Patent Description:
Patent Literature <NUM> and Patent Literature <NUM> disclose a configuration to enhance a target sound by the spectrum subtraction method. The configuration of Patent Literature <NUM> and Patent Literature <NUM> extracts a correlated component of two microphone signals as a target sound. In addition, each configuration of Patent Literature <NUM> and Patent Literature <NUM> is a technique of performing noise estimation in filter processing by an adaptive algorithm and performing processing of enhancing the target sound by the spectral subtraction method. Patent Literature <NUM> discloses a signal processing device comprising two directional microphones for collecting sound, and a signal processing device configured to perform echo reduction processing on the sound signals collected by the microphones and to calculate a coherent/correlated component between the microphone signals based on the signals resulting from the echo reduction processing. Patent Literature <NUM> discloses a signal processing device for measuring a distance from a microphone to a sound source by providing two microphones and detecting a level ratio of the sound signals output from the respective microphones, wherein one of the microphone is directional and the other omnidirectional.

In a case of a device that calculates sound of a sound source, using a microphone, the sound outputted from a speaker may be diffracted as an echo component. Since the echo component is inputted as the same component to two microphone signals, the correlation is very high. Therefore, the echo component becomes a target sound and the echo component may be enhanced.

In view of the foregoing, an object of a preferred embodiment of the present invention is to provide a signal processing device, a teleconferencing device, and a signal processing method that are able to calculate a correlated component, with higher accuracy than conventionally.

The above problem is solved by the subject matter of the independent claims.

According to the present invention, a correlated component is able to be calculated with higher accuracy than conventionally.

<FIG> is an external schematic view showing a configuration of a signal processing device <NUM>. In <FIG>, the main configuration according to sound collection and sound emission is described and other configurations are not described. The signal processing device <NUM> includes a housing <NUM> with a cylindrical shape, a microphone 10A, a microphone 10B, and a speaker <NUM>. The signal processing device <NUM> according to a preferred embodiment of the present invention, as an example, is used as a teleconferencing device by collecting sound, outputting a collected sound signal according to the sound that has been collected, to another device, and receiving an emitted sound signal from another device and outputting the signal from a speaker.

The microphone 10A and the microphone 10B are disposed at an outer peripheral position of the housing <NUM> on an upper surface of the housing <NUM>. The speaker <NUM> is disposed on the upper surface of the housing <NUM> so that sound may be emitted toward the upper surface of the housing <NUM>. However, the shape of the housing <NUM>, the placement of the microphones, and the placement of the speaker are merely examples and are not limited to these examples.

<FIG> is a plan view showing directivity of the microphone 10A and the microphone 10B. As shown in <FIG>, the microphone 10A is a directional microphone having the highest sensitivity in front (the left direction in the figure) of the device and having no sensitivity in back (the right direction in the figure) of the device. The microphone 10B is a non-directional microphone having uniform sensitivity in all directions. However, the directivity of the microphone 10A and the microphone 10B shown in <FIG> is an example. For example, both the microphone 10A and the microphone 10B may be non-directional microphones.

<FIG> is a block diagram showing a configuration of the signal processing device <NUM>. The signal processing device <NUM> includes the microphone 10A, the microphone 10B, the speaker <NUM>, a signal processing portion <NUM>, a memory <NUM>, and an interface (I/F) <NUM>.

The signal processing portion <NUM> includes a CPU or a DSP. The signal processing portion <NUM> performs signal processing by reading out a program <NUM> stored in the memory <NUM> being a storage medium and executing the program. For example, the signal processing portion <NUM> controls the level of a collected sound signal Xu of the microphone 10A or a collected sound signal Xo of the microphone 10B, and outputs the signal to the I/F <NUM>. It is to be noted that, in the present preferred embodiment, the description of an A/D converter and a D/A converter is omitted, and all various types of signals are digital signals unless otherwise described.

The I/F <NUM> transmits a signal inputted from the signal processing portion <NUM>, to other devices. In addition, the I/F <NUM> receives an emitted sound signal from other devices and inputs the signal to the signal processing portion <NUM>. The signal processing portion <NUM> performs processing such as level adjustment of the emitted sound signal inputted from other devices, and causes sound to be outputted from the speaker <NUM>.

<FIG> is a block diagram showing a functional configuration of the signal processing portion <NUM>. The signal processing portion <NUM> executes the program to achieve the configuration shown in <FIG>. The signal processing portion <NUM> includes an echo reduction portion <NUM>, a noise estimation portion <NUM>, a sound enhancement portion <NUM>, a noise suppression portion <NUM>, a distance estimation portion <NUM>, and a gain adjustment device <NUM>. <FIG> is a flow chart showing an operation of the signal processing portion <NUM>.

