Abstract:
A system and method of digitally modeling and significantly reducing the direct coupling between a telephone appliance loudspeaker and microphone employs a handset microphone in a phone appliance, such as, for example, a speakerphone, as a reference channel along with the speakerphone microphone to reduce the direct coupling between the speakerphone loudspeaker and the speakerphone microphone. Using an analog adaptive echo cancellation scheme, the sensitivity and dynamic range of a telephone device microphone to speech from talkers in an enclosure but not from signals emitted by a telephone loudspeaker are enhanced. Nonlinear distortions produced by a telephone appliance loudspeaker are measured by a telephone appliance handset microphone and used in a feedback subsystem, e.g., network, to reduce nonlinear distortions in the telephone appliance loudspeaker and loudspeaker driving circuit.

Description:
FIELD OF THE INVENTION  
       [0001]     This invention is directed, in general, to acoustic coupling in communication systems and apparatus, and more particularly to reduction of direct coupling of a speakerphone loudspeaker to a speakerphone microphone.  
       BACKGROUND OF THE INVENTION  
       [0002]     Speech typically results in reflected waves. When the reflected wave arrives a very short time after a direct sound, it is perceived as a spectral distortion or reverberation. However, when the reflection arrives a few tens of milliseconds (ms) after the direct sound (i.e., a relatively long period of time), it is heard as a distinct echo. Such echoes may be annoying, and under extreme conditions can completely disrupt a conversation.  
         [0003]     Acoustic echoes typically occur in telecommunications networks due to acoustic coupling between, for instance, a loudspeaker and a microphone (e.g., in a speakerphone). During a teleconference, where two or more parties are connected by a full-duplex link, an acoustic reflection of the far-side talker through the near-side conference room is returned to the far-side talker as an echo. Acoustic echo cancellation tends to be more difficult than network line echo cancellation since the duration of the acoustic echo is usually several times longer (100-400 ms) than typical electrical line echoes (20 ms). In addition, the acoustic echo may change rapidly at any time due to opening doors, moving persons, changing temperatures, etc., within the conference room. In other words, environmental factors may tend to exacerbate the acoustic echo heard through such devices making them more problematic to offset than their line echo counterparts.  
         [0004]     In any speakerphone appliance, the speakerphone&#39;s microphone picks-up sound produced by the speakerphone&#39;s loudspeaker. In order to prevent or reduce the chances of the far-end caller at the other end from hearing echoes of his/her own speech, the speakerphone algorithm, which controls the loudspeaker and microphone, operates to disable or sufficiently attenuate the microphone&#39;s signal path to the loudspeaker when sound is emanating from the loudspeaker. To reduce the need to attenuate the microphone&#39;s signal, and thereby improve the subjective performance of the speakerphone, current state-of-the-art “full-duplex” speakerphones employ adaptive digital filters to model the loudspeaker-to-microphone acoustic path and digitally subtract the modeled echo from the microphone path before it is transmitted back to the far-end part.  
         [0005]     Echo cancellers are used to cancel acoustic echoes. Typically, single-path echo cancellers include an adaptive filter and a subtractor. In operation, an incoming signal, for example in a conventional speakerphone, is received from a far-side talker and is heard through a speaker by a near-side talker. Unfortunately, the incoming signal is also received through the near-side microphone, which is typically positioned close to the near-side speaker. The incoming signal heard back through the near-side microphone results in an acoustic echo, which is then heard by the far-side talker. To combat this echo, the incoming signal is also applied to the adaptive filter when it first enters the echo canceller; the adaptive filter generates a replica signal of the incoming signal heard by the microphone in an attempt to model the echo signal. To accomplish this, the replica signal and the intended outgoing microphone signal, which includes the echo signal, are applied to the subtractor. The subtractor subtracts the replica signal from the outgoing microphone signal in an effort to eliminate or “cancel” the echo signal.  
