JP2004172703A - Method and apparatus of signal processing - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To obtain a clearer sound image location when forming a sound field with a speaker array. <P>SOLUTION: Series circuits consisting of digital delay circuits DL0-DLn and digital filters DF0-DFn are provided on signal lines between a source SC of a digital audio signal and speakers SP0-SPn forming the speaker array 10. A delay time for forming a focal point by the speaker array 10 is divided into an integer part and a fraction part. The delay time in the integer part is set in the circuits DL0-DLn. The pulse of a frequency for oversampling the digital audio signal is subjected to downsampling for the sampling frequency of the digital audio signal to obtain waveform data of a pseudo-pulse sequence. Of the waveform data of the pseudo-pulse sequence, the waveform data close to the fraction part is set as the filter coefficients of the digital filters DF0-DFn. <P>COPYRIGHT: (C)2004,JPO

Description

[0001]
TECHNICAL FIELD OF THE INVENTION
The present invention relates to an audio signal processing method and apparatus suitable for use in a home theater or the like.
[0002]
[Prior art]
As a speaker system suitable for application to a home theater or an AV system, there is a speaker array system (for example, see Patent Document 1). FIG. 9 shows an example of the speaker array system. In the system shown in FIG. 9, the speaker array 10 is configured by arranging a large number of speakers (speaker units) SP0 to SPn. In this case, as an example, n = 255 and the speaker diameter is several centimeters. Therefore, the speakers SP0 to SPn are actually arranged two-dimensionally on a plane. For simplicity, it is assumed that the speakers SP0 to SPn are arranged in a row in the horizontal direction.
[0003]
Then, the audio signal is supplied from the source SC to the delay circuits DL0 to DLn and is delayed by a predetermined time τ0 to τn, and the delayed audio signal is supplied to the speakers SP0 to SPn through the power amplifiers PA0 to PAn, respectively. . The delay times τ0 to τn of the delay circuits DL0 to DLn will be described later.
[0004]
Then, in any place, the sound waves output from the speakers SP0 to SPn are synthesized, and the sound pressure resulting from the synthesis is obtained. Therefore, as shown in FIG. 9, in the sound field formed by the speakers SP0 to SPn, to make the sound pressure at an arbitrary location Ptg higher than the surroundings,
L0 to Ln: distance from each speaker SP0 to SPn to location Ptg s: assuming sound speed, delay times τ0 to τn of delay circuits DL0 to DLn
τ0 = (Ln−L0) / s
τ1 = (Ln−L1) / s
τ2 = (Ln−L2) / s
...
τn = (Ln−Ln) / s = 0
Set to.
[0005]
With this setting, when the audio signal output from the source SC is converted into a sound wave by the speakers SP0 to SPn and output, the sound waves are output with a delay of time τ0 to τn shown in the above equation. become. Therefore, when those sound waves reach the location Ptg, they all arrive in the same phase, and the sound pressure at the location Ptg becomes higher than the surroundings. That is, the sound output from the speakers SP0 to SPn focuses on the place Ptg such that the parallel light focuses on the convex lens. Hereinafter, the location Ptg will be referred to as the focus, and this type of system will be referred to as the "focus type".
[0006]
FIG. 10 shows another speaker array system. In this system, delay times τ0 to τn of the delay circuits DL0 to DLn are set so that traveling waves (sound waves) output from the speakers SP0 to SPn have the same phase wavefront, and directivity is given to the sound waves. At the same time, the pointing direction is the direction of the focal point Ptg.
[0007]
This system can be considered to be a case where the distances L0 to Ln are set to infinity in a focus type system. Hereinafter, this type of system will be referred to as a "directivity type".
[0008]
As described above, according to the speaker array system 10, by appropriately setting the delay times τ0 to τn, the focal point Ptg can be formed at an arbitrary position in the sound field, and the directional direction can be adjusted. .
[0009]
In both systems, in places other than the place Ptg, the outputs of the speakers SP0 to SPn are combined in a state where the phases are shifted, so that the outputs are averaged and the sound pressure is reduced. Further, the sound output from the speaker array 10 may be once reflected on the wall surface and then focused on the location Ptg, or the directivity may be set to the direction of the location Ptg.
