JPH05276593A - Digital audio signal processing unit - Google Patents

Abstract

PURPOSE:To provide the digital audio signal mixer in which production of switching noise of an effector is reduced. CONSTITUTION:The changeover of an effector 23 of a digital audio signal mixer is implemented with cross fading. Audio data are faded out at the changeover and the audio data through the effector 23 are faded in. Coefficient data multiplied with the audio data and the coefficient data multiplied with the data through the effector 23 are changed gradually respectively from 1 to 0 (fade-out) and from 0 to 1 (fade-in). A digital audio signal faded out and an output of the effector 23 subject to fade-in by cross fading are mixed by an adder 53 and the result is outputted. Thus, production of switching noise due to a discontinuous point of the audio signal caused by the changeover of the switch is reduced and the effect is especially effective for snap shot by the cross fade operation.

Description

Detailed Description of the Invention

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a digital audio mixer, especially a digital V mixer, which mixes a plurality of digital audio signals recorded by a large number of microphones or the like into one completed audio program.
The present invention relates to a digital audio signal mixer suitable for editing TR audio.

[0002]

2. Description of the Related Art In order to mix multi-system audio signals, an audio program is created by a digital audio signal mixer (hereinafter, referred to as a digital mixer) regardless of a digital source, in order to mix multi-system audio signals. Has become the mainstream. A digital mixer for mixing digitalized audio signals of a plurality of channels with a desired mixing ratio to obtain a new digitalized audio signal of a plurality of channels is configured as shown in the block diagram of FIG. There is.

First, an outline of the digital mixer will be described with reference to FIG. The digital mixer shown in FIG. 1 is a digital mixer to which a 32-channel digital audio signal is input and a 4-channel digital audio signal is programmed and output. In FIG. 1, first, as input channels, digital audio signals # 1 to # 32 of 32 channels are input from the input terminal 1.
The audio signal of 32 channels is selected by the routing switcher 2 and 16 channels (CH1 to C
H16) becomes an audio signal, and the output of the 16 channels is a mute switch 3, a de-emphasis / phase changing device 4, a delay device 5, a filter 6, an equalizer 7,
The volume passes through the insertion 8 and is adjusted by the channel fader 9. After that, it is added to the four buses via the assignment switch 10 and the master fader 11
4 outputs, volume controlled and programmed by
2 (PGM1 to PGM4).

The main parts of the digital mixer will be described in detail below. The 32-channel digital audio signals # 1 to # 32 are digital audio data and are generally AES / EBU format signals. Specifically, it is a DAT or CDP digital output signal in which additional information is added to audio PCM data.

The routing switcher 2 is a switcher that determines which of the digital audio signals # 1 to # 32 is assigned to CH1 to CH16.
1 to CH16 all have the same signal, eg input signal # 1
Can also be assigned. Mute switch 3
Is a switch for switching so as not to transmit a digital audio signal that the operator does not need. The de-emphasis device 4 is a device that restores a high frequency signal by removing (de-emphasis) this signal processing when a signal subjected to emphasis (high frequency emphasis) processing is input.

Next, the phase changing device 4 described in the same block as the de-emphasis device will be described. This phase changing device is a device for inverting the phase of a digital audio signal. Since this phase changing device is not particularly required when listening to CDs, music tapes, records, etc., which are generally sold, it is a function which is not found in audio amplifiers attached to audio devices that are usually sold. As its application, it is mainly used for phase correction of microphone during recording and recording. A microphone vibrates the diaphragm by sound pressure and converts the vibration into an electric signal, but depending on the microphone, a positive voltage is generated at a dense place of acoustic vibration (vibration of air) and a positive place at a sparse place. There are some that generate voltage, and when recording and recording using multiple microphones, the polarities will be different, and it is necessary to match the phases.If the phases are not aligned, acoustic vibrations cancel each other out, and particularly low-pitched bass. Disappears. For this reason, the input signal recorded by the microphone upon mixing needs to be phase-matched depending on the characteristics of the microphone, and this phase inverter (phase reverse) is required for the mixer.

When editing a digital VTR audio signal, the delay device 5 may cause a delay of several frames when a video is edited by a digital multi-effector (DME) or the like. It is necessary to delay the image according to the image, and a delay device is required for this purpose. The filter 6 includes a low-pass cut filter and a high-pass cut filter, and is used in a mixer to remove an effector and noise. The high-frequency cut filter may be accompanied by hiss noise when, for example, the source of the input audio signal is an analog signal recorded on a tape or the like, and is used to remove such noise. The low-pass cut filter is used to remove low-frequency noise around the wind or the like. The equalizer 7 is used when raising or lowering the signal level of a certain sound range of an audio signal, and is mainly used as an effector for producing a sound effect in a mixer.

Next, the insertion 8 will be described. The insertion 8 has a function of releasing the contact to the outside when the operator replaces and uses an external effector or the like (limiter, filter, equalizer, etc.) outside the mixer. pointing. That is, the passage of the audio signal is cut at the insertion point and the cut end is opened to the outside. Therefore, even if the insertion function is turned on, the sound is cut off at the insertion point unless any connection is made outside the device, so that the input audio signal is not output. More specifically, referring to FIG. 31, when the insertion is turned off, the switch 13 is tilted upward so that the input and the output are directly connected and there is no insertion function. On the other hand, when the switch 13 is turned on, the switch is tilted downward so that the input and output are both input and output from the insertion point 25 to the outside. Here, the effector that the operator wants to use is connected to the insertion out, and the output of the effector is connected to the insertion in. In this way, effects that the digital mixer does not have can be applied through an external device.