         [0006]     The resulting signal, after the cancellation, is called an error signal, since it may be analyzed to determine how much of the echo signal remains after cancellation. The error signal is fed back to the adaptive filter, which adjusts its internal filter coefficients in order to maximize cancellation of the echo signal and minimize the error signal. In this manner, the filter coefficients converge (hence, an “adaptive” filter) toward values that optimize the replica signal in order to cancel, at least as much as possible, the echo signal.  
         [0007]     The use of a short analog canceller to increase dynamic range in telephone appliances is shown, for example, in U.S. Pat. No. 6,147,979, entitled, “System and method for echo cancellation in a communication system,” and in U.S. Pat. No. 6,321,080, entitled, “Conference telephone utilizing base and handset transducers.” 
         [0008]     The &#39;979 patent provides an echo canceller for reducing an echo component in an analog output signal caused by cross coupling of an incoming received signal received at a communication unit with the outgoing analog output signal, comprising 1) a digital echo canceller for digitally processing the analog output signal after being converted to digital form to reduce the cross-coupling echo component before transmission from the communication unit, and 2) an analog echo canceller connected between the communication unit output and the digital echo canceller for analog reduction of the cross coupling echo component in the analog output signal when in analog form before digital processing by the digital echo canceller.  
         [0009]     The analog echo canceller of the &#39;979 patent includes means for generating an analog cancellation signal which is substantially the same as the acoustic coupling echo component in the analog output signal before digital processing by the digital echo canceller and means for combining the analog echo cancellation signal with the analog output signal to reduce the cross coupling echo component before digital processing by the digital echo canceller. The combining means produces an echo-reduced analog output signal and includes means interposed between the combining means and the digital echo canceller to automatically amplify the echo-reduced analog output signal to a level for substantially maximum resolution digitization of the echo reduced analog output signal. The means of automatic amplification includes a variable gain amplifier, means for setting the amplifier gain at a level below a digitization saturation level, means for training the analog echo canceller until maximum echo reduction is achieved and means for increasing the gain of the variable gain amplifier after training is completed to obtain maximum digital resolution during conversion. The gain increasing means includes means for monitoring the maximum magnitude of the echo-reduced analog output signal relative to a peak conversion level and means for automatically adjusting the gain to maintain a preselected rate of repetitive conversions substantially at the peak conversion level.  
         [0010]     FIG. 4 of the &#39;979 patent, which is the prior art  FIG. 1  of this Application, includes digital filter components  96  and  98 , another D/A converter  100  for converting the output signal from filter component  98  and a gain control component  102 , all included within the digital signal processor  84 , and a summing amplifier, or summing node,  104  and a variable gain input amplifier  106  and a variable gain output amplifier  105  which are discrete from the digital signal processor  84  and alternatively integral with the digital signal processor  84 .  
         [0011]     The digital echo canceller  86  digitally processes the analog output signal from microphone  78  and variable gain input amplifier  106  after it is converted to digital form by the A/D converter  92  to digitally reduce the residual cross coupling echo component before transmission from the digital echo canceller  86  by D/A converter  94  and variable gain output amplifier  105 . However, the summing node  104  and variable gain input amplifier  106  of an analog echo canceller  109  are connected between the microphone output  80  and the digital echo canceller  86  for analog reduction of the cross coupling echo component in the analog output signal appearing on output  80  when in analog form before digital processing by the digital echo canceller  86 .  
         [0012]     The filter component  98  digitally filters the incoming signal from the A/D converter  88  to generate a digital cancellation signal which is converted to an analog cancellation signal by D/A converter  100  which is substantially the same as the acoustic cross coupling echo component in the analog output signal at output  80 . The analog cancellation signal is the negative of the echo component and summing amplifier  104  combines by summing the analog echo cancellation signal at an input  105  with the analog output signal from microphone output terminal  80  to reduce the cross coupling echo component  26  before digital processing by the digital echo canceller  86 .  