[0010]
[Patent Document 1]
JP-A-5-303381
[Problems to be solved by the invention]
By the way, in order to delay the audio signal output from the source SC without deteriorating in the delay circuits DL0 to DLn, the delay circuits DL0 to DLn need to be configured by digital circuits. Can be configured. Also, in actual AV equipment, the source SC is often a digital equipment such as a DVD player, and the audio signal is a digital signal. Therefore, the delay circuits DL0 to DLn are more preferably constituted by digital circuits. Become.
[0012]
However, when the delay circuits DL0 to DLn are configured by digital circuits, the time resolution of the audio signal supplied to the speakers SP0 to SPn is limited by the digital audio signal and the sampling interval (sampling cycle) in the delay circuits DL0 to DLn. It cannot be longer than the sampling interval. Incidentally, when the sampling frequency is 48 kHz, the sampling period is about 20.8 μs, and the sound wave advances by about 7 mm during one period. The delay of one cycle corresponds to a phase delay of 70 ° in an audio signal having a frequency of 10 kHz.
[0013]
For this reason, the phase of each sound wave output from the speakers SP0 to SPn cannot be sufficiently adjusted at the focal point Ptg, and the size of the focal point Ptg, that is, the sound image viewed from the listener may be large or blurred. .
[0014]
In addition, the variation in the phase of the sound wave at locations other than the focal point Ptg is reduced, and it is impossible to expect a sufficient decrease in sound pressure at locations other than the focal point Ptg. Therefore, also from this point, the sound image becomes large or blurred, and the original effect cannot be exhibited.
[0015]
Furthermore, a reverberation component may be added to the audio signal by the delay circuits DL0 to DLn. In such a case, if the delay processing of the cycle shorter than the sampling cycle in the delay circuits DL0 to DLn cannot be performed, the target reverberation pattern is not obtained. It may not be possible to reproduce.
[0016]
The present invention is intended to solve the above problems.
[0017]
[Means for Solving the Problems]
In the present invention, for example,
In a signal processing method for delaying a digital signal by a predetermined delay time,
The predetermined delay time is divided into an integer part and a decimal part using the sampling period of the digital signal as a unit,
Oversampling the impulse response including the delay time represented by at least the fractional part of the predetermined delay time with a period smaller than the sampling period,
A downsampling process is performed on the sample sequence obtained by this oversampling to obtain pulse waveform data of the sampling period,
This pulse waveform data is set as the filter coefficient of the digital filter,
A method for processing a digital signal, wherein the digital signal is supplied to the digital filter operating at the sampling period.
Therefore, a required fraction of the delay time is realized by the digital filter, and an appropriate delay time is given to the digital signal.
[0018]
BEST MODE FOR CARRYING OUT THE INVENTION
{Circle around (1)} Outline of the present invention In the present invention, an impulse response representing a delay is oversampled at a frequency higher than the sampling frequency of the system and expressed with a resolution higher than the sampling interval of the system, and the data of the impulse is represented by the system. , A pulse train composed of a plurality of pulses is obtained, and the pulse train is stored in a database. When delay times τ0 to τn are given to a digital audio signal, data stored in the database is set in a digital filter.
[0019]
Hereinafter, the above pulse train is referred to as a “pseudo pulse train”. Further, the symbols are defined as follows.
Fs: sampling frequency of the system.
Nov: a value indicating what fraction of the sampling period 1 / Fs the time resolution should be. It is also a multiple of oversampling with respect to the sampling frequency Fs.
Nps: The number of pulses when the shape of a pulse on the time axis of the oversampling period 1 / (Fs × Nov) is approximately represented by a plurality of pulses having a sampling frequency of Fs. It is also the number of pulses in the pseudo pulse train and the order of the digital filter that achieves the desired delay.
As an example,
Fs = 48 kHz, Nov = 8, Nps = 16
It is.
[0020]
(2) -1 Creation of database As a pre-process of reproduction by the speaker array 10, first, a pseudo pulse train is generated as described above and registered in the database. That is,
(1) Based on the required time resolution, a multiple Nov of the oversampling and the number of pulses Nps of the pseudo pulse train are assumed. In this case, as shown in FIGS. 1A and 1B, the time resolution from the M-th pulse to the next (M + 1) -th pulse is increased by Nov times. Further, a time width by Nps pulses is set on the time axis of the sampling period 1 / Fs.
