KR20070020457A - Conference system - Google Patents

Abstract

The conference system 1 includes a central unit 2 and a speaker unit 3 connectable to the central unit. The central unit 2, which serves to combine the speech signals from the speaker unit 3 and distribute the combined speech signals to the units, comprises an adaptive filter 23 for suppressing feedback. Each speaker unit 3 comprises a microphone 33, a speaker 34, an activation switch 35 and an adaptive filter 36 coupled between the microphone 33 and the speaker 34. When the speaker unit is not activated, the adaptive filter 36 acts as an echo canceller, while it acts as a feedback suppressor when the speaker unit is activated. By keeping the speaker 34 always on, any transients due to incorrect adaptation of the filter 36 are avoided.

Description

Conference system {CONFERENCE SYSTEM}

The present invention relates to a conference system. More particularly, the invention relates to a conference system comprising a central unit and at least one speaker unit that can be coupled to the central unit. The speaker unit each includes a speaker and a microphone to allow the representative to attend the meeting. The central unit combines the microphone signals from all speaker units and distributes the combined microphone signals to all speaker units, which typically do not necessarily occur after amplification of this combined signal. The speaker, or equivalent transducer, of the speaker unit renders this combined signal.

Although conferencing systems are traditionally used for meetings and meetings, the same technology is now used in automobiles, airplanes, and other means of transport, where several people want to talk in the presence of background noise.

The conferencing system uses multiple microphones in multiple discrete locations (ie, in front of each representative) to create separate signals, and conferencing in that only one or two of them are selectively rendered. It is noted that the system is different from the public speaking system. As the public speaking system also uses multiple speakers, there is no selective rendering of the microphone signal in the public speaking system.

U. S. Patent No. 5,404, 397 deals with a conference system comprising a speaker unit coupled to a central unit. This known conference system is provided with automatic speaker detection. To this end, the central unit compares the speech signal of the speaker unit and activates the unit (s) with the highest signal level. To avoid any false speaker detection due to sound made by other speakers, each speaker unit includes an echo canceller provided with an adaptive filter. When the speaker unit is activated, the speaker of such unit is switched off and the echo canceller is bypassed.

Although very efficient to suppress unwanted acoustic feedback from the speaker to the microphone of the speaker unit, it has been found that switching off the speaker introduces signal distortion. Each time the speaker switches on and off, the sound pattern detected by the microphone and processed by the echo canceller changes, i.e. the acoustic path between the speaker and the microphone alternately sums up or is eliminated. This implies that each time the speaker unit is (de) activated, the echo canceller, in particular its adaptive filter, must adapt to changes in the acoustic path. This results in a transient signal, a transient signal that is not compensated for by the echo canceller and thus distorts the (echo compensated) microphone signal. Transients occur especially when the speakers of a known speaker unit are reactivated. Transients also occur in neighboring speaker units where the microphones of these speaker units directly record the sound produced by the reactivated speakers.

It has also been found that a significant portion of the sound path recorded by the microphone of the active speaker unit comes from the speaker (s) of the neighboring speaker unit. This reduces the maximum allowable gain of the conference system when this acoustic feedback introduces howling.

It is an object of the present invention to provide a conference system which overcomes these and other problems of the prior art and avoids transients due to activation and deactivation of the speaker unit.

It is another object of the present invention to provide a speaker unit and a central unit for use in such a conference system.

Accordingly, the present invention provides a conference system comprising at least one speaker unit and a central unit, the at least one speaker unit comprising: an input for receiving a speaker signal, an output for supplying a microphone signal, a speaker coupled to the input An adaptive filter coupled between the speaker and the coupling unit, a microphone coupled to the coupling unit and an activation device coupled between the coupling unit and the output, the central unit providing an input for receiving a microphone signal and a speaker signal. A speaker of at least one speaker unit is permanently coupled to its input, and the central unit is provided with an additional adaptive filter coupled between its input and its output.