The channel faders 9 are CH1 to C
It has a function of adjusting the volume up to H16, and the volume is adjusted by these channel faders 9 (+12 dB
To -∞), the sound is sent to the mixing bus, and the sounds are mixed according to the assign switch 10.
The assign switch 10 is used for the channel fader 9
ON / OFF switch for deciding whether to mix the volume-controlled sound. It has a function of adding when the switch is on, and not adding when the switch is off, and an input to the mixing bus. Is.

Next, the master fader 11 outputs the sounds (for example, surround program) added to the four mixing buses according to the assign switch 10 to PMG1 (program 1), PGM2, PGM3, and PG, respectively.
M4, and these programmed PGM1 to PG
The output up to M4 is this master fader 1
Final volume adjustment by 1 (normal adjustment range is 0 dB
Therefore, it may have a gain though it is −∞. ) Is done. After the volume is adjusted by the master fader 11, it is again modulated into the AES / EBU format and output to the outside as the outputs of PGM1 to PGM4.

The main part of the digital mixer for the digital audio signal has been described above, but the problems of the conventional digital mixer will be described below. Therefore, first, problems of the conventional phase change and volume adjustment will be described. Conventionally, in a digital mixer having a phase changing function, as shown in FIG. 28A, audio data is multiplied by "1" by a multiplier 14 or by "-1" by a multiplier 1.
Phase inversion is realized by multiplying by 5. In this method, as shown in FIG. 28B, even if the signal 28A which is not phase-inverted is continuous, when the phase inversion switch 16 is switched to invert the phase, the waveform 28B is discontinuous depending on the timing of the phase inversion. As a result, the phase conversion waveform as shown in FIG. 28C is obtained. Due to this discontinuity, noise is generated at the time of switching.

Particularly in the case of digital audio signals,
Since the data is discrete data, in the phase conversion waveform as shown in FIG. 28C, the continuity of the data disappears at the moment when the switch is turned on, so that noises such as "buchi" and "bachi" are generated. In particular, in a digital signal, since there are discontinuities, the frequency component extends to ∞, and when actually performing D / A conversion, it is not possible to express the frequency component of one half or more of the sampling frequency, resulting in aliasing noise. When switching the digital audio signal, noise will be generated unless switching is performed so that there are no discontinuities.

To actually process the digital audio signal with the phase changing function, conventionally, in FIG. 28A, the phase is changed by multiplying "-1" by a DSP or the like. In the case of multiplication by DSP, the coefficient k is generally −1 ≦ k <1, and when multiplied by “1”, it becomes the input signal itself, and at “0.5”, the level is halved. "0" means no sound, "-1" means phase inversion, "-"
At 0.5 ", the processing is performed so that the waveform has half the level and the phase is inverted. Therefore, when an attempt is made to perform a phase inversion function on a digital audio signal, it can be realized by a circuit that selects whether the input signal is multiplied by "1" or "-1". However, such a phase changing method cannot solve the problem of signal discontinuity, and the waveform itself of FIG. 28C is output.

Next, the problems of the fader for adjusting the volume in the digital mixer will be described below. When the volume of a digital audio signal is digitally adjusted by a fader or the like, a DSP is used for actual signal processing.
The content of this processing is performed by operating the fader by multiplying the digital audio signal by a coefficient having a value of 1 to 0. For example, when multiplied by "1", the input signal is at the same level, and when multiplied by "0", there is no sound. In FIG. 29,
An example of a digital fader used conventionally is shown. In FIG. 29, the value of the control fader 17 is set to A
-The D converter 18 converts it to digital data, and this value is C
It is read by the PU 19 and converted into coefficient data of 1 to 0 to be represented. This converted coefficient data is the data interpolator 2
Written to 0.

When this circuit is actually operated, the CPU 1
The coefficient values to be transferred by 9 and the flow of the digital audio signal are in the state shown in FIG. 6C, and the digital audio data is naturally updated at a cycle of 1 / f _S (f _S is a sampling frequency). On the other hand, since all digital audio signals are communicated as serial signals inside the mixer, the digital audio data is No. 1, No. Two
... No. It changes like N. On the other hand, CPU
When writing fader coefficient Fdn-1 · · At 19, almost impossible is changing the coefficients for each f _S in synchronism with f _S of the digital audio signal.

This is because (1) the master clock of the CPU 19 must be synchronized with f _{S of the} digital audio signal, but in a device having a plurality of inputs and outputs such as a mixer, which input is to be synchronized with? , There is a problem of how to process when out of sync occurs, CPU
It is difficult to completely guarantee the operation of. (2) wherein (1) even if the problems are overcome if, although it is possible to rewrite the coefficients with CPU in synchronism with f _S of the digital audio signal, the rewritten every f _S is
The processing of the CPU becomes heavy. For example, one fader requires one CPU, which is not realistic.

Therefore, as shown in FIG. 6C, the fader coefficients Fdn-1, Fdn, Fdn + 1 can be changed only once every several changes of the digital audio signal. In fact, in the conventional digital mixer, the sampling frequency f _S of the audio signal is 44.1 kHz or 48 kHz, but the frequency of updating the fader coefficient is 60 Hz, and the audio data changes once every about 1000 times. The fader coefficient is designed to change. Eventually, the fader coefficient changes in a staircase shown as real data in FIG. 6B. If this coefficient is used as it is, and the digital audio data is multiplied by the multiplier 21 inside the DSP, modulation noise modulated at the coefficient transfer cycle (60 Hz) is generated, and noise such as jagged noise (zipper noise) is generated. ). So, in DSP,
Instead of using the fader coefficient transferred from the CPU 19 as it is, an interpolation is performed by the interpolator 20 and used for multiplication. Therefore, (B) of FIG.
First, a method to be solved will be described using interpolation data as shown in FIG.