The echo reduction portion <NUM> receives a collected sound signal Xo of the microphone 10B, and reduces an echo component from an inputted collected sound signal Xo (S11). It is to be noted that the echo reduction portion <NUM> may reduce an echo component from the collected sound signal Xu of the microphone 10A or may reduce an echo component from both the collected sound signal Xu of the microphone 10A and the collected sound signal Xo of the microphone 10B.

The echo reduction portion <NUM> receives a signal (an emitted sound signal) to be outputted to the speaker <NUM>. The echo reduction portion <NUM> performs echo reduction processing with an adaptive filter. In other words, the echo reduction portion <NUM> estimates a feedback component to be calculated when an emitted sound signal is outputted from the speaker <NUM> and reaches the microphone 10B through a sound space. The echo reduction portion <NUM> estimates a feedback component by processing an emitted sound signal with an FIR filter that simulates an impulse response in the sound space. The echo reduction portion <NUM> reduces an estimated feedback component from the collected sound signal Xo. The echo reduction portion <NUM> updates a filter coefficient of the FIR filter using an adaptive algorithm such as LMS or RLS.

The noise estimation portion <NUM> receives the collected sound signal Xu of the microphone 10A and an output signal of the echo reduction portion <NUM>. The noise estimation portion <NUM> estimates a noise component, based on the collected sound signal Xu of the microphone 10A and the output signal of the echo reduction portion <NUM>.

<FIG> is a block diagram showing a functional configuration of the noise estimation portion <NUM>. The noise estimation portion <NUM> includes a filter calculation portion <NUM>, a gain adjustment device <NUM>, and an adder <NUM>. The filter calculation portion <NUM> calculates a gain W(f, k) for each frequency in the gain adjustment device <NUM> (S12).

It is to be noted that the noise estimation portion <NUM> applies the Fourier transform to each of the collected sound signal Xo and the collected sound signal Xu, and converts the signals into a signal Xo(f, k) and a signal Xu(f, k) of a frequency axis. The "f" represents a frequency and the "k" represents a frame number.

The gain adjustment device <NUM> extracts a target sound by multiplying the collected sound signal Xu(f, k) by the gain W(f, k) for each frequency. The gain of the gain adjustment device <NUM> is subjected to update processing by the adaptive algorithm by the filter calculation portion <NUM>. However, the target sound to be extracted by processing of the gain adjustment device <NUM> and the filter calculation portion <NUM> is only a correlated component of direct sound from a sound source to the microphone 10A and the microphone 10B, and the impulse response corresponding to a component of indirect sound is ignored. Therefore, the filter calculation portion <NUM>, in the update processing by the adaptive algorithm such as NLMS or RLS, performs update processing with only several frames being taken into consideration.

Then, the noise estimation portion <NUM>, in the adder <NUM>, as shown in the following equations, reduces the component of the direct sound, from the collected sound signal Xo(f, k), by subtracting the output signal W(f, k)·Xu(f, k) of the gain adjustment device <NUM> from the collected sound signal Xo(f, k) (S13).

Accordingly, the noise estimation portion <NUM> is able to estimate a noise component E(f, k) calculated by reducing the correlated component of the direct sound from the collected sound signal Xo(f, k).

Subsequently, the signal processing portion <NUM>, in the noise suppression portion <NUM>, performs noise suppression processing by the spectral subtraction method, using the noise component E(f, k) estimated by the noise estimation portion <NUM> (S14).

<FIG> is a block diagram showing a functional configuration of the noise suppression portion <NUM>. The noise suppression portion <NUM> includes a filter calculation portion <NUM> and a gain adjustment device <NUM>. The noise suppression portion <NUM>, in order to perform noise suppression processing by the spectral subtraction method, as shown in the following equation <NUM>, calculates spectral gain |Gn(f, k)|, using the noise component E(f, k) estimated by the noise estimation portion <NUM>.

Herein, β(f, k) is a coefficient to be multiplied by a noise component, and has a different value for each time and frequency. The β(f, k) is properly set according to the use environment of the signal processing device <NUM>. For example, the β value is able to be set to be increased for the frequency of which the level of a noise component is increased.

In addition, in this present preferred embodiment, a signal to be subtracted by the spectral subtraction method is an output signal X'o(f, k) of the sound enhancement portion <NUM>. The sound enhancement portion <NUM>, before the noise suppression processing by the noise suppression portion <NUM>, as shown in the following equation <NUM>, calculates an average of the signal Xo(f, k) of which the echo has been reduced and the output signal W(f, k)·Xu(f, k) of the gain adjustment device <NUM> (S141).

