RU2237964C2 - Psycho-acoustic processor (adaptive equalizer) - Google Patents

Abstract

FIELD: sound signals processing systems engineering.

SUBSTANCE: device has controlled timbre block and control block, control output of which is connected to control input of controlled timbre block, input and output of which are respectively input and output of processor, while controlled timbre block is made for n bands, control block is made for n channels, each channel of which is used for controlling timbre block only at one frequency band, to n-band information input of control block n band outputs of timbre block are connected used for separate outputting of information in each frequency band, and n-band standard input of control block is also a standards processor input used for forming a shape of spectrum of output signal.

EFFECT: possible automatic frequency correction at whole spectrum of input signal in accordance to any chosen requirements.

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Description

The invention relates to devices for processing audio frequency signals and is used to convert the spectrum of the original signal in accordance with predetermined psychoacoustic requirements. The processor is intended for use in sound reinforcement systems, including reproduction and amplification of waveforms, as well as in sound recording systems.

A device is known that contains n band-pass filters, n variable-gain amplifiers, a constant-gain amplifier and an adder for (n + 1) inputs, the output of which is the output of the device, the input of which is the combined inputs of n-band-filters and a constant-gain amplifier , the outputs of n bandpass filters are connected to the inputs of the corresponding n amplifiers with variable gain, the outputs of which are connected to the corresponding n inputs of the adder, the (n + 1) th input of which is connected to stroke amplifier with constant gain [Kisel VA Analog and Digital Proofreaders: A Handbook. M .: Radio and communication, 1986, p. 131, fig. 4.14].

The device is an n-band equalizer, which allows you to adjust the amplitude-frequency characteristic of the signal path, thereby affecting the spectrum of the output signal. In this case, the equalizer does not take into account the spectral features of the input signals. As a result, signals with different spectra are subjected to the same frequency correction, which does not allow us to consider the equalizer as an automatic device that performs equalization of signals with different spectral properties, which is its significant drawback.

The closest in its technical essence and the achieved effect to the claimed device is a psychoacoustic processor (maximizer) containing a controlled tone block, a high-pass filter (HPF), a bandpass filter (PF) and a control unit, the output of which is connected to the control input of the tone block, the output of which is the output of the processor, the input of which is the combined inputs of the HPF, PF and timbral block, the outputs of the HPF and PF are connected to the information inputs of the control unit [Chernetskiy M. Psychoacoustic processors. - Sound producer, 2002, No. 4, p. 4].

The psychoacoustic prototype processor independently analyzes the spectrum of the input signal and, depending on the ratio of energies in the high-frequency and mid-frequency regions of the spectrum, enhances or weakens the high-frequency components. Thus, it is possible to automatically maintain in a certain ratio the balance between the mid-frequency and high-frequency parts of the output signal range, regardless of the spectral properties of the input signal.

The disadvantage of the prototype is limited functionality, consisting only in the automatic control of the high-frequency part of the spectrum of the output signal.

The technical result achieved using the present invention is to enable automatic frequency correction over the entire spectrum width of the input signal in accordance with arbitrarily specified requirements.

The technical result is achieved by the fact that in a known psychoacoustic processor containing a controlled tone block and a control unit, the control output of which is connected to a control input of a controlled tone block, the input and output of which are the input and output of the processor, according to the invention, connected to an n-band information input of the control unit n band outputs of the tone block, and the n-band reference input of the control unit is the reference input of the processor.

The invention is illustrated by functional diagrams.

Figure 1 shows a functional diagram of a psychoacoustic processor; figure 2 is a functional diagram of a control unit 2; figure 3 is a functional diagram of the shaper 9 teams.

Functional diagram of the psychoacoustic processor (figure 1) contains a controlled fuser 1 and a control unit 2. In turn, the fuser 1 contains a group of 3 bandpass filters, a group of 4 amplifiers with a variable transmission coefficient and an analog adder 5.