As described above, modulation noise occurs when the step-like coefficient which is not interpolated in FIG. 6B is used. Therefore, in order to interpolate this coefficient inside the DSP, processing is performed with two types of time constants, 20 ms and 5 ms, as interpolation time constants. The determination of this time constant was made based on the result of the actual listening test. Explaining this test method, first, the time constant of the interpolator 20 is set sufficiently large (for example, 200 ms). Therefore, when the fader 17 is actually moved up and down, since the time constant is sufficiently large, the modulation noise does not occur, but the arrival at the target value is naturally delayed as shown by the curve 6B in FIG. 6A. .. In such a case, although no modulation noise is generated, it takes time until the sound becomes louder after the fader 17 is raised (volume up), or until the sound is heard after the fader 17 is lowered (volume down). It sounds delayed. That is, the smaller the time constant is, the faster the response becomes. However, if the time constant is reduced near the limit where the modulation noise does not occur, it is possible to realize the quick response with no noise. The time constant obtained as a result of the examination in this way is 2
It is 0 ms.

Next, referring to FIG. 29, the fader control data mute switch 22 is turned on / off to control the fader 1.
When 7 is disconnected or connected, there is no noise, but the response still feels slow. That is, the response is felt at the top and bottom of the fader 17 at a speed within the allowable range but at a slow speed when the mute switch 22 is turned on and off.
Therefore, the time constant is sufficiently reduced for the mute switch 22 until the response does not feel strange. The time constant thus obtained is 5 ms. At this time,
It seems that a change in one direction such as a switch does not cause noise particularly without interpolating the coefficient, but when the fader coefficient is instantaneously changed from "0" to a certain value or from a certain value to "0", At that point, a discontinuity occurs in the audio data, and when digital conversion of the digital audio data having this discontinuity occurs, aliasing noise occurs due to the sampling theorem, and sounds like noise as described above in the phase change. .. In this way, the same time constant is set in the interpolator 20 and the mute switch 22 is set.
If you interpolate the coefficient when turning on and off and when the fader 17 is moved, noise will occur if you interpolate the coefficient by matching the time constant to the mute switch 22, and if you interpolate the coefficient with the time constant that matches the movement of the fader 17, the response There is a problem that is slow.

Next, problems of the conventional effector of the digital mixer will be described. Conventionally, when inserting a sound effect with an effector (equalizer, filter), the effector 23 is switched on and off simply by using a switch 24 as shown in FIG. 30, but especially with digital audio, data continuity is lost. Therefore, switching noise may occur. For example, as effectors, consider an equalizer, as EQ frequency characteristic shown in FIG. 30 (B), there range (f ₀
When a near level) is boosted by a considerable level, for example, +10 dB, if the switch 24 is simply switched to the effector side as in the conventional case, if the audio signal including the sound near f _{0 is} naturally input, the switch 24 is turned on / off. Causes a difference in output level. When this is illustrated, a discontinuity point is generated in the audio data 11C by switching the switch 24 like 11D in FIG. 11B, and the continuity of the audio signal is lost due to the level difference at the time of switching the switch.

Next, the problem of the insertion 8 will be described. Generally, in an audio mixer, the internal level (headroom) of the mixer is set lower than the input level in consideration of the addition of multi-input audio signals. When output from the insertion point 25 to the outside at the internal level which is lower than this input,
The full S / N cannot be obtained because the full bit cannot be input to the external effector device. Therefore, the level is returned to the input level and the insertion point 25 (see FIG.
When output from 1), if the external effector has some gain (boosting of high and low sounds), the data will be clipped as shown in FIG. 13B. as mentioned above,
The data cannot be restored once it is clipped without lowering the internal level, but if the internal level is low, normal data can be output by lowering the level with the master fader 11. If the input signals are added as they are, they will be clipped, and even if the master fader 11 is lowered, the clipped sound will be small and clipping cannot be avoided. Therefore, it is necessary to reduce the internal level in the mixer.

As described above, the insertion point 25 (FIG. 31) is provided in front of the rear channel fader 9 of the equalizer 7 (FIG. 1), so that the internal level is smaller than the input level. This has other reasons besides preventing clipping due to addition of audio data as described above. That is, the mixer has an effector with some gain. For example, the equalizer 7 can produce up to 15 sounds in a certain band.
It may lift dB. Therefore, unless the internal level is lowered, the clip by the equalizer 7 cannot be escaped even if the channel fader 9 is lowered. As mentioned above, the optimal insertion input / output level is
It depends on the usage of the operator who uses the insertion.

Next, in the conventional mixer, when the input through the routing switcher 2 is used as the input of the mixer, when the setting of the routing switcher 2 is changed, the parameter settings of the channels CH1 to CH16 of the mixer are also changed. There must be. Therefore, conventionally, when the channel was changed, it was necessary to set the changed parameters again. That is, conventionally, 16 channels C
Equalizer 7, filter 6, fader 9 etc. are provided for H1 to CH16 respectively to perform signal processing
In the conventional snapshot automation (Japanese Patent Publication No. 2-47125, etc.), parameters such as an equalizer, a filter and a fader set for the inputs of CH1 to CH16 are read out instantly. For example, in the scene setting data shown as an example in FIG. 16, parameters such as an equalizer, a filter, and a fader are instantly changed according to these data. That is, the snapshot changes the value of each parameter with respect to the input signal that has passed through the routing switcher 2. Therefore, there is a problem that which input signal # of the audio signal is input to the channel CH1 cannot be reflected in the snapshot.

