Abstract:
A multiplexed microphone signal with multiple signal processing paths is disclosed. Each signal processing path has it own priority and other characteristics. A signal path is selected based on the application of the processed signal. Similar processes within different paths may be shared to reduce computation workload.

Description:
BACKGROUND OF THE INVENTION 
     1. Field of the Invention 
     This invention relates generally to microphone audio signal processing, particularly related to multiplexed microphone signals with multiple signal processing paths. 
     2. Description of the Related Art 
     A microphone is a basic and essential element in an audio system. There are many different applications to a variety of audio systems. The most common audio systems include, at least, the following types: a teleconference system, a public addressing (PA) system, a recording studio, or some combination of the above three. 
     A simplest teleconference system is a telephone. Two people at two physically separate locations may talk to each other through a telephone network and two telephone sets.  FIG. 1  illustrates a simplest teleconference system  100 . The teleconference system  100  has two sites, a near site and a far site. At each site, there is a telephone,  110  and  150  respectively. The two telephones are connected through a network  130 , typically a Public Switched Telephone Network (PSTN), sometime referred to as Plain Old Telephone Service (POTS). The near site telephone  110  has at least a microphone  102  and a loudspeaker  104 . Typically, the telephone also has a circuitry or processor module  106  to perform some signal processing. For example, most touch-tone phones can make different tones to represent different number keys, making artificial ring tones that can be changed by a user. The telephone  150  at the far site may or may not have the same components at in the telephone  110 . For simplicity, it is assumed that the telephone  150  has at least a microphone  152 , a loudspeaker  154  and a processing module  156 . 
     In a more advanced telephone, the processor module  106  may have more circuitry or more processing power to perform many functions. One state of the art telephone is a Polycom SoundStation® VTX-1000 speakerphone, available from the assignee of the current invention. The VTX-1000 has many more features and functions. For example, it is a speakerphone that allows full-duplex mode of operation. In full-duplex mode, talkers at both sites of the conference call can speak at the same time. To allow full-duplex mode of operation, the VTX-1000 has an advanced acoustic echo canceller (AEC). Without an AEC, annoying echo-like sounds will circulate between the two sites. If AEC is not implemented, then the speech signal  172  from a talker at the far site is transmitted through the network  130  to the near site telephone  110  as signal  134 . The speech signal  134  is reproduced by the loudspeaker  104 . Since the telephone is operating in full-duplex mode, the microphone  102  is active when loudspeaker  104  is working. The microphone  102  generates a signal  132 , which contains contributions due to the far end speech signal  172  from the loudspeaker  104 . This far end signal embedded in signal  132  is transmitted back to the far end together with the near site speech signal also in signal  132 . The entire signal  132  becomes a loudspeaker signal  174  at the far end and reproduced by loudspeaker  154 . This way, the far end talker will hear his voice back from the loudspeaker  154 , like an echo. This echo speech signal produced by the loudspeaker  154  can again be picked up by microphone  152 , transmitted through network  130 , reproduced by loudspeaker  104 , picked up by microphone  102  and transmitted back to loudspeaker  154 . If nothing is done to it, the echo signal can circulate between the two sites for a long time until dissipated into background noise, which is increased due to such echoes. Without AEC, full-duplex mode operation in a speakerphone is not practical due to the echoes and the noise. 
     When a process module  106  performs echo cancellation, it estimates the contribution of echo in the microphone signal  132  and subtracts that portion from the microphone signal  132 . This way, signal  132  only contains signals due to the speech of near site talkers. Therefore, what a far end talker can hear is the speech of near site talkers alone, without echo of his own voice. At the far end, another process module  156  may perform the similar acoustic echo cancellation. To achieve optimal goal of solving the echo problem, besides acoustic echo cancellation, echo suppression and noise fill may also be used. That is to minimize the residual echo heard by participants at the far site. 
     The process modules  106  and  156  may also perform other audio signal processing. For example, such processing may include parametric equalization. A particular microphone element may not respond to sound with uniform gain for all frequencies. To compensate for this non-uniformity, the process module may apply different filters on different frequencies to enhance or attenuate the frequency to achieve the uniform gain across the spectrum. The process module may also adjust the gain to change the characteristic of the speech or to achieve other acoustic objectives. 
