JP2002529757A - Method and apparatus for performing level compensation on an input signal - Google Patents

Abstract

SUMMARY A method and apparatus for level compensation that compensates for changes in signal level caused by a signal processor are presented. The original raw input signal (10) and the output signal from the signal processor (11) are provided to a comparison unit (13) which determines the difference in signal level between the two signals. The output of the comparison unit (13) is provided to the level compensator (14) together with the output signal from the signal processor. Next, the level compensator (14) corrects the signal level of this output signal.

Description

DETAILED DESCRIPTION OF THE INVENTION

BACKGROUND OF THE INVENTION The present invention relates to a method and an apparatus for performing level compensation on an input signal. In particular, the invention relates to a method and apparatus for adjusting the output level of an input signal to compensate for level changes made to the input signal by a signal processor.

[0002] Devices in the category referred to as "signal processors" are well known in the art. Such a processor may be used to modify an input signal such as music (or other audio sound) that has been converted to an electronic analog or digital signal. Common audio processors include tone modification (e.g., equalizer or EQ).
, Vocoder, distortion effect, chorus effect, flanger effect (flanger effect), ring modulator, wah-wah effect), dynamic correction (compression, expansion,
Tremolo, vibrato, etc.), reverberation, delay, many other modifications, and combinations thereof. Other processors are well known in the art, such as common video processors (eg, color filters) and other processors used in various fields. All such processors may have the potential "side effect" of changing the absolute peak level of the original input signal with respect to other signals or devices present. This change in peak level is usually undesirable and is often not readily apparent to the operator, and can often easily affect the decision of the operator of the apparatus without the operator knowing it.

In the field of audio, there are three common reasons for “equalizing” speech: 1) removing problematic noise (eg, removing unpleasant air conditioning hum). 2) create a pleasing musical tone (eg, add some bass to make the singer's voice a pleasing “husky” voice); and 3) create an effect (eg, the Beatles) “Yellow Submarine” (“Yello
w Submarine "), as if singing through a 1920s megaphone.

[0004] An equalizer (equalizer) is used to enhance or cut off a part of speech for some purpose.
), The volume of the sound to be equalized (sound) is actually increased (when emphasized) or decreased (when interrupted). As is also well known in the art of speech engineering, it is important to allow a comparison between the original speech and the equalized speech as part of the process using an equalizer (or other speech processor). . This is achieved by a "bypass" switch that selects whether the audio passes through the equalizer circuit or bypasses it.

[0005] Acoustic psychology concerns and consequences : A study of how people hear sounds
Generally called psychoacoustics. A phenomenon well known in this field was the first Fletcher (
It is codified by Fletcher and Munson and is most commonly displayed as a graph called a "Fletcher-Manson curve" or an equal loudness contour. The data show that the human ear and brain organs interpret loudness as a function of frequency, with two distinctive effects:

[0006] 1-people hear sounds of different frequencies unevenly (at the same volume level, low-frequency and high-frequency sounds seem quieter than middle-frequency), and 2-the whole sound is loud. Over time, the sound becomes more uniform. For very loud voices, the above effects are reduced or diminished. That is,
When the volume is increased, people will hear more bass and treble, which is why some people listen to music at a very loud sound.

[0007] Humans are typically within the frequency range formed by the human voice (about 100 Hz to 4 k
Hz) is easier to hear than higher or lower frequency sounds. Both the low bass guitar note and the high violin sound must be played with significantly more energy than the voice if you want those instruments to sound as loud as the voice.

[0008] The specific effects outlined above interact with each other. As the voice gets louder, the difference in frequency has no effect on hearing, and at a very large level the amount of energy for the bass, voice and violin that sounds as loud as one another is about the same.

[0009] This is commonly experienced when using stereo equipment. If the volume is at the normal listening level, the compact disc has acceptable sound. When the volume is lowered too much, the music suddenly sounds like there is not enough bass and there are not enough cymbals or other high frequencies (the Fletcher-Manson curve shows how low the low and high frequencies are reduced). Described). Stereo devices are often supplied with "loudness" control (either on / off switches or variable knobs) to attempt to compensate for this (very limited success). But only). Since the Fletcher-Manson curve is the average of data empirically derived from many people, it is an approximation for the experience of any given individual.

Audio engineers have been working on this phenomenon many times. Below are two examples of how this technology is addressing this problem.

The first example relates to an instrument that produces only one sound, a tom-tom on a drum set. When recording a tom-tom, audio technicians use an equalizer to try to make it sound better. For example, 400-80
Pleasant sounds can be heard near the fundamental resonance of the drum between 0 Hz. Also, just above this frequency range is very unpleasant speech. The technician can use a low-pass equalizer (which blocks high frequency signals) with control to select a frequency (which can select the point above which sound is cut off). This allows the technician to block the unpleasant range and then fine-tune the low frequency to find the spot where the drum sounds best.