Next, the output (PG) of the above-mentioned mixer and other proposed mixers (Japanese Patent Application Publication No. 2-47125)
M) The level display device will be described. The mixer is
As shown in FIG. 17, it comprises a processor rack 26 for processing data and a control panel 27 for man-machine interface. The data of the meter 28 for displaying the volume / sound quality data level provided in the control panel 27 is the external output PGM1 to PGM4.
It is generated from the data (Fig. 1). Then, the level display on the meter 28 is performed not by displaying the PGM data as it is as meter data but by displaying in dB. As shown in the flowchart of FIG. 20, the meter 28
The meter data for displaying is generated by the processor rack 26. The mixer shown in FIG. 1 is a fully digital audio mixer, and therefore the meter data is also digital data. And generally 28 meters
Is a bar graph meter and is shown in FIG. 18 or FIG.
The 0 segment is used, and the meter data is 8 bits.

In the conventional meter display, a reference level is added to the meter data by the control panel 27 in the above flow chart, and the meter data is displayed by the meter 28.
32. Instead of sending the meter data to the processor rack 26 from the audio output data, the meter data is transmitted to the control panel 27, the bar graph meter 28 is turned on according to the meter data, and the reference level is displayed as shown in FIG. As shown in, the color of the segment 29 above the reference level is changed (for example, orange), or a reference level setting knob 30 is provided,
Another bar graph meter 31 that simultaneously displays the reference level
Was adopted. Such a method has a drawback that the meter display and the reference level display do not have a sense of unity, and in the latter case, another bar graph display is required, and the hardware becomes complicated.

Next, when using the digital mixers described so far, the need for maintenance arises and the functionality of the circuit blocks within the mixer must be checked. If there is no self-diagnosis system to check the function of such a circuit block, especially DSP that performs digital processing, in the mixer, it is necessary to input audio signals from all inputs and outputs and check audio signals from all outputs. .. Further, even if the self-diagnosis system is provided, if there is no reference display function of the DSP or integrated circuit device that constitutes the mixer, it is necessary to perform measurement using a measuring device or the like. As described above, the conventional self-diagnosis system cannot avoid the annoyance of requiring a jig tool and performing a complicated operation during maintenance.

[0027]

Although the various problems of the digital mixer have been described so far, the present invention is directed to a digital mixer in which a signal discontinuity due to a switch change at the time of phase change and a switch change at the time of volume adjustment. To reduce the noise generated by, to prevent clipping of the signal at the time of insertion in accordance with the level of the insertion out, to improve the S / N, when changing the input signal with the routing switcher, Equalizer, filter of each channel of the mixer,
To facilitate setting of fader level and assign switch parameters, to easily see the level display of mixer output data on the mixing console, and to simplify the configuration, and to facilitate quick and easy self-diagnosis during maintenance of the mixer. In particular, the present invention aims to reduce noise generated by signal discontinuity due to switching of effector functions.

[0028]

SUMMARY OF THE INVENTION The present invention comprises a plurality of input terminals to which digital audio signals are input, and N (N ≧ 1) processing means for applying a predetermined processing to the input signals.
When the signal processed by the processing means and the selecting means for selecting one of the input signals and the signal selected by the selecting means are switched, the level of the selected signal is set to a predetermined level. And a crossfader that boosts the level of the signal selected this time from 0 level to the original level with a predetermined time constant, mixes both, and outputs the mixed signal. It is characterized by

[0029]

[Embodiments] Each embodiment for solving the above problems of the digital mixer will be described below. [Phase change] When changing the phase of the conventional digital mixer and adjusting the volume and sound quality, the coefficient is instantly changed from "1" to "-1" by the switch, and as described above, the discontinuity point is generated and noise is generated. However, the present invention is characterized in that the multiplication coefficient is gradually changed from "1" to "-1" in order to prevent the generation of this noise. Therefore, according to the present invention, as shown in FIG. 2A, an interpolation including an IIR filter or the like is used to change the coefficient from “1” to “−1” or from “−1” to “1”. The feature is that the device 32 is used. Then, when the phase inversion is necessary, the output of the interpolator 32 for interpolating the coefficient data k and the audio data are multiplied by the multiplier 33 by switching the phase change switch 34 to the necessary side. And the phase change is exponentially gradually changed and interpolated and output.

In the interpolation characteristic shown in FIG. 2B, the coefficient k is exponentially gradually changed from "1" to "-1" or "-1" to "1" in the interpolator 32. The time constant is set after the phase change switch 34 is turned on until the phase is completely inverted. As a result of a viewer's test, it was found that it is appropriate to set the time constant for interpolating the coefficient k to 3 ms to 5 ms. When the phase changing function is operated using such an interpolated coefficient k, the signal waveform becomes as shown in FIG. 4, and continuity is maintained even in the phase inversion unit. The waveform shown in FIG. 4 shows the waveform of the analog output signal when the coefficient k gradually changes from “1” to “0” and then gradually changes to “−1”. It can be seen that the phase gradually decreases and becomes 0, and the phase of the phase-inverted waveform gradually increases and the phase is smoothly inverted.