         [0013]     After the echo component has been reduced, the gain control component  102  of the digital signal processor  84  controls the variable gain input amplifier  106  to automatically amplify the echo reduced analog output signal at the output of summing amplifier  104  to a level for substantially maximum resolution digitization of the echo reduced microphonic analog output signal on output terminal  80 . The gain control component  102  automatically controls the variable gain output amplifier  105  to compensate for changes in gain in the variable gain input amplifier  106  to maintain a uniform gain through the system. This improved resolution also enhances operation of the digital echo canceller  86  in digitally removing any residual echo component remaining after analog cancellation for there are more bits available for digitization and improved resolution of the residual echo components.  
         [0014]     In order to make the analog cancellation signal to be the same as the cross coupling echo component, it is necessary to digitally process the signals with the digital filter components  96  and  98 . The cross coupling path from D/A converter  90 , speaker  74  and microphone  78  to the output  112  of A/D converter  92  has a scaled digital impulse response represented by a(n) while the feedback path of the analog cancellation signal from D/A converter  100 , summing amplifier  104  to the output  112  of A/D converter  92  also has a scaled impulse response represented by the function f(n).  
         [0015]     In speakerphone appliances, the loudspeaker signal at the speakerphone microphone is very large and, as a result, can saturate the analog-to-digital converter servicing the speakerphone microphone. By performing some degree of analog echo cancellation, the signal level can be reduced prior to reaching the analog-to-digital converter. This is what happens in the &#39;979 patent.  
       SUMMARY OF THE INVENTION  
       [0016]     The systems and methods according to the invention help to increase the performance of a speakerphone appliance by using a handset microphone of the speakerphone appliance to digitally model and significantly reduce the direct coupling of the speakerphone loudspeaker to the speakerphone microphone.  
         [0017]     The systems and methods according to the invention use a simple, short-length adaptive filter to remove the direct coupling, which the handset microphone effectively measures. By doing so, the duplex performance of the speakerphone can be improved by increasing the accuracy of a double-talk detector used to detect when the near- and far-end talkers are speaking simultaneously.  
         [0018]     The systems and methods according to the invention use the close proximity of the handset microphone to the speakerphone loudspeaker to allow measurements of nonlinear distortions produced by the loudspeaker when used at high volume settings. The systems and methods according to the invention use such measurements in a feedback network to reduce the nonlinear distortion in the loudspeaker and the loudspeaker driving circuit (amplifier clipping for instance), thereby improving sound quality at the loudspeaker.  
         [0019]     The systems and methods according to the invention also use an analog adaptive cancellation scheme to increase the sensitivity and dynamic range of the microphone signal to speech from talkers in the room and not from signals emanated by the loudspeaker.  
         [0020]     The systems and methods of the invention perform a degree of analog echo cancellation to reduce the signal level prior to an analog-to-digital converter using a speakerphone appliance handset microphone signal as a reference channel applied to an echo canceller.  
         [0021]     The systems and methods of the invention use a linear filter to predict what the acoustic signal will be in a speakerphone appliance for the direct path from the loudspeaker to the speakerphone microphone to effectively use acoustic echo cancellation even in the presence of a distorted loudspeaker signal, that includes both amplifier distortion as well as loudspeaker distortion.  
         [0022]     The systems and methods according to the invention also reduce distortion caused by chassis rattle and buzz, which is not handled by current acoustic echo cancellation schemes.  
         [0023]     The systems and methods according to the invention adaptively measure the acoustic path from the handset to the speakerphone microphone. Moreover, because this path will not change much due to the fixed geometry of the speakerphone system, this acoustic path may be measured over a long period of time.  
         [0024]     The systems and methods according to the invention use the predicted signal from the adaptive filter to generate a canceling signal that is converted to an analog signal.  
         [0025]     The systems and methods of the invention use an analog subtraction circuit to reduce the direct coupling, sample this subtracted signal, and use this signal in an acoustic echo cancellation scheme to reduce the long tail echo of the room.  
         [0026]     The systems and methods of this invention use the handset microphone to identify a linear path to the speakerphone microphone that allows for the cancellation of direct path energy, even from a signal that is distorted by the loudspeaker or associated electronics.  