Initially, the technician cuts off unpleasant frequencies above the pleasant range. However, when doing so, the sound becomes very low, so the volume is greatly increased. Next, the technician fine-tunes for the best sound. The drum itself has a different degree of loudness at each frequency, and the best place to set the control is in the frequency range where the drum sound is relatively low. If the best part of the drum sound is very low, it may not be perceived (again due to the Fletcher-Manson effect).
From experience or training, the technician raises the volume (again) to help hear the best sound. Next, a bypass switch is used to see if the equalized signal has improved over the original (non-equalized) audio. If the unequalized sound gets too loud, it becomes impossible for the technician to determine what has been achieved (as the sound gets louder, as explained by the Fletcher-Manson curve, It almost always sounds rich and good, and it can hurt technicians' ears because the sound is too loud (because the equalized sound must be turned up too loud to hear it). Also, if the volume is increased, the speaker may be damaged because the volume is too loud.

In a second example, a singer's recorded track is mixed with the rest of the band (drums, bass, guitar, piano, etc.). As described above, music is heard "bassier" when the volume is increased, or "low bass" when the volume is reduced, as described by the Fletcher-Manson curve (
bass “bassy”) ”). As more low frequencies are added to the sound by the equalizer, the total volume increases.

If the recording of the voice sounds “thin” due to poor microphone placement,
Technicians may try to compensate by raising the low-pass equalizer. The sound of the voice improves, but the extra lows make the voice louder than the rest of the band, and will be lowered to a reasonable level. The result is that the voice no longer has enough bass (due to the Fletcher-Manson effect). Therefore, this process is repeated several times until satisfactory results are achieved.

[0015] A well-known technique for solving the problem addressed above is described in the following complicated procedure.

1—Select a channel (“EQ Ch.”) To be equalized.

2—Set up another channel that duplicates the original non-equalized input (“UNeq Ch.”).

3-EQ Ch. (I.e., listen to 'SOLO' EQ Ch.-only that channel and turn off all other channels).

4-EQ Ch. Adjust the equalizer settings for ま で until you are satisfied.

[0020] 5-Solo EQ Ch. And UNeq Ch. Are performed alternately. That is, a-EQ Ch. Of the output level of UNeq Ch. Match the output of

B—if required EQ Ch. Adjust the equalizer settings for.

6- When mixing (with other audio channels turned back on): a-EQ Ch. And UNeq Ch. Set both levels (at one time).

B-EQ Ch. And UNeq Ch. To determine and adjust the equalizer settings.

7—Adjust as needed and repeat steps 3-6 as needed.

The above solution is not only complicated, but many performers and many audio engineers do not understand or recognize this problem or how to compensate for it. This means that the amplified sound of many clubs, concerts, weddings, etc. is often poor (
For example, it is evident from the fact that it is harsh, piercing, bass is too strong, too loud, etc.).

In the art, there are two very common devices (other than equalizers, reverberators, and other processors) that actually change the volume of the source matarial. These devices change moment-to-m
oment change).

1—Loudness Control (a switch or knob found on many home listening devices) is an attempt to compensate for the Fletcher-Manson effect at low listening levels. This is a special tone adjustment that emphasizes both lows and highs with one control (but the lows and highs emphasized by this are the same as the normal lows and highs controls) Somewhat different).

A 2-AGC (Automatic Gain Control) amplifier circuit is used to compress (reduce) the dynamic range of a signal. This means that no matter how loud (or soft) the original signal (input) is, the output is always the same volume. This includes household products (VCRs and inexpensive cassette tape recorders) that usually have low-quality built-in microphones, devices that intentionally sacrifice voice quality for speech intelligibility (CB and ham radio,
On some phones). The output does not depend on either the sound source or any particular settings. Audio fidelity is deliberately sacrificed to overcome the limitations of a noisy environment or to allow the use of inexpensive devices. In practice, such devices usually do not even have an input volume control, but some high quality products offer the option of using an AGC circuit or an actual input level control circuit. Such devices have an on / off switch for the AGC circuit. When the AGC is turned on, the actual input circuit is bypassed, and when the AGC is turned off, the AGC circuit is bypassed.

AGC generally involves a dynamic processor. This is an expert device that allows the user to control the degree to which the dynamic range reduction occurs. With AGC, such devices actually change the dynamic range of the input signal.