FIG. 3 shows an example in which the interpolator 32 is a DSP. The interpolator 32 includes a phase change switch 35, coefficient multipliers 36, 37 and 38, and an adder 3
9, a multiplier 40, and a 1-sample delay RAM 41. This interpolator 32 converts the input coefficient data k into [0.5]
Alternatively, it is set as [-0.5], and when there is a phase change on command, the switch 35 switches to the coefficient "-0.5" side, and when there is a phase change off command, it switches to the coefficient "0.5" side. ..
This coefficient data k is multiplied by the coefficient "0.004" by the coefficient multiplier 36.
5 ”is multiplied. The coefficient data of one sample before is delayed by one sample period by the delay RAM 41, multiplied by the coefficient “0.9955” by the coefficient multiplier 37, and added by the output of the coefficient multiplier 36 and the adder 39. Then, the output data of the adder 39 and the digital audio data are multiplied by the multiplier 40, and doubled by the coefficient multiplier 38 to output a digital audio signal whose phase is exponentially changed. In this way, by interpolating the coefficient by which the audio data is multiplied when the phase is changed by the interpolator 32, the phase change can be achieved without causing discontinuity in the audio signal as shown in FIG. Although the above-mentioned interpolation characteristic is changed exponentially, it is also possible to adopt linear interpolation in which the coefficient k changes linearly.

[Volume / Sound Quality Adjustment] As described above, by interpolating the coefficient when the mute switch 22 is turned on / off and the control fader 17 (FIG. 29) is moved with the same time constant, the time constant is adjusted to the mute switch 22. Modulation noise (zipper noise) occurs when the coefficient is interpolated, and when the coefficient is interpolated with a time constant that matches the movement of the control fader 17,
There is a problem that the response becomes slow.

According to the present invention, in order to eliminate the modulation noise and speed up the response, the interpolator detects whether the mute switch is turned on or off or the control fader is moved, and the time constant suitable for each is detected. The feature is that the coefficient is interpolated to adjust the volume. FIG. 5 shows a system for adjusting the volume by interpolating the coefficient according to the present invention. First, the coefficient value set by the control fader 17 passes through the mute switch 22 and is AD-converted by the AD converter 18. When the mute switch 22 is turned on, the coefficient itself set by the control fader 17 is communicated with the AD converter 19 when the switch is turned off.
Here, the CPU 1 sends an on / off command for the mute switch 22.
9 determines and is digitally processed in the CPU 19.

The CPU 19 reads the fader coefficient data AD-converted by the AD converter 18, and writes it in the data interpolation block 42. This data interpolation block 42
Consists of two data interpolators having interpolation time constants 5 ms and 20 ms, one data interpolator 43 having a time constant of 5 ms and the other data interpolator 44 having a time constant of 20 ms, As shown in FIG. 6A, the data interpolator 43 has a short time constant (curve 6A,
5 ms) and the data interpolator 44 performs interpolation with a long time constant (curve 6B, 20 ms).

The operation of the data interpolator will be described with reference to the block diagram of FIG. 5 and the coefficient interpolation flowchart of FIG. The digitally converted data interpolation coefficient is C
The coefficient written in the PU 19 and written in the CPU 19 is written in the register 45 in the data interpolation block 42. Note that DRAM may be used for writing instead of the register. The coefficient data written in the register 45 is transferred to the delay RAM 4 when the next data is transferred.
Written in 6. Therefore, in the data interpolation block 42, the register 45 stores the current coefficient value Fdn as
The delay RAM 46 stores the coefficient value Fdn-1 of one sample before.

In such a state, the mute switch 22
Is turned on and off and the coefficient changes, the value of Fdn or Fdn-1 becomes 0, so Fdn = 0 or Fd
When n-1 = 0, the data interpolator 43 is used to interpolate the target value with a short time constant of 5 ms, and other Fdn ≠ 0.
When Fdn-1 ≠ 0, the interpolator 44 is used to perform interpolation with a long time constant of 20 ms with respect to the target value. By interpolating in this way, as shown in FIG. 6B, smooth interpolation data can be obtained when the coefficient value of the control fader 17 changes with respect to the real data actually transferred from the CPU 19. Interpolation is performed and the mute switch 22
When is turned on and off, it can interpolate with fast response. Therefore, it becomes possible to approximate the actual change in the volume to the operation of the mute switch 22 and the control fader 17.

[Switching of Effector] As described above, when the effector is simply switched by the switch 24 (FIG. 30), the continuity of digital audio data is lost and switching noise occurs. Therefore, in the present invention, not the changeover switch but the effector 23 is crossfaded as shown in FIG.
FIG. 11 shows the transition of the signal when the effectors are switched by the crossfade. When this effector is activated (corresponding to switch-on), as shown in FIG. 11 (A), a signal (11
A) gradually fades out, and the signal (11B) passing through the effector gradually fades in. The time constant of this crossfade is 3 ms to 5 ms. Further, FIG. 11B shows the actual transition of the audio signal when the crossfade is activated and when it is not activated. First, when the crossfade is not activated, there is no change in the signal, so the input signal is output as it is (waveform 11C). When the switch is simply switched as in the conventional case, the discontinuous waveform (waveform 11D) is obtained as described above. However, when crossfading is switched as in the present invention, the effector-on signal is generated as shown in FIG. 11A. Then, the effector-off signal gradually changes, so that the signal has no discontinuity (waveform 11E).