By providing a speaker that is permanently coupled to the input of the speaker unit and thus permanently activated, transients caused by switching on or off the speaker are avoided. Since the speaker typically renders the combined microphone signal of all active speaker units, it almost continuously produces the sound recorded by the microphone. As a result, the adaptive filter of the associated speaker unit continues to adapt its filter parameters to the same acoustic path, resulting in stable filtering without transients.

By providing an additional adaptive filter in the center unit, any inverse effects of the speaker that remain active are compensated. The additional adaptive filter in the central unit serves as an acoustic feedback suppressor that removes any feedback from the output signal of the central unit to the input signal.

In one preferred embodiment, the central unit also includes a decorrelator. The correlator, which can be arranged substantially parallel to the adaptive filter, serves to remove any correlation between the input signal and the output signal of the adaptive filter. In the absence of a correlator, the adaptive filter may have a tendency to reduce the amplitude of the combined microphone signal and it is also possible to introduce signal distortion. The correlator is preferably composed of a frequency shifter. However, phase shifters and / or time-variable delays can also be used as correlation cancellers.

The central unit may also include a dynamic echo suppressor, which serves to suppress the remaining echo in the residual signal of the adaptive filter.

The time span of the adaptive filter may be defined as the product of the filter length (number of delay units) and the sampling frequency. Although various time ranges may be used, the adaptive filter of the speaker unit preferably has a time range between 20 ms and 45 ms, preferably between 30 ms and 35 ms. In particular, a time range of about 32 ms is suitable. Such a relatively short time range results in an adaptive filter that can converge quickly when the speaker unit is inactive, since the microphone signal only contains echoes from other speaker units.

Although various types of adaptive filters can be used, when the echo-to-reverberation ratio (ENR) is below a certain threshold, the adaptive filter has an adaptive speed that is substantially proportional to the estimate of the echo-to-reverberation ratio in the microphone signal. Preferably, the threshold value is equal to one. In one such embodiment, the filter reacts quickly when the microphone signal contains only reverberation, and slowly responds when the microphone signal contains significant non-reflective signal components, such as the desired voice.

In another preferred embodiment of the conference system according to the invention, the adaptive filter of the central unit has a time range between 125 ms and 500 ms, preferably between 200 ms and 300 ms. Particularly preferred is a time range of about 250 ms. In general, it is preferable that the time range of the (additional) adaptive filter of the central unit is larger than the time range of the adaptive filter of the speaker unit (s), preferably considerably larger. In this way, the adaptive filter of the speaker unit (s) is arranged to eliminate direct echoes, while the adaptation filter of the central unit is arranged to eliminate indirect or diffused echoes.

The conference system of the present invention may advantageously be mounted on a vehicle such as a car, bus or truck. The speaker unit may be portable and a clip may be provided for clipping the speaker's clothing. However, the speaker unit may be embedded in a seat, ceiling, wall, floor or other part of the vehicle.

The present invention also provides a speaker unit for use in the conference system described above, the speaker unit comprising: an input for receiving a speaker signal, an output for supplying a microphone signal, a speaker coupled to the input, between the speaker and the coupling unit An adaptive filter coupled to the microphone, a microphone coupled to the coupling unit, an activation device coupled between the coupling unit and the output, and the speaker is permanently coupled to the input.

The present invention further provides a central unit for use in the conference system described above, which central unit has an input for receiving a microphone signal, an output for supplying a speaker signal and further adaptation coupled between its input and output. Include a filter. The central unit of the invention may further be provided with a correlator, a dynamic echo suppressor and / or an amplifier.

The invention is now described with reference to exemplary embodiments illustrated in the accompanying drawings.

1 schematically illustrates an exemplary conference system comprising a speaker unit and a central unit.

2 is a schematic circuit diagram of a conference system according to the present invention;

3 is a schematic circuit diagram of a first embodiment of a central unit according to the invention;

4 is a schematic circuit diagram of a second embodiment of a central unit according to the invention;

5 is a schematic circuit diagram of one embodiment of a speaker unit according to the invention.