The psychoacoustic processor DI input is the combined inputs of N filters 3, the outputs of which are connected to the inputs of the corresponding N amplifiers 4, the outputs of which are connected to the N inputs of the analog adder 5, the output of which is the DO output of the processor, the control input of which is the control input of the control unit 2, which controls the output Δu (ω) of which, consisting of N channels, is connected respectively to the N control inputs of N amplifiers 4, N channel tone outputs are connected to the N-channel information input u (ω) of the control unit 2 Robot 1, which, in turn, are outputs of the corresponding N amplifiers 4, the N-channel reference input u ₀ (ω) of the control unit 2 performs the functions of the N-band reference input of the processor.

The functional diagram of the control unit 2 (Fig. 2) contains a group of 6 detectors, a group of 7 low-pass filters (low-pass filters), a group of 8 analog adders and a shaper of 9 commands, which, in turn, consists of a group of 10 analog keys. The outputs of N detectors 6 through the corresponding N low-pass filters 7 are connected to the first information inputs of the corresponding adders 8, the second information inputs of which comprise the N-channel reference input u ₀ (ω) of the control unit 2, the N-channel information input u (ω) of which comprise the corresponding inputs detector 6 outputs n adders 8 are connected to respective inputs of n Δu (ω) command generator 9 outputs Δu (ω _n) which is the control output Δu (ω) control unit 2, which serves as a control input SB generates a corresponding input STUDIO 9 formed by combining the enable inputs OE N keys 10 (Figure 3), N inputs which are the inputs of the shaper 9 and N outputs respectively output shaper 9.

Psychoacoustic processor (figure 1) operates as follows.

An input signal lying in the frequency range [ω _n ; ω _in ] (ω _n and ω _in , the lower and upper boundaries of the frequency range, respectively) and the frequency correction to be applied, is fed to the input DI of the n-band timbral unit 1, which is divided into n bands of width Δω. The width of the spectral window Δω is selected so that the entire range is overlapped, i.e., NΔω = ω _in -ω _n . The band signals allocated by the filters 3 are supplied to the corresponding N information inputs u (ω) of the control unit 2, in which they are compared with the reference values specified by the N-channel reference input u ₀ (ω) of the said block. The comparison result, which is N control signals Δu (ω), is supplied from the output of block 2 to the control inputs of the tone block, where it affects the transmission coefficients of the respective amplifiers from group 4. Thus, by changing the transmission coefficients of the amplifiers 4, the spectrum shape (envelope) is converted the output signal in accordance with the requirements specified by the reference input u ₀ (ω) of the control unit 2.

The signals taken from the outputs of the amplifiers 4 and carrying information about the distribution of energy in the analyzed frequency range [ω _n ; ω _in ], can be compared with the reference signals on various grounds. For example, you can compare the average power, RMS or RMS values, depending on which the structure of the control unit 2 is selected. Figure 2 presents the functional diagram of block 2 under the assumption that the average rectified values are compared. The average straightened values are formed in the chain detector 6 - low-pass filter 7. In the comparison schemes of group 8, performing the function of subtraction, the difference is calculated

Δu (ω _n ) = u (ω _n ) -u ₀ (ω _n ), (1)

where u (ω _n ) is the average rectified value at the output of the nth low-pass filter 7 (n = 1, 2, ..., N) and corresponds to a frequency band with a central value of ω _n ;

u ₀ (ω _n ) is the reference value, represented as the average rectified value and corresponding to the frequency band with the central value of ω _n .

Given that the spectrum is converted at the designated N reference points, the number of which (the number of bands) is selected depending on the required accuracy, the external driving action can be represented as an N-dimensional vector of reference values

U ₀ = {u ₀ (ω ₁ ), u ₀ (ω ₂ ), ..., u ₀ (ω _N )} (2)

Vector (2) determines the desired shape of the spectrum of the output signal by forming an N-dimensional vector of control actions ΔU, which is the difference of the vector of input signals

U = {u (ω ₁ ), u (ω ₂ ), ..., u (ω _N )} (3)

and vectors (2), i.e.

ΔU = UU ₀ = {Δu (ω ₁ ), Δu (ω ₂ ), ..., Δu (ω _N )} (4)

Thus, due to the presence of a feedback circuit (the output of block 2 is the control inputs of amplifiers 4), the average rectified voltage values at the outputs of amplifiers 4 tend to the corresponding reference values u ₀ (ω _n ), which should ultimately lead to a change and maintenance of the shape of the spectrum output signal according to the reference samples u ₀ (ω _n ).