FIG. 9 shows D in the actual crossfading.
It shows an example of a block configuration diagram of processing in the SP. Figure 9
In, the fade-in and fade-out shown in FIG. 8 are processed in the DSP as follows. When the crossfade switch 47 is on, the coefficient data “0.5” is input, and when the crossfade switch 47 is off, the coefficient data “0” is input to the coefficient multiplier 48 having a multiplication coefficient of 0.0045. When the output of the effector 23 is faded in, the coefficient data “0.5” is input to the coefficient multiplier 48, the output data thereof, the 1-sample delay circuit 49, and the coefficient multiplier 50.
The signal passing through (multiplication coefficient 0.9955) is added by the adder 51.
Is added in. The fade-in coefficient data k processed in this way gradually increases exponentially. The coefficient data k and the output of the effector 23 are multiplied by the multiplier 52 and input to the adder 53 as a fade-in signal.

On the other hand, the digital audio signal is multiplied by the multiplier 54.
The coefficient data “0.5” is calculated by the adder 55 as the coefficient data k, and the fade-out coefficient data (0.5−k) is input to the multiplier 54 and is multiplied by the audio signal. , Exponentially faded out. This faded-out audio signal is also input to the adder 53 and mixed with the signal multiplied by the effector, and the mixed output is passed through the coefficient multiplier 56 (multiplication coefficient 2) to output the cross-faded audio signal. .. When the crossfade switch 47 is turned off, the coefficient data “0” is input to the coefficient multiplier 48 and k = 0.
Therefore, the effector output is not faded in, and the audio signal is multiplied by the coefficient data “0.5” in the multiplier 54, passed through the adder 53, and doubled by the coefficient multiplier 56 to be subjected to the effector processing. Is output without. Further, as shown in FIG. 10, when a plurality of effectors exist in parallel, that is, switching between a sound that does not pass through the effector and a sound that passes through the effector N, “effector 1”
Switching between a plurality of types of effectors, such as switching between “effector N” and “effector N”, can be performed without cross-fading. These switchings actually use the crossfade on / off information as the DSP.
To perform crossfade processing inside the DSP.

If the cross-faded effector is adopted in a digital mixer for video editing, it becomes possible to operate it under the control of a video editor or the like. When controlling with a video editor, it is general to store various types of patterns such as effectors in a snapshot memory in advance and call the snapshot number. Therefore, for the input 16 channels (CH1 to CH16) of the digital mixer shown in FIG. 1, fader values, effector parameters such as an equalizer and a filter, etc. are stored, and when a snapshot is called, 16 channels worth of channels are stored. The parameters of all effectors may be changed at the same time. In such a case, audio data is more likely to be discontinuous than in the case of a single effector. In addition, when manually operating the effector, the boost amount etc. are often raised and lowered, but when turning the effector on and off using snapshots, most of the time, the preset parameters are called, so the scene reproduction Sometimes it is not possible to gradually increase the effect amount. On the other hand, when the crossfade operation is performed as in the present invention, it becomes a particularly effective means at the time of snapshot.

[Insertion] As described above, the optimum input / output level of the insertion differs depending on the usage method of the operator who uses the insertion. Therefore, in the present invention, first, the insertion is performed with the digital signal as it is. This means that once converted into an analog signal, full digital processing cannot be performed, and highly accurate processing may not be possible. Further, the present invention allows the user to set the input / output level of the insertion. Even if it outputs the input level to the insertion as it is, or outputs it as the internal level (the level that has dropped the level of headroom), it causes clipping,
A characteristic is that the user can change the input / output level because a sufficient S / N cannot be obtained. For example, if the external effector 59 (FIG. 12) is an equalizer or the like having a gain, it is necessary to lower the insertion out level, and if the external effector is a limiter or the like, the external effector 59 does not have a gain. Boost to the level necessary to improve and insert out, etc.
It is possible to freely change the insertion out level at the operator's discretion. As a result, the clip is prevented from occurring and a sufficient S / N ratio is achieved.
Can be obtained. The variable range of the level performed by the operator is set from the internal equalizer input level (-30 dB) to the input level (0 dB).

FIG. 12 shows a block diagram of the present invention. When performing an insertion, the operator uses an insertion out level shifter 57 for level shifting the input signal provided in the mixer. Determine the level. The mixer also receives the insertion-in signal at the level of the insertion-out signal, for which the mixer comprises an insertion-in level shifter 58. Level shifter 5
The operator can easily perform the insertion by controlling 7 and 58 in conjunction with each other and automatically setting them. As described above, when the operator freely changes the level of the insertion out signal and lowers the insertion out signal level from the input signal as shown in (A) of FIG. 13, (B) of FIG. As shown, the insertion-in signal does not exceed the full bit level, which makes it possible to prevent clipping.

[Method of Changing Mixer Parameter] In the conventional mixer, when the input through the routing switcher 2 is used as the input of the mixer, when the setting of the routing switcher 2 is changed, channels CH1 to CH1 of the mixer are changed.
The CH16 parameter settings also had to be changed. Therefore, conventionally, when the channel was changed, it was necessary to set the changed parameters again. Therefore, the present invention is characterized in that the parameter is set for the input of the routing switcher 2 instead of setting the parameter for the channel that has passed through the routing switcher 2 of CH1 to CH16 as in the conventional case. There is. In the case of the digital mixer shown in FIG. 1, since the input of the routing switcher 2 is 32 channels and the output is 16 channels, the scene data is doubled.
The scene settings are as shown in 5. By looking at this scene data and the setting of the routing switcher 2, the CPU 19 changes the value of each parameter for CH1 to CH16.