[0030] In view of the above, intelligent correction of levels can be made to compensate for level changes caused by the processor, without the need for compensation, or even the need to understand the need for compensation. There is a need for a method and apparatus that allows a user to make better choices about the use of a device.

SUMMARY OF THE INVENTION These and other needs are satisfied by the method and apparatus of the present invention. According to embodiments of the present invention, intelligent correction of the level of the output signal is made to compensate for peak level changes.

In the first embodiment of the present invention, the unprocessed sound source signal is first compared with the final processed signal to determine the amount of change, and then the level is adjusted using the comparison. Standard components are used to: 1-compare the pre-processor signal (pre-processor signal) (the original unprocessed signal) with the post-processor signal (after processing) (post-processor signal); Record the level difference between them, and then 3- build a stand-alone device that compensates for the processed signal, for example, by matching the post-processor signal level with the pre-processor signal level.

In this embodiment, one goal is to provide a single overall level change, and the combined use of any number of processors results in one final level change. So this circuit works for any combination of changes. Thus, only one circuit is required for a signal path, regardless of the number of processors between the "pre-" and "post-" read points.

Also, the processed signal may be provided to a level compensator, and the output of the level compensator and the unprocessed signal may be provided as inputs to a comparator. The output of the comparator then provides a feedback signal to the level compensator. Thus, the level compensator modifies the level of the processed signal until the output of the level compensator equals the level of the raw signal.

In a second embodiment of the invention, the compensation provided for the processed signal is derived empirically, rather than by direct measurement. The actual level change resulting from the operator using a given processor control is measured. Each control on the processor that "generates" an undesirable level change is also used to control a circuit that provides level compensation by effecting the desired reverse level change. This embodiment knows the range of changes in the properties of the signal to be processed,
Suitable for situations where the range of changes in the processor control settings is known. According to this embodiment of the invention, developing compensation for a single control of the processor includes the following steps. 1 While listening to the normal signal source and observing the typical signal source,
For example, the control of the processor is adjusted by rotating the rotary knob by 30 degrees. 2 For each position of the control, the resulting change in the overall signal level is recorded (by an objective measurement and / or by an experienced observer matching the perceived level). 3 Repeat using sufficiently various sound source materials. 4. Edit the results and design a circuit to cancel the level changes caused by changes in processor control. This circuit should be adjusted by the same controller that adjusts the parameters of the processor. The two implementations using a standard potentiometer as a controller are as follows.

A. For example, having a separate gain circuit in series with the processor, the gain being adjusted by a separate potentiometer element coaxial with the processor controller; b. Modify the processor to incorporate the desired change in gain. The ability to individually adjust compensation for particular situations and / or uses is a particular advantage of this method. Another advantage is that the circuitry required to achieve it is minimal (and therefore inexpensive).

In a third embodiment of the present invention, there is provided a method of compensating for a level change when the processor itself changes the level dynamically, such as a compressor, expander, etc. Although it is desirable for them to change the dynamic range of the source material, these devices have created a peak level (R) for the rest of the acoustic environment.
It is usually undesirable to change the loudest spot).

In audio, such devices generally operate by comparing the level of an input signal to a threshold, which is a constant reference voltage, whose level is controlled by a user using a controller. Is set. When the signal source crosses the set threshold level, the signal is processed via a single control that most often (for example) sets the ratio of the applied processing. At any given setting of these two controls, the total effect on the peak (maximum instant) of the signal is the simple difference from the product of the threshold and the ratio setting in decibels. Function. This function is often expressed as a control voltage, and generally is a VCA (voltage controlled amplifier) that constantly performs the desired change in level.
amplifier).

An embodiment of the method uses the same control settings for the operation of the processor operating the second control signal path using a separate constant reference voltage as input. First
The control signal is responsive to the audio input signal. The second control voltage depends on the control knob of the processor, but not on the input signal, resulting in a desired constant level of compensation. This compensation control voltage is applied to the same device (eg, VC
A). Also, since the amount of level compensation is determined by the same control that sets the change in peak level output, the amount of level compensation automatically tracks (and accurately compensates for) changes in processor settings. The result is an integrated processor and level compensation system.

(Detailed Description) Referring to FIG. 1, a block diagram of an apparatus for performing level compensation according to an embodiment of the present invention is shown. Level compensation refers to performing one of the following steps.

1. As the processor reduces the overall level of the input signal, the apparatus of the present invention increases the gain to match the original signal level.

[0042] 2. As the processor increases the overall level of the input signal, the inventive device reduces the gain to match the original signal level.