     
    
     BRIEF DESCRIPTION OF THE DRAWINGS  
       [0027]     The foregoing advantageous characteristics and features of the systems and methods according to the invention will be explained in greater detail and others will be made apparent from the detailed description of the exemplary embodiments of the systems and methods of the present invention which is given with reference to the several figures of the drawing, in which:  
         [0028]      FIG. 1  is a functional block diagram of a PRIOR ART digital echo canceller employed to cancel acoustic cross-coupling echo components in a telephone appliance communication system;  
         [0029]      FIG. 2  is a functional block diagram of a first exemplary embodiment of an (a two-stage digital) echo canceller and receive-path distortion compensation according to the systems and methods of this invention;  
         [0030]      FIG. 3  is a functional block diagram of a second exemplary embodiment of an echo canceller with receive-path compensation for loudspeaker distortion according to the systems and methods of this invention;  
         [0031]      FIG. 4  is a functional block diagram of a third exemplary embodiment of an echo canceller of the present invention using a reference signal derived from a second microphone according to the systems and methods of this invention in a speakerphone telephone system;  
         [0032]      FIG. 5  is a functional block diagram of a fourth exemplary embodiment of an echo canceller of the present invention using a reference signal derived from a second microphone according to the systems and methods of this invention in a speakerphone telephone system; and  
         [0033]      FIG. 6  is a functional block diagram of a fifth exemplary embodiment of an (a two-stage digital) echo canceller of the systems and methods according to the invention using a reference signal derived from a second microphone and having receive-path distortion compensation. 
     
    
     DETAILED DESCRIPTION  
       [0034]     In  FIG. 2 , a signal from a far side, such as, for example, a network (not shown) is received at input terminal  110  and is received by distortion compensator  1010 . Distortion compensator  1010  outputs a signal which, at node  1020 , is directed to both echo canceller A  1100  and to voice activation detector (VAD)  1040 . From the VAD  1040 , the signal is output to digital-to-analog converter (D/A)  1050 , and then to loudspeaker  1300 . Voice activation detector (VAD)  1040  examines its incoming signal and determines if it contains significant energy and is likely to be speech rather than non-speech, for example, noise. The significance is determined by a configurable parameter as is known in the art. If the voice activation detector  1040  determines that the input signal contain speech, it outputs an “adapt” control signal to echo canceller B,  1200 . If the input signal is determined not to contain speech, the voice activated detector  1040  outputs a “do not adapt” control signal to the echo canceller B,  1200 . The purpose of the “do not adapt” control signal is to prevent adaptation of echo canceller B,  1200 , in the absence of speech in the receive path from the network,  110 . The purpose of echo canceller A,  1100  which may be, for example, a long-tail echo canceller, is to model the acoustic path between the loudspeaker and speakerphone microphone, while the purpose of echo canceller B,  1200  which may be, for example, a short-tail echo canceller, is to model the path between the handset microphone and speakerphone microphone.  
         [0035]     After being converted to an analog signal in D/A converter  1050 , the signal is emitted as sound by loudspeaker  1300 , which may be, for example, a speakerphone loudspeaker or a handset speaker. The sound emitted by loudspeaker  1300  is detected by handset microphone  1400  via, usually, a relatively short direct acoustic path  10  between the loudspeaker  1300  and handset microphone  1400 .  
         [0036]     The sound emitted by loudspeaker  1300  is also detected by speakerphone microphone  1500  via a, usually, relatively longer direct acoustic path  20  as well as by a number of relatively long indirect acoustic paths. The signal generated by speakerphone microphone  1500  is amplified in summing amplifier  1510  as is an analog signal input to the summing amplifier  1510  from echo canceller B,  1200 , when in operation, via digital to analog converter  1210 . Summing amplifier  1510  outputs the sum of the speakerphone microphone output and the signal output by D/A converter  1210  from echo canceller B,  1200 , if any. This summed signal is output by analog-to-digital converter  1520  to be adaptively applied to echo canceller B,  1200  via node  1530  and to be inputted to adder  1540  to be summed with the signal output by echo canceller A  1100 .  
         [0037]     A summed signal is output by adder  1540  to node  1550  after which it is adaptively applied to echo canceller A,  1100  and to codec  200  which is connected to a network (not shown).  