The output signal W(f, k)·Xu(f, k) of the gain adjustment device <NUM> is a component correlated with the Xo(f, k) and is equivalent to a target sound. Therefore, the sound enhancement portion <NUM>, by calculating the average of the signal Xo(f, k) of which the echo has been reduced and the output signal W(f, k)·Xu(f, k) of the gain adjustment device <NUM>, enhances sound that is a target sound.

The gain adjustment device <NUM> calculates an output signal Yn(f, k) by multiplying the spectral gain |Gn(f, k)| calculated by the filter calculation portion <NUM> by the output signal X'o(f, k) of the sound enhancement portion <NUM>.

It is to be noted that the filter calculation portion <NUM> may further calculate spectral gain G'n(f, k) that causes a harmonic component to be enhanced, as shown in the following equation <NUM>. <MAT> <MAT>.

Here, i is an integer. According to the equation <NUM>, the integral multiple component (that is, a harmonic component) of each frequency component is enhanced. However, when the value of f/i is a decimal, interpolation processing is performed as shown in the following equation <NUM>.

Subtraction processing of a noise component by the spectral subtraction method subtracts a larger number of high frequency components, so that sound quality may be degraded. However, in the present preferred embodiment, since the harmonic component is enhanced by the spectral gain G'n(f, k), degradation of sound quality is able to be prevented.

As shown in <FIG>, the gain adjustment device <NUM> receives the output signal Yn(f, k) of which the noise component has been suppressed by sound enhancement, and performs a gain adjustment. The distance estimation portion <NUM> determines a gain Gf(k) of the gain adjustment device <NUM>.

<FIG> is a block diagram showing a functional configuration of the distance estimation portion <NUM>. The distance estimation portion <NUM> includes a gain calculation portion <NUM>. The gain calculation portion <NUM> receives an output signal E(f, k) of the noise estimation portion <NUM>, and an output signal X' (f, k) of the sound enhancement portion <NUM>, and estimates the distance between a microphone and a sound source (S15).

The gain calculation portion <NUM> performs noise suppression processing by the spectral subtraction method, as shown in the following equation <NUM>. However, the multiplication coefficient γ of a noise component is a fixed value and is a value different from a coefficient β(f, k) in the noise suppression portion <NUM>. <MAT> <MAT> <MAT>.

The gain calculation portion <NUM> further calculates an average value Gth(k) of the level of all the frequency components of the signal that has been subjected to the noise suppression processing. Mbin is the upper limit of the frequency. The average value Gth(k) is equivalent to a ratio between a target sound and noise. The ratio between a target sound and noise is reduced as the distance between a microphone and a sound source is increased and is increased as the distance between a microphone and a sound source is reduced. In other words, the average value Gth(k) corresponds to the distance between a microphone and a sound source. Accordingly, the gain calculation portion <NUM> functions as a distance estimation portion that estimates the distance of a sound source based on the ratio between a target sound (the signal that has been subjected to the sound enhancement processing) and a noise component.

The gain calculation portion <NUM> changes the gain Gf(k) of the gain adjustment device <NUM> according to the value of the average value Gth(k) (S16). For example, as shown in the equation <NUM>, in a case in which the average value Gth(k) exceeds a threshold value, the gain Gf(k) is set to the specified value a, and, in a case in which the average value Gth(k) is not larger than the threshold value, the gain Gf(k) is set to the specified value b (b < a). Accordingly, the signal processing device <NUM> does not collect sound from a sound source far from the device, and is able to enhance sound from a sound source close to the device as a target sound.

It is to be noted that, while, in the present preferred embodiment, the sound of the collected sound signal Xo of the non-directional microphone 10B is enhanced, subjected to gain adjustment, and outputted to the I/F <NUM>, the sound of the collected sound signal Xu of the directional microphone 10A may be enhanced, subjected to gain adjustment, and outputted to the I/F <NUM>. However, the microphone 10B is a non-directional microphone and is able to collect sound of the whole surroundings. Therefore, it is preferable to adjust the gain of the collected sound signal Xo of the microphone 10B and to output the adjusted sound signal to the I/F <NUM>.

The technical idea described in the present preferred embodiment will be summarized as follows.

As with Patent Literature <NUM> (<CIT>) and Patent Literature <NUM> (International publication No.<CIT>), in a case in which echo is generated when a correlated component is calculated using two signals, the echo component is calculated as a correlated component, which causes the echo component to be enhanced as a target sound. However, the signal processing device according to the present preferred embodiment, since calculating a correlated component using a signal of which the echo has been reduced, is able to calculate a correlated component, with higher accuracy than conventionally.