This allows the user to determine the effect of the processor without being “fooled” by signal level differences. This can be done without any user involvement, which is a tedious and tedious process, as described above. A switch is available to disable the circuit if desired. As described in the following embodiments, the present invention can be applied to compensate for the volume of an audio signal. As those skilled in the art will appreciate, the present invention (e.g.,
The circuit of FIG. 1) can be applied to compensate for processor-generated level changes in other types of signals (eg, video signals, MRI, CAT scans).

Referring to FIG. 1, an input signal 10 is provided to a processor 11 whose signal is to be modified (eg, a digital signal processor). For example, the signal is modified by the processor by lowering the level of a particular frequency band of the signal (eg, as in an audio filter). The original raw signal and the modified signal from the processor (ie, the modified signal) are provided to the comparator 13 (ie, the modified signal is provided either before or after the level compensator 14).
. The comparison device 13 outputs a difference signal indicating a difference between the original signal and the corrected signal.
Level compensation circuit 14 receives the modified signal and the difference signal and performs level compensation on the modified signal (eg, corrects for gain increase or decrease generated by processor 11). Level compensated signal (level compensated signal) if desired
May be provided to an output device 12 such as a recording device, a speaker, or the like.

As indicated above, one example of a processor is an equalizer-like sound quality (tone
quality). Sound quality, also called timbre, is the result of the balance of different frequencies measured in hertz (cycles / second). Human hearing is also referred to as the sound spectrum, from 20 Hz to 20 KHz (2 Hz).
0,000 Hz). Equalizers, tone adjustments, filters, etc. are devices that change sound quality. They all operate by allowing the volume of selected portions of the audio frequency range to be increased or decreased. A device that raises or lowers the entire frequency range uniformly is simply a volume control.
control). The equalizer operates as a volume control for a limited part of the listening range.

A first embodiment of the present invention is shown in connection with FIGS. Referring to FIG. 9, a configuration diagram of this embodiment is shown. FIG. 10 shows a more detailed example of this circuit (ie, a Root-Mean-Square (RMS) level compensation system). FIG. 9 shows a post-processed signal (“audio signal”) from the signal processor 61 and a pre-equali
Three signals are input to the RMS level compensator 62: a "control" signal 10 and a post compensator "control" signal input from the RMS level compensator 62. The RMS levels of the input signal and the control signal are measured by the RMS converter 72A and the RMS converter 72B (FIG. 10), respectively, and compared (for example, by the window comparator 73). If the control signal is greater than the output signal, the gain of the comparator is increased by adjusting the digital control resistor 78, since the signal processor 61 reduces the level of the signal at that point. If the control signal is small, the gain will be reduced. The rate of change of the gain is determined by the frequency of the oscillator 75 and the resolution of the resistor. The gain changes until the output level matches the control signal level or until the gain limit is reached. If there is no output signal detected upon sensing by the signal detector 76 and if the mode of operation is a "hold" mode (as determined by a Match / Hold Switch 77), The value of the resistor does not change and the gain remains the same.

Referring to FIG. 11, a more detailed schematic diagram of the circuit of FIG. 10 is shown, where the block shown as DS1666 includes a digitally controlled attenuator.
Acting as a controlled attenuator, the range is limited by resistor R2, allowing the output gain stage of operational amplifier AR2 to boost the audio signal. Up / down, increment, which are the control inputs to the DS1666
), And chip selection are "window comparator", "oscillator" and "
Operated by a "disable" circuit. The "RMS converter (RMS-conv)" converts the "control (CTRL)" signal and the "output (OUT)" signal into RMS log-equivalent sig.
nal). The “hysterisis” circuit has “CTRL” set to “OUT”.
Is much larger or smaller. If there is a large difference, the "oscil" will increment or decrement the DS1666. The direction is determined by the upper "hysteresis" circuit output. XOR gate U3 provides the required timing of the chip select input along with the increment input.

Referring to FIG. 12, a circuit diagram of the sub-circuit of FIG. 11 is shown. RMS conversion is
It is achieved by a THAT2252 log converter and rectification (by diode D1 and capacitor C3). Resistor R16 provides its own release time by discharging capacitor C3. The oscillator is a standard 555 timer circuit activated by a reset pin. Note that the comparator VR1 in the hysteresis block has a degree of hysteresis that allows for an approximate level match within, for example, ± 2 dB. The signal detection VR2 provides a negative going AC signal (n
egative going AC signal). This serves as a safety feature to prevent the compensation circuit from accidentally responding to inappropriate signals, especially during the recovery time of the RMS converter.