[0021]
(2) Since the multiple of the oversampling is the value Nov, as shown in FIG. 1B, during the period from the Mth pulse to the (M + 1) th pulse, Nov oversampling pulses rise. Become.
And
m = 0, 1, 2,..., Nov-1
Then, on the time axis of the sampling period 1 / Fs, the position of the oversampling pulse is (M + m / Nov). Alternatively, on the time axis of the oversampling period 1 / (Fs × Nov), the position of the oversampling pulse is (M + Nov × m).
[0022]
(3) As shown in FIG. 1C, a pseudo pulse train is obtained by down-sampling the oversampling pulse of the term (2) from the sampling frequency Fs × Nov to the sampling frequency Fs.
In this case, for example, a method is conceivable in which each series of the term (2) is frequency-axis transformed using FFT, and inverse FFT is performed on the time axis while leaving only effective values up to the sampling frequency Fs. Since there are many down-sampling methods including the design of an aliasing filter, they will not be described here.
[0023]
(4) Thereafter, the pseudo pulse train (sequence of the number of pulses Nps) obtained by the term (3) is assumed to be a pulse standing at the time position (M + m / Nov) on the time axis of the sampling period 1 / Fs. deal with. In this case, on the time axis of the sampling period 1 / Fs, the value M is an integer, and the value m / Nov is a decimal.
[0024]
(5) As shown in FIG. 1D, the value M is regarded as offset information, the value m / Nov is regarded as index information, and the correspondence between the information and the waveform data of the quasi-pulse train obtained in the section (4). The table is registered in the database 20.
[0025]
2 to 5 show waveforms, gain characteristics, and phase characteristics of the pseudo pulse train formed by the items (1) to (4). 2 to 5 show the case where Nov = 8 and Nps = 16 as described above, and show m = 0 to 7.
[0026]
For example, when m = 0 shown in FIG. 2, the time-base waveform has a value of 1.0 at the eighth sample and 0.0 at the other sample values, so that it is simple by eight sample periods (8 / Fs). Shows the transfer characteristic for delaying. Hereinafter, it is shown that the peak position in the time axis waveform gradually moves to the ninth sample as the value m increases. At this time, although the respective frequency gain characteristics are almost flat, it can be seen that the phase delay of the frequency phase characteristics increases as the value m increases. That is, the delay processing with the time resolution of 1 / (Fs × Nov) is realized by the filtering processing of the sampling frequency Fs.
[0027]
The above is the pre-processing necessary for reproduction. Thereafter, the reproduction processing described below is executed using the information in the database 20.
[0028]
(2) -2 Processing at the time of reproduction At the time of reproduction by the speaker array 10, reproduction is performed using the database 20 created according to the above (2) -1 as follows. That is,
(11) A digital filter is provided in series with the delay circuits DL0 to DLn. This digital filter is used for delay, and its filter coefficient is set as described later.
[0029]
(12) The delay times τ0 to τn corresponding to the position of the focal point Ptg (or the directional direction) are obtained, multiplied by the sampling frequency Fs, and the delay times τ0 to τn are set as “delay samples” on the frequency axis of the sampling frequency Fs. To "number". At this time, the delay times τ0 to τn may be values having fractions that cannot be represented by the resolution of the delay circuits DL0 to DLn. That is, the delay times τ0 to τn and the number of delay samples need not be integral multiples of the resolution of the delay circuits DL0 to DLn.
[0030]
(13) The number of delay samples obtained in (12) is divided into an integer part and a fractional part (fractional part), and the integer part is set as the delay time of the delay circuits DL0 to DLn.
[0031]
(14) It is determined which of the index information m / Nov stored in the database 20 is closer to the fractional part of the number of delay samples obtained in (12). That is, it is determined which of 0 / Nov, 1 / Nov, 2 / Nov, ..., (Nov-1) / Nov the decimal part is closer to. When it is determined that the decimal part is close to Nov / Nov = 1.0, the integer part is moved up by one, and the decimal part is determined to be close to 0 / Nov.
[0032]
(15) According to the determination result of the item (14), the waveform data of the corresponding pseudo pulse train is extracted from the database 20, and set as the filter coefficient in the FIR digital filter of the item (11).