The signal processing portion <NUM> calculates an output signal W(f, k)·Xu(f, k) being a correlated component by performing filter processing by an adaptive algorithm, using a current input signal or the current input signal and several previous input signals.

For example, Patent Literature <NUM> (<CIT>) and Patent Literature <NUM> (International publication No. <CIT>) employ the adaptive algorithm in order to estimate a noise component. In an adaptive filter using the adaptive algorithm, a calculation load becomes excessive as the number of taps is increased. In addition, since a reverberation component of sound is included in processing using the adaptive filter, it is difficult to estimate a noise component with high accuracy.

On the other hand, while, in the present preferred embodiment, the output signal W(f, k)·Xu(f, k) of the gain adjustment device <NUM>, as a correlated component of direct sound, is calculated by the filter calculation portion <NUM> in the update processing by the adaptive algorithm, as described above, the update processing is update processing in which an impulse response that is equivalent to a component of indirect sound is ignored and only one frame (a current input value) is taken into consideration. Therefore, the signal processing portion <NUM> of the present preferred embodiment is able to remarkably reduce the calculation load in the processing to estimate a noise component E(f, k). In addition, the update processing of the adaptive algorithm is the processing in which an indirect sound component is ignored and the reverberation component of sound has no effect, so that a correlated component is able to be estimated with high accuracy. However, the update processing is not limited only to one frame (the current input value). The filter calculation portion <NUM> may perform update processing including several past signals.

The signal processing portion <NUM> (the sound enhancement portion <NUM>) performs sound enhancement processing using a correlated component. The correlated component is the output signal W(f, k)·Xu(f, k) of the gain adjustment device <NUM> in the noise estimation portion <NUM>. The sound enhancement portion <NUM>, by calculating an average of the signal Xo(f, k) of which the echo has been reduced and the output signal W(f, k)·Xu(f, k) of the gain adjustment device <NUM>, enhances sound that is a target sound.

In such a case, since the sound enhancement processing is performed using the correlated component calculated by the noise estimation portion <NUM>, sound is able to be enhanced with high accuracy.

The signal processing portion <NUM> (the noise suppression portion <NUM>) uses a correlated component and performs processing of reducing the correlated component.

More specifically, the noise suppression portion <NUM> performs processing of reducing a noise component using the spectral subtraction method. The noise suppression portion <NUM> uses the signal of which the correlated component has been reduced by the noise estimation portion <NUM>, as a noise component.

The noise suppression portion <NUM>, since using a highly accurate noise component E(f, k) calculated in the noise estimation portion <NUM>, as a noise component in the spectral subtraction method, is able to suppress a noise component, with higher accuracy than conventionally.

The noise suppression portion <NUM> further performs processing of enhancing a harmonic component in the spectral subtraction method. Accordingly, since the harmonic component is enhanced, the degradation of the sound quality is able to be prevented.

The noise suppression portion <NUM> sets a different gain β(f, k) for each frequency or for each time in the spectral subtraction method. Accordingly, a coefficient to be multiplied by a noise component is set to a suitable value according to environment.

The signal processing portion <NUM> includes a distance estimation portion <NUM> that estimates a distance of a sound source. The signal processing portion <NUM>, in the gain adjustment device <NUM>, adjusts a gain of the collected sound signal of the first microphone or the collected sound signal of the second microphone, according to the distance that the distance estimation portion <NUM> has estimated. Accordingly, the signal processing device <NUM> does not collect sound from a sound source far from the device, and is able to enhance sound from a sound source close to the device as a target sound.

The distance estimation portion <NUM> estimates the distance of the sound source, based on a ratio of a signal X'(f, k) on which sound enhancement processing has been performed using the correlated component and a noise component E(f, k) extracted by the processing of reducing the correlated component. Accordingly, the distance estimation portion <NUM> is able to estimate a distance with high accuracy.

Claim 1:
A signal processing device (<NUM>) comprising:
a first microphone (10A), which is a directional microphone, configured to collect a first sound signal Xu;
a second microphone (10B), which is a non-directional microphone, configured to collect a second sound signal Xo; and
a signal processing portion (<NUM>) configured to perform echo reduction processing on one of the first sound signal Xu and the second sound signal Xo, and to calculate a correlated component between the first sound signal Xu and the second sound signal Xo, converting the first sound signal Xu and a signal of which the echo has been reduced from the second sound signal Xo by the echo reduction processing, or a signal of which the echo has been reduced from the first sound signal Xu by the echo reduction processing and the second sound signal Xo into signals of a frequency axis.