     The process modules may also include automatic gain control (AGC) to accommodate the different loudness of speech from different talkers. There are various factors that may affect the gain of a microphone to speech, such as the loudness of the talker, the distance between the talker and the microphone or the orientation of the microphone and the talker. The use of AGC can avoid the wide fluctuation of the speech reproduced by a loudspeaker. 
     Another application of microphone signals is a public addressing system or a sound reinforcement system, as illustrated in  FIG. 2 . Such a system is typically used in theatres, auditoriums or large classrooms. One of the main differences of system  200  and system  100  is that system  200  is typically used at one site. The microphone  202  and loudspeaker  204  are located at the same general location such that sound from the loudspeaker  204  is picked up by the microphone  202 . The microphone  202 , process module  206  and loudspeaker  204  can form a closed loop. Unlike system  100 , system  200  does not have two sites and cannot have the echo problem. There is no need for acoustic echo cancellation. But it has its own problem, a feedback problem. If the closed loop has an overall gain above unity for a particular frequency, then for that frequency, system  200  has a positive feedback loop which reinforces itself until it makes a very loud squeaky noise, typically referred to as howling. The howling is very disruptive to meetings, lectures or artistic performances. It may also be destructive to acoustic equipment involved in the loop. Eliminating or avoiding feedback is a major concern in making and operating an audio reinforcement system  200 . In doing so, a slight degradation of the acoustic performance is acceptable. A typical method for eliminating feedback is to reduce the overall gain below unity for all frequencies. This may limit the amount of amplification in the reinforcement system, which is the main purpose of using such a system in the first place. More advanced methods to avoid feedback can dynamically detect and attenuate only the frequency that is likely to cause the howling, while keeping the gain for other frequencies intact, i.e., the gain for other frequencies possibly can be above unity. The selective attenuation of some frequencies can affect the sound quality, due to the missing portion of the spectrum and the artificial distortion. 
     As illustrated in  FIG. 2 , process module  206  may also perform many microphone signal processes  212 , including parametric equalization (PEQ), noise cancellation (NC), feedback elimination (FBE), dynamic process compression (DP), automatic gain control (AGC), and automatic mixing (AM). After performing desired processes on the microphone signal, the signal may be amplified by an amplifier  214  to form a loudspeaker signal  234 . Loudspeaker signal  234  is reproduced by a loudspeaker  204 . 
       FIG. 3  illustrates another system  300 , typically used in sound recording studios, radio broadcasting stations or court recorders. System  300  has a microphone  302 , a process module  306  and a recorder or other equipment  304 . The main difference between system  300  and systems  100  and  200  discussed earlier is that there is no closed loop in system  300 . The microphone  302  generates a signal  332 , processed by process module  306 , sent to recorder  304  (or other equipment for signal disposal) and that is the end of the system. There is no feedback from the processed signal to microphone  302 . Therefore, there is no need to perform some of the processes discussed in systems  100  and  200 , namely the echo cancellation, echo suppression and feedback elimination. Without the limitations imposed by the AEC and FBE processes, system  300  is typically focused on achieving the best sound quality possible, which is a requirement in a typical sound recording studio for recording a music performance or for a radio broadcasting stations for transmitting a live performance. When such a system is used for a court recorder, reliability is paramount, i.e., all words spoken or sounds must be recorded. In a typical system  300 , the microphone signal processes  312  may include PEQ, NC, DP and AGC etc. 
     SUMMARY OF THE INVENTION 
     As discussed above, different applications of microphone signals may require different processes. Some of the processes are similar, for example, most of the systems use AGC and PEQ. Some processes are different, for example AEC, FBE etc. Some processes necessary for one application may be in conflict with the purpose of another application. For example, feedback elimination is necessary for sound reinforcement application, but can degrade the acoustic quality. Feedback elimination should not be used in a sound recording application. 
     For clarity, systems  100 ,  200  and  300  are described separately and apply to different applications. But in actual applications, these systems may be used together in a single setting. For example, in a distance learning application as illustrated in  FIG. 5 , there is a local site and a far site. A professor is speaking at the local site. Students at both the local site and the far site can ask questions or otherwise interact with each other and the professor. The lecture is also recorded for use by students who do not have access to either the local classroom or a teleconference unit. In this case, the teleconference between the local site and the far site prefers the use of a conference system, similar to system  100  as shown in  FIG. 1 . But the interaction between the professor and the students at the local site prefers a sound reinforcement system as shown in  FIG. 2  such that speech of the professor and questioning student can be heard by all people. The recording for non-participating students prefers a recording system  300  as shown in  FIG. 3 . The currently available audio systems cannot satisfy all desires for the three applications. Most of the time, only one of the desires is satisfied and the other two desires are ignored. Sometimes, none of the desired goals is achieved. 