The conferencing system 1 shown simply by way of non-limiting example in FIG. 1 comprises a central unit 2 and a speaker unit 3. The central unit 2 combines the voice signals from the speaker unit 3 and distributes the combined voice signals to the unit. Each speaker unit 3 comprises a speaker 34 for rendering a voice signal coupled with a microphone 33 for producing a voice signal. Each speaker unit 3 is provided with an activation button (not shown) for activating the unit's microphone 33. In a typical conference system, the microphone 33 is usually turned off and only the speaker unit activated by the user (commonly referred to as the "delegate") produces a speech signal.

The conference system of the present invention can be used in a conference room or a conference hall, but can also be mounted in a vehicle, such as a car, bus, truck, airplane or ship. The speaker unit may be portable and may be provided with a clip for hanging the speaker's (passenger and / or driver / pilot) clothing. However, the speaker unit may be embedded in a seat, ceiling, wall, floor or other part of the vehicle.

In the circuit diagram of FIG. 2, a conference system 1 according to the invention is shown schematically. The conference system 1 of FIG. 2 also includes a central unit 2 and two speaker units 3. Although only two speaker units 3 are shown, it will be appreciated that more speaker units 3 can be connected to the central unit 2, for example four, eight, ten or 21 speaker units. .

The central unit 2 is an input 21 for receiving a microphone signal from the speaker unit 3 and a speaker signal (ie, a combined, filtered and / or amplified microphone signal) for supplying the speaker unit 3 to the speaker unit 3. An adaptation filter 23 for filtering the output 22 and the microphone signal and a coupling unit 29 for combining the microphone signal with the filter signal, ie the signal output by the filter 23.

The speaker unit 3 has an input 31 for receiving a speaker signal, an output 32 for outputting a microphone signal, a speaker 34 coupled to the input 31, and an input 31 for receiving a speaker signal, respectively. Coupled filter 36, microphone 33 for producing a microphone signal, coupling unit (signal adder) 39 and coupling unit 39 (and thus microphone) for combining the microphone signal and the filter output signal. (33)} to a switch 35 for selectively connecting to the output 32.

As is apparent from FIG. 2, the speaker units 3 are arranged side by side, and each unit 3 is connected to the input 21 and the output 22 of the central unit 2. It is understood that the output 32 of each speaker unit 3 can be connected to a separate input 21 of the central unit via a separate wire or other suitable connection. Therefore, the central unit 2 can all have a plurality of inputs 21 which can be connected to the coupling unit 29.

The adaptive filter 23 of the central unit 2 serves as an acoustic feedback suppressor. The adaptive filter 23 models a path of sound existing between the speaker 34 and the microphone 33 of any active speaker unit, and outputs a signal that comes close to the microphone signal made by these sound paths. In the coupling unit 29, this filter signal is subtracted from the microphone signal. As a result, the signal r represents a "pure" microphone signal, i.e. a signal produced by the speaker ("representative") rather than by the speaker.

As the acoustic path can change over time, such as due to the movement of people or items in a meeting room, the filter is adaptive, ie, repeatedly or continuously so that the filter coefficients of the filter best fit the acoustic path at a particular point in time. Is adapted. Adaptive filters are well known, but the prior art associated with conferencing systems failed to announce or suggest a central unit provided with adaptive filters.

In the conference system of the present invention, the speaker unit 3 is also provided with an adaptive filter. These adaptive filters 36 act as acoustic feedback suppressors (AFS) when each speaker unit is active, and act as acoustic echo cancellers (AEC) when each speaker unit is inactive. It will be appreciated that the speaker signal is derived from the microphone signal of the speaker unit, whereas in the case of AEC, the speaker signal is derived from an external signal.