A possibly more accurate version of this embodiment of the invention uses the same steps as above, but the comparison involves an analysis of the frequency spectrum of the signal before and after processing. The applied level compensation is based on the signal level difference and the frequency at which the level difference occurs. For example, a Fast Fourier Transform (FFT) is used, but other frequency band inspection techniques are possible. An example of this procedure is shown below. 1—FFT of the pre-processing signal and the post-processing signal is performed. 2-Both RMS values are derived for the selected bandwidth. In typical audio work, 1/3 octave is common, which will calculate about 30 bands for each of the two FFTs. Even less bandwidth may be sufficient, especially if carefully determined. 3- "weigh" the result of step 2 by the "average" Fletcher-Munson curve. 4-Subtotals before and after processing are obtained by summing the results of step 3. 5. Take the difference between the subtotals of step 4 (subtract one from the other). 6-Adjust the final output level according to the result of step 5.

Because the purpose of this embodiment of the invention is a single global level change, no matter how many processors are in the combination, only one final level change results. , This circuit works for any combination of changes.
Thus, only one circuit is required for a given signal path, regardless of the number of processors between the "before" and "after" read points.

A second embodiment of the present invention is shown in FIGS. 2 to 8, where the level compensation is derived empirically. Integrated level compensation for audio signal processors
The derivation of (integrated level compensation) depends on the characteristics of the signal to be processed and the estimation of the level to be compensated. That is, it must be assumed that a change in the processor settings will affect its level as well, since all its expected input signals will be similar. For example, a notch filter can be used to modify the sound of a cymbal, and setting the notch frequency to 400 Hz would increase the hearing level by 6 dB over the 1.25 kHz setting for most cymbals. Become. To maintain an equal level, set the 400 Hz frequency setting to 1.25 kHz.
When changing to z, the gain must be compensated by -6 dB. Referring to FIG. 2, an overview of how to track frequency and gain via dual potentiometer 24 is shown.

An example of the method of this embodiment is shown below. 1 Select the desired processor. In this example, the frequency is variable (400 to 125
0 Hz) and a fixed depth (depth) notch filter (as shown in FIG. 3). 2. For example, select a typical sound source when hitting a cymbal. For example, a starting parameter and an output level having a notch frequency of 1.25 kHz and an output gain reference of 0 dB are set. 3 Change the parameter in selected increments and adjust the gain if necessary to match the previous level. The new parameter settings and new gain settings are recorded. Iterate over the useful range of variable parameters. For example, as shown in Table 1, Row 1 shows the change in relative gain required to maintain equal levels of cymbal speech at 1/3 octave intervals from 400 Hz to 1.25 kHz. 4 Repeat steps 2 and 3 for many similar sources. Table 1 is data from 10 similar cymbals.

[0053]

[Table 1]

5 Determine the desired average relative gain at each increment of the parameter. Equation Eq. Using 1, the average of each column in Table 1 can be calculated. This is also useful for calculating the gain range for each frequency. Equation Eq. 2 first adjusts the relative gain data and collects it around the total average point to optimize the deviation. Equation Eq. 3 and Eq. 4 calculates the adjusted column maximum and minimum values. The result is shown in FIG.

[0055]

(Equation 1)

6 In the analog circuit of this example, the frequency setting must be correlated with the gain setting by the potentiometer. For example, the use of a dual potentiometer allows the addition of a gain circuit (FIG. 4) so that gain and frequency can be controlled simultaneously from a single knob. Equation Eq. 5 and Eq. 6 describes the relationship between notch frequency, gain, and resistance (f (x), G (x), and R (x)) in this example. The results are shown in Table 2.

[0057]

(Equation 2)

[0058]

[Table 2]

In Formula 6, the value of resistor R 4 determines the shape of the curve shown in FIG. 7 Find the closest match between the gain setting and the frequency setting. Since the frequency circuit is pre-determined as a functionally optimal configuration, the gain circuit must be adjusted and matched. By appropriate selection of the resistance R4, the resulting gain change as seen in the line of FIG. 5 can be approximated by the desired change of gain for frequency as shown by the diamond in FIG. . The resistor R2 sets the overall gain. 8 Repeat for other parameters.

An alternative to the exemplary method in step 6 is to change the integrated gain with the frequency change by operating the filter circuit (see FIG. 5A and also FIGS. 6 and 7) and the digital version (see FIG. 8). included. One version of the digital version has average gain data stored in memory and interpolates gain at the frequency between measurement points. Another variant uses the relationship between f (x) and G (x) to calculate gain.

FIG. 5A shows a notch filter with interconnect depth and frequency level compensation. Gain transfer function for audio frequency
n) is the equation Eq. Indicated by 7.