         [0038]     The signal output from handset microphone  1400  is digitized by A/D converter  1410  and input to distortion analyzer  1420 , which also receives far-end signals from network terminal  110 . The distortion analyzer  1420  outputs a signal in the form of a control signal, which is applied to distortion compensator  1010  to reduce and/or eliminate distortion from the signal input to distortion compensator  1010  so that the signal to be output by distortion compensator  1010  has reduced distortion or no distortion.  
         [0039]     The combination of analog-to-digital converter  1520 , echo canceller B,  1200 , the two inputs to echo canceller B,  1200 , and the handset microphone  1400  is an analog subtraction circuit according to the systems and methods of this invention.  
         [0040]     Advantages of this exemplary embodiment of the systems and methods according to this invention include effective and improved echo cancellation due to use of distortions from the speaker that are inherent in the reference channel of the first stage of the echo canceller, which comprises elements  1200 ,  1210 ,  1400 ,  1410 ,  1500  and  1510 .  
         [0041]     In  FIG. 2 , the distortion analyzer  1420  compares the far-end receive signal with the loudspeaker signal as detected by the handset microphone  1400 . The distortion analyzer  1420  derives a measure of the distortion, which may or may not be present, and generates and sends a corrective signal to distortion compensator  1010 . The distortion compensator  1010  provides a correction signal to the far-end signal to reduce distortion at the loudspeaker  1300 .  
         [0042]     In various exemplary embodiments of the systems and methods of the invention, the output of the distortion compensator  1010  is used to limit the maximum amplitude of the receive signal from the distortion compensator. Known distortion compensators, having various degrees of sophistication, may be used in this regard.  
         [0043]     The analog output signal of the handset microphone  1400  is converted to a digital signal in A/D converter  1410  and that digital signal is fed to distortion analyzer  1420  to generate a control signal to apply to distortion compensator  1010 , and is used for echo cancellation in echo canceller B  1200 .  
         [0044]     Echo canceller B,  1200  effectively models the acoustic path between the handset microphone  1400  and speakerphone microphone  1500 , while not diverging significantly during periods of doubletalk.  
         [0045]     Moreover, cancellation during the first stage of echo cancellation provides improved dynamic range at the A/D converter  1520  that receives the output of the summing amplifier  1510 , thereby helping to reduce the occurrence of saturation at the A/D converter  1520  based on relatively loud signals received by the speakerphone microphone  1500 .  
         [0046]     The exemplary embodiment of the systems and methods of this invention shown in  FIG. 3  differs from the exemplary embodiment of  FIG. 2  in that a VAD  1040  and echo canceller B,  1200  (and its output) are not employed. Thus, this exemplary embodiment is less complex than the system shown in  FIG. 2 .  
         [0047]     Like the system shown in  FIG. 2 , the system shown in  FIG. 3  reduces distortions from the loudspeaker  2300 , which leads to more effective echo cancellation. The handset microphone  2400  is used as input to the receive-path distortion compensation mechanism. The output of the receive-path distortion compensation mechanism or subsystem feeds both the loudspeaker  2300  and the echo canceller  2100 . In  FIG. 3 , the output from A/D converter  2520  is added to the echo cancellation signal generated by echo canceller  2100  at node  2540  and is subsequently sent to a network via codec  220 , as well as to echo canceller  2100  via node  2550  to adaptively update echo canceller  2100 .  
         [0048]     The exemplary embodiment of the systems and methods of this invention shown in  FIG. 4  differs from the systems and methods according to the invention as shown in  FIG. 2  in that ( 1 ) in  FIG. 4 , there is no distortion analyzer or distortion compensator, and there is one echo canceller  3100  that receives an output signal from the handset microphone  3400  as a reference for the single echo canceller  3100 . Also, in  FIG. 4 , unlike in  FIG. 2 , there is no summer to add the speakerphone microphone output signal and an echo canceller signal prior to A/D  3520 . Nevertheless, use of distortions from the loudspeaker are implicitly found in the reference channel of the echo canceller  3100 , which are used to adjust, and effectively improve cancellation of the signal sent to the network (not shown) via codec  320 .  