[0033]
As described above, the total delay time of the delay circuits DL0 to DLn and the digital filter for the audio signal is the delay time τ0 to τn obtained in the section (12). Therefore, in the case of the focus type system, the sound output from the speakers SP0 to SPn is focused on the position of the focus Ptg, and the sound image is clearly localized. In the case of a directional system, the directional direction matches the location Ptg, and the sound image is also clearly localized.
[0034]
Also, since the sound from the speakers SP0 to SPn is more accurately aligned in phase at the focal point Ptg, at this time, the phase is more varied at a place other than the focal point Ptg, and as a result, other than the focal point Ptg, The sound pressure at the place can be further reduced. Therefore, the localization of the sound image becomes clear also from this point.
[0035]
Strictly speaking, the time resolution does not increase in all the bands, and depending on the downsampling method, it may be difficult to obtain the time resolution for the high frequency band. Considering the sound pressure difference from a place other than the focal point Ptg (or a non-directional direction), there is actually an effect that the directivity is sufficiently enhanced in almost all frequency bands.
[0036]
(3) -1 First Embodiment FIG. 6 shows an example of a reproducing apparatus according to the present invention. That is, a digital audio signal is extracted from the source SC, and the audio signal is sequentially supplied to the digital delay circuits DL0 to DLn and the FIR digital filters DF0 to DFn, and the filter output is supplied to the power amplifiers PA0 to PAn.
[0037]
In this case, the delay time of the delay circuits DL0 to DLn is an integer part shown in the item (13). Further, the FIR digital filters DF0 to DFn set the filter coefficients in accordance with the term (15) to delay the decimal part shown in the term (13). Further, in the power amplifiers PA0 to PAn, the digital audio signal supplied thereto is D / A converted and then power-amplified or D-class amplified and supplied to the speakers SP0 to SPn.
[0038]
Further, a database 20 is prepared. The database 20 has a correspondence table between the offset information M and the index information m / Nov according to the items (1) to (5) and the waveform data of the pseudo pulse train obtained in the item (4). Then, the database 20 is searched according to the decimal part of the item (13), and the search result is set in the FIR digital filters DF0 to DFn. Further, the integer part of the term (13) is set to the delay time of the delay circuits DL0 to DLn.
[0039]
According to such a configuration, even if the delay time τ0 to τn required to focus on the location Ptg (or to set the location Ptg in the directivity direction) exceeds the resolution of the delay circuits DL0 to DLn, the FIR digital The delay time of the filters DF0 to DFn realizes a fractional part exceeding the resolution.
[0040]
Therefore, in the case of the focus type system, the sound output from the speakers SP0 to SPn is focused on the position of the focus Ptg, and the sound image is clearly localized. In the case of a directional system, the directional direction matches the location Ptg, and the sound image is also clearly localized.
[0041]
(3) -2 Second Embodiment In the reproducing apparatus shown in FIG. 7, FIR digital filters DF0 to DFn also serve as delay circuits DL0 to DLn. That is, in this case, the database 20 is searched according to the index information m / Nov, and based on the search result, the offset information M is set in the FIR digital filters DF0 to DFn, and the delay time of the delay circuits DL0 to DLn is added. At the same time, the waveform data of the index information m / Nov is set.
[0042]
Therefore, also in this reproducing apparatus, since the focal point Ptg or the directional direction is appropriately set, a clear localization of a sound image can be obtained.
[0043]
(3) -3 Third Embodiment The reproducing apparatus shown in FIG. 8 is a reproducing apparatus shown in FIG. 7 in which digital filters DF0 to DFn also realize sound effects such as equalizing, amplitude (volume), and reverberation. is there. For this reason, in the convolution circuits CV0 to CVn, the external data to be the target acoustic effect is convolved with the data extracted from the database 20, and the output is set to the FIR digital filters DF0 to DFn.
[0044]
{Circle around (4)} The delay processing according to the present invention is not limited to the application to the speaker array 10 described above. For example, if the present invention is applied to a channel divider used in a multi-way speaker system, it is possible to perform so-called time alignment in which the positions of virtual sound sources between a low-range speaker and a high-range speaker are finely adjusted. In an apparatus for reproducing high-quality audio by means of SACD, DVD-Audio, or the like, it is desired that the arrangement position of the super tweeter in the front-rear direction can be adjusted in millimeters.