     Currently, even if a microphone system or audio system is installed for one particular application, the system still has to be modified or adjusted extensively for that particular application. It is time consuming, costly and confusing. To custom-manufacture or configure a microphone system or audio system useful for only one particular application is possible, but it increases the cost and is not desirable. 
     It is more to desirable have a system or method that can adapt to a particular application easily. It is very desirable to have a system that can accommodate all application goals at the same time and avoid the apparent conflicts between them. 
     The current invention uses a process module that can route a microphone signal to different processing paths. Each path is customized to achieve the goal for a particular application. The identical processes within different paths may be performed by the same process module to avoid duplication and save processing power. When installing the system, a process path is selected for a particular application. No complicated configuration is required. All potentially conflicting processes are accommodated within the same processor. 
    
    
     
       BRIEF DESCRIPTION OF THE DRAWINGS 
       A better understanding of the invention can be obtained when the following detailed description of the preferred embodiment is considered in conjunction with the following drawings, in which: 
         FIG. 1  illustrates a prior teleconference system. 
         FIG. 2  illustrates a prior art sound reinforcement system. 
         FIG. 3  illustrates a prior art sound recording system. 
         FIG. 4  illustrates a microphone processing system according to an embodiment of the current invention. 
         FIG. 5  illustrates a situation where all three applications are used. 
         FIG. 6  illustrates a signal routing in one embodiment with multiple microphones. 
         FIG. 7  illustrates another signal routing in an embodiment that makes use of an existing prior art audio system. 
     
    
    
     DESCRIPTION OF THE PREFERRED EMBODIMENT 
     The current invention includes devices and methods to multiplex microphone signals, where each signal is used for a particular application. Each signal path is independent from another signal, so conflicting signal processes may be applied for the different signals. Some processes are used in several signal paths, then such processes may be shared among the signal paths. 
       FIG. 4  illustrates one embodiment of the current invention. A microphone  402  generates microphone signal  404 . The signal is processed by parametric equalizer (PEQ)  412 , acoustic echo cancellation (AEC)  414  and noise cancellation (NC)  416 . These processes are common in all applications. Accordingly, they are shared among all signal processing paths. The resulting signal is  406 . Then the signal processing path splits into several paths. In this example, four paths are shown: an ungated path, a gated path, a sound reinforcement path and a user defined path, as denoted by the output signals  433 ,  453 ,  473  and  493 . The ungated path includes auto gain control (AGC)  424 , dynamic process compression (DP)  426  and fader mute (FM)  431 . The gated path includes echo suppression and noise fill (SNF)  442 , AGC  444 , DP  446 , automatic microphone mixing (AM)  448  and FM  451 . Similarly, the sound reinforcement path includes feedback elimination (FBE)  462 , AGC  464 , DP  466 , AM  468  and FM  471 . The customized path may have some of the above mentioned processes or user customized processes  482 ,  484 ,  486 ,  488  and  491 . This path allows a user of the system to mix and match pre-defined processes. It also allows the user to create his unique processes. It is noted that AGC  424 ,  444  and  464 , DP  426 ,  446  and  466 , AM  448  and  468 , and FM  431 ,  451  and  471  are similar process in each path, so the processor is the same among the different paths and is shared among them. This way, computational power is shared by the different paths. 
     The ungated signal  433  is configured to be used in an open-loop system, such as a sound recording system. The signal  433  is processed to achieve the highest quality and reliability. Any sound picked up by the microphone  402  is presented at signal  433  with high fidelity. Typically, only one or a few microphone signals are mixed for each output  433 . Signal  433  may be recorded by a high quality sound recorder or broadcasted to others. 
     A second path generates a gated signal  453 . The gated signal  453  is configured to be used in a closed-loop system, more particularly, a conferencing system. The echo suppression and noise fill process (SNF)  442  complements an AEC  414  to reduce echo heard by people at a far site. A noise fill is typically necessary to avoid dead silence at the far site, when people at the near site are not talking. Because of the echo suppression and noise fill process, the gain of the local microphone can vary dynamically depending on whether there are any people talking. In a conference setting, local speech is not reproduced in local loudspeaker, so it does not matter whether the gain varies. If a gated signal  453  is reproduced in a local loudspeaker, such as in a local sound reinforcement system, then the SNF  442 -caused variation can be noticeable and sometimes annoying. 