The parameter setting of the present invention will be specifically described. As an example, it is assumed that audio input # 3 is selected for CH1 by the routing switcher 2. At that time, it is assumed that the scene data of FIG. 15 is called by the snapshot. Then, the parameter of CH1 is changed according to the data of audio input # 3. Thus, the audio inputs # 1 to # 3 selected by the routing switcher 2 for the inputs up to CH16.
The scene can be set by referring to the signals up to 2. As another example, all the audio inputs # 1 are input to the inputs of CH1 to CH16 in the routing switcher 2.
When is selected, all the parameter values are set to the values of the input audio # 1 when the scene of FIG. 15 is called at this time.

FIG. 14 shows the configuration of a digital mixer for setting parameters for the input of the routing switcher 2 of the present invention. After mixing the digital audio input signals from # 1 to #N with the routing switcher 2, the M channel digital audio signals from CH1 to CHM are passed through the effectors such as the filter 6 and the equalizer 7 to the channel fader 9 at the bus output. It is added to the mixing bus according to the assignment switch 10 through the. Scene number, depending on the operator
When the routing switcher 2 is selected, the CPU 1
9 calls the selected scene data from the scene data memory 60, and stores the routing switcher 2 and parameter settings of the number of input channels. When #x (1.ltoreq.x.ltoreq.N) is routed to the mixer input CH1, the CPU 19 causes the filter 6, the equalizer 7, and the fader 9 to set the parameter setting for #x according to the selected scene data. , Assign switch 10. In other words, the parameters can be changed according to the sources # 1 to #N, and M>
Even in the case of N, it is not necessary to change the parameter settings of the channels CH1 to CHM each time the routing switcher 2 is switched.

The parameter setting operation will be described more specifically by comparison with the conventional example. In the mixer having the routing switcher 2 at the input, snapshot automation is conventionally performed on the selected signal, but in the case of the present invention, the parameter is applied to the signal before being selected as described above. Can be changed. By doing so, for example, when an effect such as a telephone voice is desired to be applied to the audio input signal # 1, and when there is a snapshot in which such parameters can be set, this snapshot is conventionally used as CH1.
To CH16, the settings for effecting the telephone are recalled for the channel. Therefore, if the telephone scene data has a telephone effect on CH2, the audio input signal # 1 to be subjected to the effect must be selected for CH2. Therefore, as an actual operation, it is necessary to perform an operation of recalling the snapshot and then reselecting the audio input signal # 1 to the channel having the effect of the telephone. In this case, the operator needs to know which effect is applied to which channel.

On the other hand, in the case of the present invention, since the snapshot recall is performed from the audio input signal # 1 to the audio input signal # 32, for example, if the signal to which the telephone effect is desired is the audio input signal # 1. , You can recall the scene that applies the effect to audio input signal # 1. Therefore, CH1 to C
Regardless of which input of H16 the audio input signal # 1 is selected, when the scene is recalled, the telephone effect can be applied to the selected channel of the audio input signal # 1. That is, the audio input signal #
If the effects to be applied from 1 to # 32 are made in each scene, the operator only needs to recall the snapshot and does not need to reselect the input for each channel.

[Output Level Display Device of Mixing Console] Next, a display device of a mixing console suitable for the mixer and other proposed mixers (Japanese Patent Application Publication No. 2-47125) will be described. The mixer is composed of a processor rack 26 that processes data and a control panel 27 that controls a man-machine interface, as described above with reference to FIG. The data of the meter 28 that displays the volume / sound quality data of the control panel 27 is the above-mentioned four external output PGMs.
It is generated from the data of 1 to PGM4. And meter 2
The display to 8 is performed in dB display instead of displaying PGM data as it is as meter data. Figure 2
Meter data for displaying a meter is generated by the processor rack 26 as shown in the flowchart of 0. The mixer shown in FIG. 1 is a fully digital audio mixer, and therefore the meter data is also digital data. Generally, the meter is a bar graph meter of 100 segments and the meter data is 8 bits.

In the conventional meter display, the reference level is added to the meter data by the control panel 27 and the meter data is not sent to the meter portion as described in the above flow chart, but the audio output data is sent by the processor rack 26. The meter data is transmitted to the control panel 27, the bar graph meter is turned on according to the meter data, and the reference level is displayed by changing the color of the segment above the reference level or by providing a reference level setting knob. The method (FIG. 32) of providing another bar graph meter for displaying the reference level was taken. With such a method, there is no sense of unity between the meter display and the reference level display, and in the latter case, another bar graph display is required, and the hardware is complicated.

The present invention has solved the above-mentioned problems, and the method of lighting the meter data of the present invention will be described below. In describing the present invention, first, a method of creating meter data for displaying a meter will be described with reference to FIGS. 18, 19 and 21. In the segment display meter 28 shown in FIG. 18, a case where the reference level is -20 dB and -30 dB and -10 dB are transferred as audio data will be described below. It is also assumed that -20 dB is lit from the bottom of 20 segments, -30 dB is lit from the bottom of 10 segments (B in FIG. 18), and -10 dB is lit from 30 segments (C of FIG. 18) in 100 segments of the bar graph meter.