[0045]
Further, in the above description, the data in the database 20 may be calculated in advance and prepared in a memory such as a ROM, or may be calculated in real time as needed.
[0046]
In addition, in order to reduce the calculation speed when calculating data in the database 20, the resources required for the calculation, or the amount of data in the memory, the data in the database 20 may or may not be used depending on the focal point Ptg and the location of the pointing direction. Can be used properly. For example, when the focal point Ptg is located in the lateral direction of the listener, there is no problem even if the accuracy is lower than when the focal point Ptg is located in the front direction. By automatically controlling so as to reduce the number Nps, the total data amount and calculation amount can be suppressed.
[0047]
Furthermore, the number of values Nov and Nps can be automatically changed according to the position and the directional direction of the focal point Ptg, or the calculation amount and the calculation capability of the hardware in each case. In addition, for example, when the position of the focal point Ptg, the directional direction, and the like are dynamically changed in real time to enhance the effect, the processing can be continuously performed. In this case as well, the values Nov and Nps can be dynamically changed.
[0048]
[List of abbreviations used in this specification]
AV: Audio and Visual
CCD: Charge Coupled Device
D / A: Digital to Analog
FFT: Fast Fourier Transform
FIR: Finite Impulse Response
SACD: Super Audio CD
[0049]
【The invention's effect】
According to the present invention, since the fraction of the delay time that cannot be realized by the delay circuit is realized by the digital filter, the sound output from the speaker is given a necessary delay, and therefore, the position of the focal point and the directional direction are provided. , The sound image is localized. In addition, since the sound pressure is reduced in places other than the focus and the directional direction, the localization of the sound image becomes clear from this point. Further, when an acoustic effect such as reverberation is added, appropriate characteristics can be obtained.
[Brief description of the drawings]
FIG. 1 is a diagram for explaining the present invention.
FIG. 2 is a characteristic diagram for explaining the present invention.
FIG. 3 is a characteristic diagram for explaining the present invention.
FIG. 4 is a characteristic diagram for explaining the present invention.
FIG. 5 is a characteristic diagram for explaining the present invention.
FIG. 6 is a system diagram illustrating one embodiment of the present invention.
FIG. 7 is a system diagram showing another embodiment of the present invention.
FIG. 8 is a system diagram showing another embodiment of the present invention.
FIG. 9 is a plan view for explaining the present invention.
FIG. 10 is a plan view for explaining the present invention.
[Explanation of symbols]
Reference Signs List 10 speaker array, 20 database, DL0 to DLn delay circuit, DF0 to DFn FIR digital filter, PA0 to PAn power amplifier, Ptg focus, SC source, SP0 to SPn speaker (speaker unit)

Claims (13)

A signal processing method for delaying a digital signal by a predetermined delay time divided into an integer part and a decimal part in units of a sampling period of the digital signal,
An impulse response including a delay time represented by at least the decimal part of the predetermined delay time is oversampled at a cycle smaller than the sampling cycle, and a downsampling process is performed on a sample sequence obtained by the oversampling. To obtain the pulse waveform data of the above sampling period,
This pulse waveform data is set as the filter coefficient of the digital filter,
A signal processing method in which the digital signal is supplied to the digital filter operating at the sampling period. The signal processing method according to claim 1,
Of the predetermined delay time, the digital delay circuit operating at the sampling period performs an integer multiple of the sampling period, and performs the remaining delay processing including the delay time represented by the decimal part by the digital filter. A signal processing method to be performed. The signal processing method according to claim 1,
The oversampling period of the oversampling process is 1 / N of the sampling period of the digital signal (N ≧ 2 is an integer), and the delay time represented by the decimal part is an integer (m) of the oversampling period. A signal processing method in which m / N is applied as the decimal part when the number is close to double. The signal processing method according to claim 3,
The pulse waveform data to be delayed by a delay time which is m / N (m = 1 to N-1) of the sampling period is stored in a database in advance,
A signal processing method in which pulse waveform data close to the decimal part is extracted from the stored pulse waveform data and set as a filter coefficient of the digital filter. The signal processing method according to claim 1,
A signal processing method in which a transfer characteristic for giving a predetermined sound effect is convolved with the pulse waveform data and set as a filter coefficient of the digital filter. A signal processing method for delaying at least one digital audio signal among digital audio signals supplied to each of a plurality of speakers by a predetermined delay time,