The input signal in the general case is a random function of time, the spectrum of which, even with the simplifying assumption that it is stationary and ergodic, can be estimated fairly accurately only with a large observation interval. However, in real conditions, much less time is allocated for making decisions on frequency correction of time than is required for a high-precision estimate. This is explained by the fact that in our task, the decision should be made already at the beginning of the signal, the musical work, and not after its end, when an extensive amount of statistical data has been accumulated. This means that the duration of the averaging interval when calculating the average rectified values of u (ω _n ) will also be quite small. Consequently, the voltage u (ω _n ) will fluctuate randomly with a dispersion depending on the averaging interval, which is determined by the time constant of the low-pass filter 7. Since the vector U _o (2) is a constant, then the vector of control actions ΔU ( 4). For this reason, a very important task arises additionally about the method of transmitting control actions to amplifiers 4. Various options are possible. Consider two of them, illustrated by the structural diagrams of figure 2 and figure 3.

In direct control, when the difference signal Δu (ω _n ) is immediately supplied to the control input of the 4-n amplifier, the transmission coefficient of the latter will also change randomly, as well as the value of Δu (ω _n ). Such a control mode is implemented if a constant logic level “1” is supplied to the control input of the CO of the shaper 9 commands. In this case, through the keys 9, the control signals Δu (ω _n ) freely pass through the feedback circuit to the control inputs of the amplifiers 4. The second option provides for the supply of control signals in a pulsed mode, during the time-limited action of the logical “l” at the input of CO. For example, by a single corrective action during the entire playback session of a particular music program.

The application of the first of the described control modes is possible at relatively large averaging intervals and some uniformity of musical material, then continuous correction may not be noticed at all, especially when playing modern popular music on middle-class household equipment. This control method is implemented relatively easily and inexpensively by using electronically controlled amplifiers. The second option is preferable when correcting musical programs with complex content, for example, recordings of a symphony or jazz orchestra. To the reproduction of such material by potential listeners, quite high demands are made, characterized by tracking the smallest nuances in the sound picture, not to mention the general timbre coloring. Of course, to listen to such material and with existing requirements for fidelity, they develop and use High End equipment, which differs mainly in a wide dynamic range with minimal linear and non-linear distortions. In order to minimize distortion in the path of the useful signal, they seek to reduce the number of active elements, in particular electronic gain control devices. In such designs motorized passive regulators of analog type have long been used, easily remembering their last state (in this case, the position of the rotor of the electric motor). In addition, motorized regulators are in good agreement with the pulse correction mode, since such regulators do not require constant maintenance of the control voltage on them. In view of the foregoing, it is easy to understand that in professional and high-quality household frequency correction paths, it is advisable to use a pulse control mode with motorized regulators. However, in this case, the moment of introduction of the correction signals will be set externally, for example, by a sound engineer. At the same time, the formation of control actions will occur automatically and continuously.

The psychoacoustic processor described above, in contrast to the known frequency correction devices, supports a given shape of the spectrum of the output signal, regardless of the spectral features of the input signal, automatically adapting to it. (Of course, this is possible if the spectrum of the original signal contains components that need to be amplified or attenuated.) Using the inventive processor, which is essentially an adaptive equalizer, allows, for example, to automatically take into account the psychophysiological features of hearing in accordance with the isophonic curve previously introduced into the processor, specifying the relative levels of frequency components within the audible spectrum. In addition, the processor under consideration, by means of a once experimentally determined reference spectral envelope, can be used to correct acoustic flaws in the room for listening to music programs.

Claims (1)

A psychoacoustic processor comprising a controlled tone block and a control unit, the control output of which is connected to a control input of the controlled tone block, the input and output of which are the input and output of the processor, characterized in that the controlled tone block is n-band, the control block is n-channel, each the channel of which serves to control the timbral unit in only one frequency band, n-band timbral block outputs connected to the n-band information input of the control unit are used for I have a separate output of information in each frequency band, and the n-way reference input of the control unit is the reference input of the processor, which serves to set the shape of the spectrum of the output signal.

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