[0062]

(Equation 3)

The filters are arranged so that a single element of the potentiometer can change both frequency and gain. FIG. 6 shows a functional diagram of a standard filter 41, a Sallen-Key second-oder high pass filter, which is a dual potentiometer indicated by R21 and R22 in element 41 of FIG. Has been manipulated to produce frequency-gain tracking. Normally, output # 2 (see FIG. 6, element 43) is connected to ground. Alternatively, connecting R22 to the current amplifier AR6 of FIG.
4, at the output # 1 (see FIG. 6, element 43), the same transfer function and the current R3 + R22
A second output that is inversely proportional to R = R4 + R21 = R3 + R22 and C =
It should be noted that when C1 = C2, the transfer functions of the two outputs both pass high at the pole of f = 1 / (2πRC) and Q = 0.707. Mixer (mixer)
Is -R7 / (R8 + R). Thus, the frequency and gain of the high pass filter can be controlled by a single control element of the potentiometer. Referring to FIG. 8, a digital version of the level compensation is shown next to the known system. It is well known in the art to provide a user interface for selecting a function (e.g., setting a high-pass filter) and to provide an input regarding frequency (e.g., a cut-off frequency of the high-pass filter). The filter coefficients required to achieve the desired function are selected from memory based on the input frequency.
Its output is the resulting adjustment of the filter frequency. According to an embodiment of the present invention, the memory is further used to store a gain value selected based on a frequency input from a user. Accordingly, level compensation is performed based on the selected frequency, as described in detail above.

In addition to the need to separately compensate for each control that produces a level change, there are situations where this solution is not sufficient. In predictable situations (where the combination of sound source and equalization type is consistent), the above process yields very good results. The more limited the type of sound source material and the more limited the range / effect of the equalizing element, the easier it is to develop an empirically derived compensated processor. However, the more diverse the sound source, the more tests are required, and eventually the test results will be inconsistent.

The most complicated situation is that for all possible human listeners (with the Fletcher-Manson curve being an average and each person being different from that), for all possible human listeners For a possible sound source (speech, music, sound effect, etc.), a design with a knob having a wide range (e.g., frequency control in the range of 20-20 kHz) would be performed. This is not possible in practice because changes in the sound source will interfere with a given equalizer. For example, assume that a bass boost frequency knob that works well for a wide range of sound sources, such as an orchestra, is designed.
It is likely to be performed due to the Manson effect). It is assumed that there is a portion of the music where only the music triangle is played. When adjusting the equalizer knob at low frequencies, the triangle has no low frequencies that can be emphasized,
Triangle has no effect on tone. The change is not expected to be audible at all, but the gain compensator continues to add levels, so turning the knob back and forth will change the volume of the triangle. In the case of a triangle (or any sound that does not have a low frequency), this knob turns into a volume control and confuses the user. That is, a circuit that compensates based solely on the setting of the equalizing knob (as determined by any method, including those described above) will not perform well in significantly different situations. In such a situation, the first embodiment described above is preferred.

A third embodiment of the present invention is shown in FIGS. 13 to 16 as a dynamic processing compensation circuit. Dynamic processors limit peaks to prevent overload,
Change the signal gain to achieve any number of effects, such as compressing the dynamic range of the signal, expanding the range, or maintaining a longer note. Since the total gain varies based on the input signal level, the sound source adjustment compensation method described above
It is very difficult to compensate for parameter adjustments using the (source-adjusted compensation method). Such methods tend to undo the work of the processor. Instead, according to embodiments of the present invention, expected gain changes can be compensated for in the following procedure. 1. Assume the desired normalized listening level after processing. The simplest normal point is 0 dBV. 2. Without compression at a gain of 1, a change in gain from compression of the 0 dBV input signal would be matched by an equal and opposite gain change to maintain the 0 dBV output since a 0 dBV input signal would produce a 0 dBV output. Should. For example, if it is desired to reduce the dynamic range from 90 dB to 60 dB, -90
Starting from dBV, a compressor is used to gradually reduce the gain as the input signal rises to 0 dBV. However, the output of 0dBV input is -30dB
V. It is more probable that the range of −60 to 0 dBV is desirable instead of −90 to −30 dBV. Therefore, the gain should be increased by 30 dB. 3. The gain adjustment is shown in FIG. Using a voltage controlled amplifier (VCA) having positive and negative control voltage inputs, a constant control voltage equal to the control voltage expected at the other terminal having an input of 0 dBV is provided at one terminal. For example, a 0 dBV input signal at a certain parameter setting will generate +100 mV at the negative control port to reduce the gain. The +100 mV voltage at the positive control port negates this reduction. Also, equal and opposite control voltages may be combined at a single terminal of the same circuit.