     A third signal path generates a sound reinforcement signal  473 . The sound reinforcement signal  473  is configured for use in a sound reinforcement system. SNF  442  is not used. The main reason for this is the doubletalk problem. In an audio conference, there are times when only people at one conference site are talking, i.e., single-talk, and there are times when people at more than one site are talking, i.e., doubletalk. SNF  442  works differently depending on whether there is single-talk or doubletalk in the conference. It is not a problem in a conference application, as discussed above related to the second signal path. But when the amplitude of local speech is reproduced by local loudspeakers, the fluctuation in the gain of the local speech can be noticeable and problematic. It is as if someone is mischievously turning the amplifier volume dial down or up as soon as you start speaking or stop speaking. By removing SNF  442 , the associated doubletalk problem is eliminated. The gain of the speech remains stable. Instead, FBE  462  is used. FBE reduces the feedback problem by attenuating a frequency that the FBE predicts to be likely to cause howling. Because of this attenuation, the sound spectrum is artificially altered. The resulting sound quality is lower. The particular frequency which is attenuated may vary with time, so the overall degradation of the sound quality may be minor. Even so, at any particular time and at a particular frequency, the distortion can be substantial. If that particular frequency at that time is significant for some reason, then the signal  473  could be unacceptable. That is why signal  473  is not suitable for use in a court reporting application, where reliability is paramount. 
     In both the gated and sound reinforcement paths, automatic microphone mixing (AM)  448  and  468  are used. In a case of multiple microphones generating a single signal, an AM shuts off the microphone where no speech is detected and only opens the microphone where speech is detected. This way, noise signals from microphones that do not have speech signals are not mixed into the final speech signal. The SNR of the resulting mixed speech signal is improved. In a single signal processing situation, AM is essentially an on/off switch. When there is no speech signal detected at the microphone, the AM turns the signal off, such that the noise from this microphone is not supplied to downstream signal processing. When there is speech signal, then the signal is turned on and supplied to downstream processes. This improves signal quality for both versions. It improves gain before feedback in the sound reinforcement version. AM is not used in the ungated version to avoid possible attenuation of the local speech. And by definition, the ungated version is typically used for an application where there is minimum background noise (i.e. recording studio) or where all “noises” are, “signals” (i.e. court reporting). 
       FIG. 4  only illustrates the audio signal processing part of an audio system that is relevant to the current invention. Audio sinks for the output signals, i.e., the destinations of the various output signals, are not shown. The output signals may be transmitted to the various audio sinks through the interfaces  435 ,  455 ,  475  and  495 . For each of the sinks, any of the several versions of the microphone signal may be selected. Although three of the output signals are processed and configured for three particular uses, they can be used for any purposes. Thus the audio sinks for the output signals can be many things that can accept audio signals, e.g., a loudspeaker, a conference unit at a far end site, a tape recorder, a radio transmitter, or other broadcast transmitter, etc. 
     Referring back to the setting illustrated in  FIG. 5 , the audio system  510  at the near site can employ the embodiment in  FIG. 4 . Using the embodiment of the current invention, the goal for each application can be achieved. The microphone signal  532  generated by microphone  502  is processed by a process module  506  as shown in  FIG. 4 , in three different paths for different applications. An ungated signal  538  is the output signal from the ungated path. It is recorded by recorder  582  for future use. In a court setting, the recorder  582  could be a court recorder. 
     The gated signal  536  is the output signal from the gated path. It is transmitted through a network  530  to the far site. This signal is substantially echo free. 
     The local sound reinforcement signal  534  is the output signal from the sound reinforcement path. It is combined with the loudspeaker signal  537  from the far site at a mixer  541  to form a local loudspeaker signal  539 . Local loudspeaker signal  539  is reproduced by loudspeaker  504 . So at the near site, both the local speech  532  and the far site speech  537  are amplified and can be heard by people at the near site of the conference. 