As can be seen in FIG. 2, the adaptive filter 36 is disposed alongside the acoustic path extending from the speaker 34 to the microphone 33. When the speaker unit is not activated (when the switch 35 is opened), the microphone 33 records sound made by the speaker 34 of the speaker unit itself and the speaker of another speaker unit. The adaptive filter 36 of the speaker unit, when subtracted from the microphone signal at the coupling unit 39, is a filter signal that substantially cancels the sound from the speaker 34 of the same speaker unit (acoustic echo canceller (AEC)). To produce The filter signal also typically cancels sound from the speaker (s) of any neighboring speaker unit reaching the microphone directly or indirectly through nearby objects. As will be explained later with reference to FIG. 5, the adaptive filter 36 attempts to remove any correlation between the speaker signal and the microphone signal. Any correlation between these signals when the speaker unit is inactive will be due to the microphone 33 taking the sound of the speaker (s) 34. When the speaker unit is inactive, the adaptive filter 36 can react quickly and accurately model the acoustic path (s).

When the speaker unit 3 is activated (when the switch 35 is closed), the microphone signal is supplied to the input 21 of the central unit via the output 32 of the speaker unit 3, and adapts to the central unit at the central unit. It is filtered by the filter 23. According to the present invention, the speaker 34 is in an active state when the speaker unit is active. As a result, the sound produced by the speaker 34 now contains a microphone signal, which significantly increases the correlation of the speaker signal with the microphone signal. Now both the speaker unit adaptive filter 36 and the center unit adaptive filter act as acoustic feedback suppressors (AFS).

The adaptation speed of AFS is lower than that of AEC. In the case of AEC, the microphone signal contains only echo, whereas in AFS, the microphone signal contains both the echo and the desired audio signal. In the case of AFS, fast adaptation can result in the desired degradation of the voice.

In the conferencing system of the present invention, the combined action of the adaptive filters 23, 36 eliminates direct sound from the speaker, first reflections from nearby objects, and any diffused feedback from other objects. In addition, the speaker unit adaptive filter 36 completely cancels direct sound and any first reflections when the speaker unit is active and thus avoids introducing any transients.

One alternative embodiment of the central unit 2 is shown in FIG. 3. This embodiment also provides an input 21 for receiving the microphone signal z, an output 22 for supplying the speaker signal x, an adaptive filter 23 for producing a filter signal y and a residual signal. and a combining unit 29 for combining the signals z and y to produce (r). The central unit 2 of FIG. 3 also includes an update unit 24 and a correlation canceller 26. The updating unit 24 receives both the residual signal r output by the combining unit 29 and the uncorrelated signal x, and determines the correlation of these two signals. The coefficients of the filter 23 are then adapted in such a way that the correlation is minimized. In the absence of correlator 26, the adaptive filter attempts to suppress the residual signal z since signal z and signal z are substantially the same (another signal processing element is sent to central unit 2). May be present and these processing elements are not shown in FIG. 3 for clarity. The correlator 26 is preferably configured by a frequency shifter that shifts the frequency of the residual signal r by a few kilohertz. Instead of or in addition to frequency shifting, correlation canceller 26 may include phase shifting and / or time-varying delays. The correlator 26 not only prevents signal distortion but also increases the adaptation speed of the adaptive filter 23.

The embodiment of the central unit 2 shown in FIG. 4 includes, in addition to the components mentioned above with reference to FIG. 3, a dynamic echo suppressor (DES) 27 and an amplifier 28. The dynamic echo suppressor 27 receives the microphone signal z, the filter signal y and the residual signal r to produce a compensated residual signal r '. Such dynamic echo suppressor serves to temporarily reduce the amplitude of the residual signal when the change in the acoustic path causes the acoustic feedback compensation signal made by the adaptive filter to contain the phase error. Such a change in the acoustic path is usually introduced when the speaker unit is activated or deactivated. The dynamic echo suppressor therefore further reduces the undesirable effects of speaker unit (de) activation.