         [0049]     Moreover, in the exemplary embodiment of  FIG. 4 , as in the exemplary embodiment  FIG. 2 , distortions from the loudspeaker inherent to the reference channel of the echo canceller, are used and lead to more effective echo cancellation. The signal output by the single echo canceller  3100  uses as a reference the signal through analog/digital converter  3410  provided by handset microphone  3400 . Also, in  FIG. 4 , the output from A/D converter  3520  is added to the echo cancellation signal generated by echo canceller  3100  at node  3540  and is subsequently sent to a network via codec  320 , as well as to echo canceller  3100  via node  3550  to adaptively update echo canceller  3100 .  
         [0050]     The exemplary embodiment of the systems and methods of this invention shown in  FIG. 5  differs from the embodiment shown in  FIG. 2  in that, in  FIG. 5 , there is no distortion analyzer or distortion compensator, and there is no analog summing of the speakerphone microphone output signal with an echo canceller signal. Additionally, in  FIG. 5 , the echo canceller  4200  is fed the output signal of only the handset microphone  4400 .  
         [0051]     In the exemplary embodiment of  FIG. 5 , as in the exemplary embodiment of  FIG. 2 , distortions from the loudspeaker are inherent to the reference channel of the echo canceller B  4200 , which includes elements  4400 ,  4410 ,  4200 ,  4500 ,  4520  and  4540 , leading to more effective echo cancellation. In the exemplary embodiment of  FIG. 5 , the echo canceller B,  4200  effectively models the acoustic path between the handset microphone  4400  and speakerphone microphone  4500 , while not diverging significantly during periods of doubletalk. Additionally, the echo cancellation signal generated by echo canceller A,  4100  is summed at node  4560  with the signal output from summing node  4540 , which is sent to codec  420  and, at node  4570 , to adaptively update echo canceller A,  4100 .  
         [0052]     The exemplary embodiment of the systems and methods of this invention shown in  FIG. 6  differs from the exemplary embodiment of  FIG. 2  in that the signal output from the echo canceller B,  5200  is not summed with the signal provided by the speakerphone microphone  5500 . However, distortions from the loudspeaker  5300  are implicit to the reference channel of the first-stage echo canceller B  5200 , leading to more effective echo cancellation. Moreover, echo canceller B,  5200 , which may be a short-tailed echo canceller, is adapted more slowly than echo canceller A,  5100 , which may be a long-tailed echo canceller, effectively modeling the direct path between the speakerphone loudspeaker  5300  and speakerphone microphone  5500 , including loudspeaker distortions, while not diverging significantly during periods of doubletalk.  
         [0053]     Also, in the exemplary embodiment of  FIG. 6 , echo canceller B,  5200  receives its reference signal via handset microphone  5400  via A/D converter  5410  and echo canceller A  5100  receives its reference signal from the far-end receive signal as modified indirectly by A/D converter  5410 , distortion analyzer  5420 , and distortion compensator  5010 . In the embodiment shown in  FIG. 6 , the error cancellation signal generated by echo canceller B,  5200  is fed directly to the summing node  5540 , which sums the adaptive echo cancellation signal output by echo canceller B,  5200  with the signal generated by A/D converter  5520 , and forwards the summed signal to summing node  5500 , where is it combined with the output signal from echo canceller A,  5100 . Echo canceller B,  5200  is adaptively updated by the signal output from node  5540  via node  5550 , and echo canceller A,  5100  is adaptively updated by the signal output from summing node  5560  via node  5570 .  
         [0054]     Those skilled in the art who now have the benefit of the present disclosure will appreciate that the present invention may take many forms and embodiments. Some embodiments have been presented and described so as to give an understanding of the invention. It is intended that these embodiments should be illustrative, and not limiting of the present invention. Rather, it is intended that the invention cover all modifications, equivalents and alternatives falling within the spirit and scope of the invention as defined by the appended claims.