecture outlined in  FIG. 1 , employing de-activation of the blanking network.  
         [0015]      FIG. 4  shows time domain test data of the present invention, consistent with the architecture outlined in  FIG. 1 , which depicts the relationship between the input signal and the output signal to the user&#39;s headset. 
     
    
       [0016]     While the invention is amenable to various modifications and alternative forms, specifics thereof have been shown by way of example in the drawings and will be described in detail. It should be understood, however, that the intention is not to limit the invention to the particular embodiments described. On the contrary, the intention is to cover all modifications, equivalents, and alternatives failing within the spirit and scope of the invention as defined by the appended claims.  
       DETAILED DESCRIPTION  
       [0017]     Generally, the present invention relates to a system, method and apparatus of automatic signal level control and noise suppression in telephone system, preferably in headset systems. A functional block diagram embodiment of an automatic signal level control system  100  for telephone headset systems is depicted in  FIG. 1 . An audio input signal  110  is received from either an operator&#39;s console or a host telephone handset jack. The input signal  110  is routed to the primary side of coupling transformer  120  whose output voltage may be controlled by adjusting variable resistor VR 2  located on the secondary side of transformer  120 . This structure functions as a signal splitter to pass the audio signal through two paths, where one path will be processed for noise suppression/gain control and the other path will be switched in and out of circuit in response to the processing of the first path.  
         [0018]     Transformer  120  may be a step-up transformer to amplify weak input signals  110  and variable resistor VR 2  may be pre-set at the factory (or user adjusted) for maximum or desired sensitivity. As mentioned earlier, systems of this type may be vulnerable to input “spikes” or bursts which may come about from switching transients within the telephone system itself or from external phenomenon such as lightning strikes near telephone lines.  
         [0019]     The signal passing through transformer  120  is effectively split into two identical signals (though not necessarily of identical signal level) traveling on a first channel or path  130  and a second channel or path  200 . In the course of normal operation, i.e., absent from the spikes or bursts mentioned earlier, the input signal  130  incident upon detector and amplifier assembly  140  is sufficiently small so as to not activate (turn-on) the detector circuitry  140 , and the following linear chain of circuitry and amplifiers (units  150 ,  160 , and  170 ) are also biased “off” such that signal  180  does not activate or turn-on the switched attenuator unit  190 . In this environment, absent from input spikes or bursts, the pathway for audio transmission follows signal  200 , which is tapped-off transformer  120  by variable resistor  125 .  
         [0020]     Signal  200  is incident upon the limiter circuitry consisting of resistor  201  and diodes  202  and  203 . Diodes  202  and  203 , which under normal operating conditions are not conducting, clip their output signal  204  to a maximum of approximately +/−0.5 volts peak when high level transients are present to protect components downstream. Diodes  202  and  203  may also normalize the amplitude of high level signals so that attenuation may be constant regardless of signal level. Signal  204  is coupled to amplifier unit  210 , which has an automatic gain control unit  220  in its feedback loop. The automatic gain control unit  220  may have a dynamic range in the neighborhood of 25 to 40 decibels (dB). The output of amplifier  210  is coupled to variable resistor  230 , which may be adjusted by the headset user and the output signal  240  is routed to the user&#39;s headset earpiece.  
         [0021]     However, in the event of a signal spike or burst in the input signal  110  sufficient that transformer output signal  130  passes a threshold level (typically in the range of −10 dBv) to activate detector  140 , the circuitry defined by elements  140  through  190 , which were “off” during the spike-less environment, now all come into play. The output of the detector and amplifier unit  140  is coupled into inverting amplifier unit  150  whose output is filtered by the resistor/capacitor pair units  151 / 152  which introduces a charging delay to prevent fast transients from blanking the audio signal. The purpose of this circuitry to prevent false triggering of the noise suppression effect. If this delay was not present, a short spike would trigger suppression When the voltage across capacitor  151  charges to approximately 1 volt, steering diode  153  begins to conduct and allows current to flow to the hold timer circuit  160 . The hold timer circuit  160  may be designed to initiate a 20 millisecond delay in its output signal to stabilize the circuitry from going into oscillations when operating in the high gain mode. The output of the hold timer circuit  160  may be inverted by amplifier  170 , whose output  180  may in turn initiate turn-on of the switched attenuator unit  190 .  