To implement this method, the same circuit is used to modify both the RMS output signal and a constant voltage source equal to the RMS output at the normalized input level (in this case, 0 dBV). . In this embodiment, a dual potentiometer is used for the threshold and ratio knobs and both control voltage circuits can be adjusted simultaneously so that the two control voltages change in tandem. Other circuits can be used to achieve the same function.

[0068] Referring to FIG. 14, a graph of the transfer function V _in pairs V _out of _the ordinary compressor is shown. Note how the peak level decreases for low thresholds and large ratios. FIG. 15 shows a graph of a level compensated compressor at various settings with the level normalized to 0 dBV (FIG. 15A shows the same effect normalized to 14 dBV). For example, at a 2: 1 ratio, compression of the -16 dBV threshold reduces the gain by -8 dB at 0 dBV input. Therefore, the gain is compensated by +8 dB. At ∞: 1, the gain must be compensated for +16 dB. It should be noted that normalizing at high levels of the dynamic range will increase the loudness of most of the signals, which are generally below 0 dBV. However, this gain adjustment is usually desirable, unless the purpose is simply to "prune the top." In FIG. 16, a detailed schematic of a dynamic processor using a THAT4301 dynamic processor chip is shown with an RMS converter and log-based (log
-based) VCA.

Embodiments of the present invention compensate for level differences caused by changes in the signal processor. The compensator should only modify the level based on the modification of the processor's signal. To avoid false activation of the level compensation, a detection circuit or switch should freeze the level compensator from changing its modification when the signal processor is not being adjusted.

[Brief description of the drawings]

FIG. 1 is an overall configuration diagram of an apparatus configured according to an embodiment of the present invention.

FIG. 2 is an overall configuration diagram of a second embodiment of the present invention.

FIG. 3 is a detailed circuit diagram of the processor of FIG. 2;

FIG. 4 is a detailed circuit diagram of the amplifier of FIG. 2;

FIG. 5 is a graph showing the relationship between frequency and level compensation in the example operation of FIGS.

FIG. 5A is a notch filter with interconnect depth and frequency level compensation configured according to an embodiment of the present invention.

FIG. 6 is a block diagram of a level compensator configured according to an embodiment of the present invention, in which a single element of a potentiometer can change both frequency and gain.

FIG. 7 is a detailed circuit diagram of the device of FIG. 6;

FIG. 8 is a configuration diagram of a digital level compensation system configured according to an embodiment of the present invention.

FIG. 9 is a configuration diagram of an apparatus configured according to the first embodiment of the present invention.

FIG. 10 is a detailed example of the device of FIG. 9;

FIG. 11 is a more detailed circuit diagram of some elements of FIG.

FIG. 12 is a more detailed circuit diagram of some elements of FIG.

FIG. 13 is a configuration diagram of a third embodiment of the present invention.

FIG. 14 is a graph showing the effect of compression known in the art.

FIG. 15 is a graph illustrating the effect of compression with level compensation according to an embodiment of the present invention.

FIG. 15A is a graph similar to FIG. 15, normalized to different levels.

FIG. 16 is a detailed circuit diagram of a dynamic processor that implements a third embodiment of the present invention.

──────────────────────────────────────────────────続 き Continuation of front page (81) Designated country EP (AT, BE, CH, CY, DE, DK, ES, FI, FR, GB, GR, IE, IT, LU, MC, NL, PT, SE ), OA (BF, BJ, CF, CG, CI, CM, GA, GN, GW, ML, MR, NE, SN, TD, TG), AP (GH, GM, KE, LS, MW, SD, SL, SZ, UG, ZW), EA (AM, AZ, BY, KG, KZ, MD, RU, TJ, TM), AL, AM, AT, AU, AZ, BA, BB, BG, BR, BY , CA, CH, CN, CU, CZ, DE, DK, EE, ES, FI, GB, GD, GE, GH, GM, HR, HU, ID, IL, IN, IS, JP , KE, KG, KP, KR, KZ, LC, LK, LR, LS, LT, LU, LV, MD, MG, MK, MN, MW, MX, NO, NZ, PL, PT, RO, RU, SD, SE, SG, SI, SK, SL, TJ, TM, TR, TT, UA, UG, UZ, VN, YU, ZWF Term (reference) 5D108 AA01 AB09 BF02 5D378 JB00 JC09 JC10

Claims (19)