The dynamic echo suppressor 27 corrects the amplitude of the frequency component of the input signal z (with pure delay aside) without changing its phase. This determines the frequency spectrum (Fourier transform) of both the input signal, which is the filter signal y, and the residual signal r, to obtain the transformed signals Y, Z and R, and to obtain the combined transformed signal R '. Determine the magnitude of the transformed signal (Y, Z, and R) and the phase of R using the magnitudes of Y, Z, and R, and use the magnitude of the combined transformed signal (R ') By reconstructing the signal r ', it is achieved. Dynamic echo suppressors of this type are described in US patent application US 2003/0026437, the entire contents of which are incorporated herein by reference.

As mentioned above, the adaptive filter of the central unit compensates for the echo that is caused by the speakers of all speaker units and arrives at the microphone (s) of the active speaker unit (s) mainly through reflection from the wall. In the particularly advantageous embodiment of FIG. 4, the dynamic echo suppressor 27 eliminates any remaining echo.

The speaker unit 3 of FIG. 5 also includes an input 31, an output 32, a microphone 33, a speaker 34, a switch 35 and an adaptive filter 36. In addition, the speaker unit of FIG. 5 is shown to include an update unit 37 for updating the filter coefficients of the adaptive filter 36. The function of the update unit 37 is essentially the same as that of the update unit 24 of the central unit 2, and thus need not be described herein.

It is noted that in use the adaptive filter 37 of any active speaker unit 3 is arranged substantially parallel to both the adaptive filter 24 and the correlator 26 of the central unit 2. The advantage of incorporating the correlator 26 in the central unit 2 applies equally to the speaker unit 3.

It is desirable to have a relatively short time range to allow rapid adaptation of the speaker unit adaptation filter 36. The time range of the filter is defined as the product of the filter length (the number of delay units in the digital filter) and the sampling frequency. In one preferred embodiment, the filter has a time range between 20 and 45 ms, in particular between 30 and 35 ms. Although a time range of about 32 ms is particularly advantageous, it has been found that other time range values may be used. Such a relatively short time range allows the speaker unit adaptive filter 36 to only compensate for the reverberation produced by the speaker of the same unit and the speaker (s) of any adjacent speaker unit. These echoes arrive at the microphone either directly or indirectly through reflections from nearby objects.

It is more advantageous when the central unit adaptive filter 23 has a larger time range than the speaker unit adaptive filter 36, especially when it has a considerably larger time range. The adaptive filter 23 of the central unit 2 preferably has a time range between 125 ms and 500 ms, preferably between 200 ms and 300 ms. Particularly preferred is a time range of approximately 250 ms. In this way, the central unit adaptive filter 23 is arranged to compensate for the echo, ie the echo spread from the wall and other non-adjacent objects.

To allow for a smooth change from the AEC mode of the adaptive filter 36 when the speaker unit 3 is not active, and from the AFS mode when the speaker unit is active, the adaptation speed is mic if the ENR ratio does not exceed a certain threshold. It is desirable to make it proportional to the estimate of the echo to non-repercussion (ENR) ratio of the signal. The adaptation speed of the filter can be adjusted by changing its step-size, which is well known to those skilled in the art. The ENR can be estimated based on the residual signal output by the coupling unit 39 and the input signal of the adaptive filter 36, which is the same as the input signal of the update unit 37.

Therefore, the updating unit 37 is an ENR (Echo Versus Non-Echo) estimator for producing an ENR estimation signal, such as a comparator for comparing the ENR estimation signal with a (stored) threshold equal to 1 and such a signal does not exceed the threshold. Otherwise it may contain circuitry for adapting the adaptive speed of the adaptive filter to the ENR estimation signal. In one such embodiment, the adaptive filter responds relatively quickly when the microphone signal only contains echoes, and responds relatively slowly when the microphone signal contains the desired voice.