     The audio system  550  at the far site can be similar to the audio system  510  at the near site as discussed above, but it is not necessary. For example as shown in  FIG. 5 , system  550  may be a prior art conference unit. System  550  has a microphone  562 , loudspeaker  564  and a process module  566 . Since the audio system is only need to function as a conference unit, a prior art unit is sufficient. It is neither used for sound recording, nor for sound reinforcement. But if an audio system according to the current invention is available at the far site, then people at the far site would have the flexibility to add the two other functions that are available at the near site. If the far site has a system similar to the near site, then it can be used as a sound reinforcement system to accommodate many listeners at the far site. Also, it may record the lecture using its own recording device, instead of waiting for the near site to send the recording. 
     Most of the data processes can be implemented in a single data processor, such as a DSP.  FIG. 6  illustrates one embodiment that utilizes the capacity of a DSP to minimize the size and number of discrete components in an audio system. In this example, three input signals  612 ,  614  and  616  are shown, with four possible output signals  632 ,  634 ,  636  and  638 . The input signals may come from various sources, such as microphones  602 ,  604  or a telephone network interface  606 . The input signals are converted to digital signals from analog signals when necessary, for example by A/D converters  622 ,  624  or  626 . Each signal can be processed by a DSP  620 , which may perform many different processes, such as those discussed in reference to  FIG. 4 . Unlike many existing systems, each signal may be processed by the DSP  620  into different versions, such as discussed in reference to  FIG. 4 , i.e., ungated, gated or sound reinforcement versions. These different versions may be output as independent signals. For each of the audio sinks, any of the several versions of each source may be selected. For example, output signal  632  may be the gated version of signal  612 ; output signal  634  may be the sound reinforcement version of signal  612 ; output signal  636  may be an ungated version of signal  614 ; and output signal  638  may be a gated version of signal  616 . Similarly, the output signals may be a combination of processed input signals. In another example, output signal  632  is a mixture of gated version of signal  612  and  614 . Signal  634  is a mixture of ungated version of signal  616  and the sound reinforcement versions of signal  602  and  604 . There are many other possible combinations. The system is very flexible to adapt to a particular need. One benefit of such a system is that most of the signal processing, such as signal routing and mixing, is performed in the digital domain within the DSP. No rewiring of electrical cables is necessary. The output signals can be sent via appropriate interfaces for desired applications. 
     In prior art systems that include an adequate DSP, the current invention can be practiced by changing the process module in an existing audio system or reprogramming the processor in such a system. Such an upgrade can expand the capabilities of audio systems at very small incremental cost. 
     The current invention may also be practiced using a prior art system with limited capabilities, such as a Peavey Media Matrix and a Polycom Vortex conference unit. One such application is shown in  FIG. 7 . An audio system  720  has multiple inputs and multiple outputs. Each input may be independently processed and be sent out of the system. The system  720  includes some of the desired processes as discussed in  FIG. 4 . Others functions may be in other systems such as  729 . When various systems are combined, then an equivalent system similar to that shown in  FIG. 4  can be formed, where conflicting versions of a single signal may be created. In  FIG. 7 , microphone  702  generates a signal  712 . Signal  712  is digitized when necessary by A/D converter  722 . Signal  712  is processed by processor  723  in system  720 , which performs parametric equalization and noise cancellation processes. The output signal  732  is sent out of interface  742  as signal  770  and fed back to the inputs of system  720 . Signal  770  is split into three paths to make three versions, similar to those shown in  FIG. 4 . One path  774  is processed by processor  725  of system  720 , which generates an ungated signal  734 . The second signal  777  is processed by processor  727 , which generates a gated signal  737 . The third signal  778  is fed to another processor  729 , outside of system  720 . System  720  does not have a feedback elimination processor. So another system that has such capability is used. Process  729  generates a sound reinforcement signal  738 . This way, using two systems and some wiring back and forth, three conflicting versions of the same input signal  712  are generated. This embodiment of the current invention is more cumbersome. It may reduce the number of signals that can be processed because it may use several processors to process one signal. But it does have the advantage of using existing equipment. 
     According to the embodiments of the current invention, a microphone signal can go through several different processing paths. Each path is configured for a particular application. Different paths share the common processes to reduce computation loads. The individual processes may also be combined differently by a user to make a customized signal processing for a highly specialized application. The above discussion has focused on three common audio system applications that are distinct. Sometimes they have conflicting objectives or priorities. There are many other applications and processes not mentioned here. The current invention, where a signal can go through different processing paths and sharing common processes, is still applicable to them. 
     While illustrative embodiments of the invention have been illustrated and described, it will be appreciated that various changes can be made therein without departing from the spirit and scope of the invention.