[Claims] 1. A level compensation system for compensating a signal level change by a signal processor from a first signal level for an input signal to a second signal level for a corrected signal in at least one frequency band. The system is adapted to be coupled to an output of the signal processor, the input signal;
A comparison unit adapted to receive the modified signal from the signal processor, wherein the comparison unit is adapted to determine a signal level difference between the input signal and the modified signal. A comparing unit coupled to an output of the comparing unit and adapted to be coupled to an output of the signal processor, the level compensating unit being adapted to correct the signal based on the signal level difference. A level compensation unit adapted to modify the signal level of the signal. 2. The level compensation system according to claim 1, wherein said input signal is an audio signal. 3. A signal processor coupled to the comparison unit and the level compensation unit, the signal processor comprising: an equalizer, a vocoder, a distortion effect component, a chorus effect component, a flanger effect component, and a ring. 3. The level compensation system according to claim 2, comprising a signal processor including at least one of a modulator, a wah-wah effect component, a compressor, an expander, a tremolo component, a vibrato component, a reverberation component, and a delay component. . 4. A level compensation system for compensating a signal level change by a signal processor from a first signal level for an input signal to a second signal level for a corrected signal in at least one frequency band. A comparison unit adapted to receive the input signal; and a level compensation unit adapted to be coupled to an output of the signal processor and an output of the comparison unit, wherein the level compensation unit is A level compensation unit adapted to modify the signal level of the modified signal based on an output of the comparison unit, wherein the comparison unit outputs the input signal and a signal output by the level compensation unit. Level compensation system adapted to output a signal indicative of a signal level difference between . 5. The level compensation system according to claim 4, wherein said input signal is an audio signal. 6. A signal processor coupled to the comparison unit and the level compensation unit, the signal processor comprising: an equalizer, a vocoder, a distortion effect component, a chorus effect component, a flanger effect component, and a ring. The level compensation system of claim 5, comprising a signal processor including at least one of a modulator, a wah-wah effect component, a compressor, an expander, a tremolo component, a vibrato component, a reverberation component, and a delay component. 7. The first route mean squared converter adapted to receive the input signal, the comparison unit further adapted to receive the input signal, and the second route adapted to receive the modified signal. 5. The level compensation system of claim 4, comprising: a mean squared converter; and a comparator coupled to the first and second root mean squared converters. 8. The circuit, wherein the level compensation unit further comprises a digitally controlled resistor having a first input coupled to the comparator and a second input adapted to receive the modified signal. The circuit is
9. The level compensation system according to claim 8, comprising a circuit adapted to modify a volume level of the modified signal based on an output of the comparator. 9. A method for performing level compensation on an input signal, the method comprising: processing an input signal in a signal processor to generate a corrected signal; and adjusting a signal level of the input signal to a signal level of the corrected signal. And modifying the signal level of the modified signal based on the comparing step. 10. The method of claim 9, wherein said comparing step calculates a signal level difference between said input signal and said modified signal. 11. The method of claim 10, wherein said modifying step modifies a signal level of said modified signal to match a signal level of said input signal. 12. A method for performing level compensation, comprising: (a) supplying an input signal to a signal processor; (b) adjusting a control of the signal processor by a predetermined amount; Comparing the level of the input signal with the level of the modified signal output by the signal processor; and (d) determining a difference value representing the level difference between the input signal and the modified signal. (E) repeating steps (a)-(d) for a plurality of input signals; and (f) controlling the signal processor control based on the difference value determined in step (d). Calculating the expected difference in signal level caused by the adjustment. 13. The method of claim 12, wherein step (b) includes adjusting the control multiple times. 14. The method of claim 14, further comprising: (g) modifying a signal level of the modified signal based on the expected difference in the signal level calculated in step (f). 15. A method for performing level compensation, comprising: (a) providing an input signal to a dynamic signal processor;
Wherein the dynamic signal processor is adapted to change a gain of the input signal depending on a predetermined ratio when the input signal has a signal level crossing a predetermined threshold; and (b) Providing an output control voltage signal from a dynamic signal processor to an amplifier that causes a change to the level of the input signal; and (c) normalized based on the setting of the predetermined ratio and the predetermined threshold. Providing a constant control voltage equal to an expected output control voltage signal from the dynamic signal processor of an input signal; and (d) based on a difference between the output control voltage and the constant control voltage. Modifying the input signal in the amplifier. 16. A circuit for controlling gain and frequency modification of an input signal, the filter circuit being adapted to receive the input signal and modify the signal level of the input signal in at least one frequency range. A filter circuit wherein the frequency range is selected based on a control input; a gain circuit adapted to receive the modified signal from the filter circuit and a signal from the control input. A gain circuit, wherein the gain circuit is adapted to modify a gain of the modified signal based on a signal from the control input. 17. The circuit of claim 16, wherein said control input selects a resistance value of each of said filter and said gain circuit. 18. The control input of claim 17, wherein said control input is part of a potentiometer.
Circuit. 19. The circuit of claim 18, wherein said control input is generated from a single control element of said potentiometer.