         [0022]     The switched attenuator unit  190  may attenuate the input to amplifier unit  210  to a level approximately 30 dB below the threshold of the AGC unit  220 , thereby the output of the AGC unit  220  drops by about 30 dB from its normal (i.e., the case where the signal input is less than −10 dBv) AGC&#39;d output whenever the switched attenuator unit  190  is activated. This increased attenuation in signal  204  may be sufficient that the headset user is avoided the nuisance of hearing the short audio burst that would have otherwise made its way through amplifier  210  and eventually into their earpiece. Of course, during the attenuation period, the user may hear a very weak signal, but this outcome is less intrusive and indeed safer than hearing the noise which was attenuated. In reality, the user would hear nothing useful anyway if the attenuation did not occur and could be injured if the amplitude of the noise peaked.  
         [0023]     In the event that multiple or back-to-back spikes or bursts come into the system via the input signal line  110 , the above sequence of events may be repeated as necessary to attenuate the would-be annoying signal(s) and indeed, the user would not be aware of the repeated attenuations. If the noise continues for an extended period of time, the system could be programmed to inject a pleasing tone or even a vocal warning (such as “noise control in effect” etc.) so that the user did not assume that he/she had lost the connection. Likewise, an indicator light or readout to the effect “noisy environment—noise suppression functioning” could be displayed.  
         [0024]      FIG. 2  shows time domain test data  200  taken from one embodiment, consistent with the architecture depicted in  FIG. 1 , of the present invention. Signal  202  is injected into the primary side of coupling transformer  120  ( FIG. 1 ; element  120 ) at a level below the blanking threshold of −10 dBv. At time  204 , the amplitude of injected signal  202  is increased above the blanking or squelching threshold of −10 dBv. Synchronous with this at time  204 , signal  206  begins to charge positively due to a capacitive element in the detector and amplifier unit ( FIG. 1 ; element  140 ). The charging time of the capacitive element in the detector and amplifier unit may introduce an intentional time delay to prevent fast transients from inadvertently initiating a blanking sequence. When the voltage across the capacitive element reaches a predetermined value, typically one volt, steering diode ( FIG. 1 ; element  153 ) may begin to conduct as shown in signal  208  at time  210 , which in turn may activate the hold timer circuit ( FIG. 1 ; element  160 ). The hold timer circuit may introduce a time delay, typically on the order of 20 milliseconds, prior to activating a post amplifier device ( FIG. 1 ; element  170 ) as shown in signal  212  at time  214 . The leading edge of signal  212 , at time  214 , activates the switched attenuator device ( FIG. 1 ; element  190 ) which as described earlier may introduce an attenuation on the order of 30 dB in the signal to the user&#39;s headset, thereby “blanking out” the unwanted signal burst.  
         [0025]     Care must be taken to synchronize the attenuation of signal  204  by attenuator  190  so that the moment of attenuation of that signal corresponds to the same portion (in time) of the arrival of the noise signal at that point. In this analog embodiment, this is accomplished by careful tuning of the R-C circuits in both first and second channels so that the attenuation control signal  180  attenuates at precisely the time the noise arrives at the switched attenuator  190  on  204 . Either the first channel path  130  has to have a faster response time than the second path  200  or the second channel must have a slower response time than the first, otherwise, the attenuation control signal from the first channel will misalign with the identical noise pulse on the second channel and the user will either miss valuable audio information or a portion of the noise itself.  
         [0026]     The above circuit has accomplished the desired result by analog means but of course a digital version of this concept could likewise be implemented by persons skilled in the art. In the digital version of this invention, the timing problem is solved by tagging each signal with synching information, by noting the address of the byte which corresponds to the first noise segment, or by other traditional signal means. The synchronized timing of signals in the second channel is not dependent upon R-C timing but clocking of the synching methods.  