It is noted that the switch 35 can be configured as a remotely controlled electromechanical or electromechanical switch such as a hand operated switch, key or button or relay. The switch 35 can therefore be controlled directly or indirectly by a representative or central unit or central control unit associated with the speaker unit. In the latter case, the chairman can remotely operate the switch 35.

It is also noted that in the foregoing discussion, it is assumed that all signals are digital signals having a particular value at a particular discrete point in time. However, the present invention is not limited to such, and analog embodiments can also be considered. Similarly, although the present invention has been described with reference to a speaker unit having a single microphone and a single speaker, the present invention can be applied even when using a speaker unit having a plurality of microphones and / or speakers and / or an equivalent transducer.

The present invention is based on the fact that switching the speakers of the conference system speaker unit on or off can result in a transient that causes signal distortion. The present invention benefits from another fact that if an adaptive filter is provided in both the speaker unit and the center unit, it can be turned on permanently.

It is also noted that any terminology used herein should not be considered as limiting the scope of the invention. In particular, the word "comprising" does not mean excluding any element not specifically described. A single (circuit) element can be replaced by a number of (circuit) elements or their equivalents.

It will be appreciated by those skilled in the art that the present invention is not limited to the above-described embodiments, and that many modifications and additions can be made without departing from the scope of the invention as defined in the appended claims.

As mentioned above, the present invention is applicable to a conference system comprising a central unit and at least one speaker unit that can be coupled to the central unit.

Claims (11)

A conference system (1) comprising at least one speaker unit (3) and a central unit (2), The at least one speaker unit 3 includes an input 31 for receiving a speaker signal, an output 32 for supplying a microphone signal, a speaker 34 coupled to the input 31, and a speaker 34. And an adaptive filter 36 coupled between the coupling unit 39, a microphone 33 coupled to the coupling unit 39, and an activation device 35 coupled between the coupling unit 39 and the output 34. and, The central unit 2 comprises an input 21 for receiving a microphone signal and an output 22 for supplying a speaker signal, The speaker 34 of at least one speaker unit 3 is permanently coupled to its input 31, The central unit (2) is provided with an additional adaptive filter (23) coupled between its input (21) and its output (22). The conferencing system according to claim 1, wherein the central unit (2) further comprises a decorrelator (26). 3. The conferencing system of claim 2, wherein the correlator (26) consists of a frequency shifter. A conference system according to claim 1, wherein the central unit (2) further comprises a dynamic echo suppressor (27). The conference system according to claim 1, wherein the adaptive filter (36) of the speaker unit (3) has a time range between 20 ms and 45 ms, preferably between 30 ms and 35 ms. 2. The adaptive filter 36 according to claim 1, wherein the adaptive filter 36 is based on an estimated value of the echo to non echo ratio in the microphone signal when the echo to non-echo ratio (ENR) is less than a certain threshold. And are arranged to have a substantially proportional adaptation speed, wherein the threshold is preferably equal to one. The conference system according to claim 1, wherein the adaptive filter (23) of the central unit (2) has a time range between 125 ms and 500 ms, preferably between 200 ms and 300 ms. The conference system of claim 1 mounted to a vehicle. A speaker unit (2) for use in the conference system (1) according to claim 1, wherein the speaker unit has an input (31) for receiving a speaker signal, an output for supplying a microphone signal, and the input (31). Loudspeaker 34 coupled to speaker 34, adaptive filter 36 coupled between speaker 34 and coupling unit 39, microphone 33 coupled to coupling unit 39, coupling unit 39 and output ( 34 an activation device 35 coupled therebetween, The speaker unit (34) is permanently coupled to the input (31). A central unit 2 for use in the conference system 1 according to claim 1, wherein the central unit has an input 21 for receiving a microphone signal, an output 22 for supplying a speaker signal and its input And a further adaptive filter (23) coupled between the output and the output. 11. Central unit according to claim 10, further provided with a correlator (26), a dynamic echo suppressor (27) and / or an amplifier (28).

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