The predetermined delay time is divided into an integer part and a decimal part using the sampling period of the digital audio signal as a unit,
An impulse response including a delay time represented by at least the decimal part of the predetermined delay time is oversampled at a cycle smaller than the sampling cycle, and a downsampling process is performed on a sample sequence obtained by the oversampling. To obtain the pulse waveform data of the above sampling period,
This pulse waveform data is set as the filter coefficient of the digital filter,
A signal processing method, wherein the digital audio signal is supplied to the digital filter operating at the sampling period. A signal processing device for delaying a digital signal by a predetermined delay time divided into an integer part and a decimal part in units of a sampling period of the digital signal,
An impulse response including a delay time represented by at least the decimal part of the predetermined delay time is oversampled at a cycle smaller than the sampling cycle, and a downsampling process is performed on a sample sequence obtained by the oversampling. An arithmetic circuit for calculating the pulse waveform data of the sampling period,
A digital filter that sets pulse waveform data obtained by the arithmetic circuit as a filter coefficient,
A signal processing device configured to supply the digital signal to the digital filter that operates at the sampling period. A signal processing device for delaying a digital signal by a predetermined delay time divided into an integer part and a decimal part in units of a sampling period of the digital signal,
An impulse response including a delay time represented by at least the decimal part of the predetermined delay time is oversampled at a cycle smaller than the sampling cycle, and a downsampling process is performed on a sample sequence obtained by the oversampling. Storage means for storing the pulse waveform data of the sampling period obtained by
A digital filter from which pulse waveform data stored in the storage means is extracted and set as a filter coefficient,
A signal processing device configured to supply the digital signal to the digital filter that operates at the sampling period. In the signal processing device according to claim 7 or 8,
The oversampling period of the oversampling process is 1 / N of the sampling period of the digital signal (N ≧ 2 is an integer), and the delay time represented by the decimal part is an integer m times the oversampling period. A signal processing device adapted to apply m / N as the decimal part when it is close. The signal processing device according to claim 8,
The pulse waveform data corresponding to the plurality of decimal parts is stored in the storage means in advance,
A signal processing device wherein pulse waveform data close to the decimal part is extracted from the stored pulse waveform data and set as a filter coefficient of the digital filter. In the signal processing device according to claim 7 or 8,
The pulse waveform data further includes a processing circuit that generates a composite waveform data by convolving a transfer characteristic that gives a predetermined sound effect,
A signal processing device configured to set the synthesized waveform data generated by the processing circuit to a filter coefficient of the digital filter. A signal processing device for delaying at least one digital audio signal among digital audio signals supplied to each of a plurality of speakers by a predetermined delay time,
The predetermined delay time is divided into an integer part and a decimal part in units of a sampling period of the digital audio signal, and the impulse response including at least the delay time represented by the decimal part of the predetermined delay time is sampled. An arithmetic circuit that performs oversampling at a cycle smaller than the cycle, performs downsampling processing on a sample sequence obtained by the oversampling, and calculates pulse waveform data at the sampling cycle;
A digital filter that sets the pulse waveform data obtained by the arithmetic circuit as a filter coefficient,
A signal processing device configured to supply the digital audio signal to the digital filter that operates at the sampling period. A signal processing device for delaying at least one digital audio signal among digital audio signals supplied to each of a plurality of speakers by a predetermined delay time,
The predetermined delay time is divided into an integer part and a decimal part in units of a sampling period of the digital audio signal, and the impulse response including at least the delay time represented by the decimal part of the predetermined delay time is sampled. Storage means for oversampling at a cycle smaller than the cycle and storing pulse waveform data of the sampling cycle obtained by performing downsampling processing on a sample sequence obtained by the oversampling;
A digital filter that takes out the pulse waveform data stored in the storage means and sets it as a filter coefficient,
A signal processing device configured to supply the digital signal to the digital filter that operates at the sampling period.

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