         [0027]      FIGS. 1   a  &amp;  1   b  taken together illustrate a flow chart according to the present invention which discloses the steps which comprise the preferred embodiment of the invention. In step  100 , the input signal (see  FIG. 1 ; element  110 ) may be digitized or processed in its original analog form. In step  200 , the input signal is split 50/50 into two equal time synchronized signals B &amp; A. In step  300  an intentional short time delay is introduced into signal B to allow for comparing signal A with the predetermined threshold mentioned earlier. In step  400 , signal A is compared with the threshold and if A is less than the threshold, then B is routed to the user&#39;s earpiece, otherwise signal B is blanked in step  500  per the discussions described earlier. If B is blanked in step  500 , a wait command is issued in step  600  to allow for the input spike to dissipate. In step  700 , the input signal is digitized again and split as before, and the new A is compared with the threshold level, and if A is now reduced below the threshold signal, then signal B is routed to the user&#39;s earpiece, otherwise signal B is blanked again in step  800 .  
         [0028]     Subsequent to the second blanking of the input signal in step  800 , a message may be sent to the user in step  900  informing the user of the “blanked” condition. In step  1000 , signal A is again compared with the threshold and as before, if A is now reduced below the threshold signal, then signal B is routed to the user&#39;s earpiece in step  1001 , otherwise signal B is blanked again in step  1100 . The above interrogation of A is repeated successively until A falls below the threshold, wherein signal B will then be routed to the user&#39;s earpice in step  1200 .  
         [0029]      FIG. 3  shows test data  300 , which is a continuation in the time domain of the data, depicted in  FIG. 2 . Whereas  FIG. 2  outlined the linear series of events which ultimately led to “blanking” an input spike above the −10 dbv threshold level,  FIG. 3  depicts the series of events when the input signal is reduced below the −10 dBv threshold level, and the circuitry resumes normal audio operation without switched attenuation. Signal  302  is shown initially above the −10 dBv threshold level. At time  304 , input signal  302  is reduced below the −10 dBv threshold and synchronous with this, the capacitive element in the detector and amplifier device ( FIG. 1 ; element  140 ) begins to discharge as depicted in signal  306 . When the voltage across the capacitive element falls below approximately 1 volt, the steering diode ( FIG. 1 ; element  153 ) stops conducting as depicted in signal  308  at time  310 , this in turn deactivates the hold timer device ( FIG. 1 ; element  160 ). And finally, the switched attenuator device ( FIG. 1 ; element  190 ) is turned off, as depicted in signal  313  at time  314 .  
         [0030]      FIG. 4  shows time domain test data  400  which depicts the relationship between the input signal (see  FIG. 1 , element  110 ) and the output signal  404  to the user&#39;s headset (see  FIG. 1 ; element  240 ). Initially, the input signal  402  is injected below the threshold level of −10 dBv and the output signal  404  shows modulation indicative of normal audio transmission. At time  406 , the input signal  402  is increased above the threshold level of −10 dBv, and approximately 20 milliseconds later at time  408 , the output signal is significantly attenuated (i.e., blanked) for the entire duration of the time the input signal is shown above the threshold level. At time  410 , the input signal  402  is reduced back to below the threshold level of −10 dBv and remains below the threshold level for the remainder of the data. Approximately 150 milliseconds later at time  412 , the output signal  404  to the user&#39;s headset resumes normal audio transmission and one complete cycle of blanking and recovery has been demonstrated.  
         [0031]     The present invention should not be considered limited to the particular examples described above, but rather should be understood to cover all aspects of the invention as fairly set out in the attached claims. Various modifications, equivalent processes, as well as numerous structures to which the present invention may be applicable will be readily apparent to those of skill in the art to which the present invention is directed upon review of the present specification. The claims are intended to cover such modifications and devices.