First, the 8-bit data transferred from the processor rack 26 is converted 61 into 100-bit data. That is, -30 dB audio data can be represented as 0a (hexadecimal) with 8 bits and -10 dB with 1e, and thus 100 bit data is created from this data. When expressed in 100 bits, 0a = 000 ... 000000000000000000001000000000 1e = 000 ... 100000000000000000000000000000 becomes 0a, the 10th bit from the bottom is "1" and the rest are all "0", 1e the 30th bit from the bottom is "1" Becomes This data is converted 62 from the bit which is "1" to all the lower bits to "1". That is, it is converted into 0a → 000 ... 000000000000000000001111111111 ... (F1) 1e → 000 ... 111111111111111111111111111111 ... (F2). Data for displaying the reference level is added to this. Since the reference level is -20 dB, it can be expressed as 14 (hexadecimal), so 14 = 000 ... 0000000000100000000000000000
The 20th bit from 00 and LSB becomes "1". The logical sum 63 of this data and each data after conversion is calculated.
Then, meter data (−30 dB) = 000 ... 000000000010000000001111111111 meter data (−10 dB) = 000 ... 111111111111111111111111111111 are generated. By lighting the segment of the meter with this data, lighting such that the reference level and the audio data level are simultaneously displayed as shown in FIG. 18 can be realized.

In the case of actual implementation, the lowest segment (LSB) of the meter is turned on at -∞ level, that is, even if the audio data is "0", the LSB segment is turned on (A in FIG. 18). In this case, the logical sum 63 of the reference level data "14" and the logical sum 64 of the data 67 "01" is obtained. Data such as −30 dB and −10 dB must be converted into LSB after conversion like F1 and F2.
Since it is 1, it is not necessary to logically OR "01", but the above operation is required to light the LSB 67 when the data is 0. Alternatively, only the LSB can be always turned on by hardware. Further, as another lighting example, when the reference level display data is not logically ORed 63 but exclusive ORed 65, as shown in FIG. 19, even if the audio level is higher than the reference level, the reference level (−20 dB). Can be displayed. The control panel 27 is provided with a setting device so that the user can set the reference level display data 66 to the desired data.

[Self-Diagnosis of Mixer] When using the mixer described above, maintenance is required, and the function of the circuit block in the mixer must be checked. If there is no self-diagnosis system that checks the function of such a circuit block, especially DSP that performs digital processing, in the mixer, it is necessary to input audio signals from all inputs and outputs and check audio signals from all outputs. .. Even if there is a system for self-diagnosis, if the reference of the DSP or integrated circuit device that composes the mixer is not displayed, it is necessary to sequentially measure with a measuring instrument or the like.

FIG. 22 shows the DS incorporated in the digital mixer.
FIG. 11 is a P block diagram showing an example in which 16 channels of digital audio signals are input to the routing switcher 2. In FIG. 22, eight stereo signals of digital audio signals # 1/2 to # 15/16 are first input to the routing switcher 2. The signal selected by the routing switcher 2 is AES / EBU
The output is demodulated by the demodulation ICs DI1 to DI8, and the output is DSP (Digital Signal Processor) DSP1_1.
Input to the DSP1_4. Further, the outputs of DSP1_1 to DSP1_4 become the inputs of DSP2_1 to DSP2_4. These DSPs perform signal processing such as equalizer, filter and delay. In addition, DSP2_1-DS
The output of P2-4 is input to DSP3_1 to DSP3_4, and the DSP3_1 to DSP3_4 perform processing such as fader.

The signals passed through these DSPs are finally added by the adder 66 and passed through the DSP3, and the output signals PGM1 / 2 and PGM3 / 4 of the DSP3 are input to the DSP4 and the modulators DO1 and DO2, respectively. .. Modulator D
The signals input to O1 and DO2 are modulated into AES / EBU signals and output PGM1 / 2, PGM3 / 4 externally. The signal input to the DSP 4 is converted into meter display data by the DSP 4, converted into a parallel signal by the serial / parallel converter S / P, read by the CPU 19, and displayed on the display 28 of the control panel. .. Further, the signals modulated by the modulators DO1 and DO2 are respectively returned to the input of the routing switcher 2 via a loop, and self-diagnosis is performed via this loop.

Next, the self-diagnosis method of the present invention will be described below with reference to the self-diagnosis flowcharts shown in FIGS. And in this flowchart,
Control panel 27 (Fig. 1)
Although there is a process to be displayed in 7), the conditions for judging an error at this time are: (1) DC data (5555aaaa) cannot be generated. (2) Data cannot be passed through. (3) "0" data cannot be generated. This is the case when any one of the three points is applied. Further, the condition (3) may not be checked if the condition (1) is not satisfied. Further, when the DC data of the condition (1) is generated, a multiplier inside the DSP, an ALU, a RAM
Also check functions such as. Therefore, the signal processing mode in each DSP of the self-diagnosis described in this flowchart is A: data is made to pass (input = output), B:
The MPY, ALU, RAM, etc. inside the DSP are checked, and if there is no abnormality, DC data that is not 0 is generated (for example, 5555aaa) and C: 0 is output. ing.

The flow of self-diagnosis will be described below with reference to the flowchart. [Step 1] Data by DSP4 (5555aaa)
(Processing B), all other DSPs pass data through (Processing A). Also, the routing is stereo input # 1/2 is 1, # 3/4 is 2 ...
# 15/16 is set to 8. [Step 2] S / P the data generated by the DSP 4
Convert the data and read the data with the CPU, CP the data
Compare with the U data (5555aaa). If the result of the comparison is that it was not received correctly (NO), the DSP
4 is regarded as an error and the reference number of the IC constituting the DSP 4 is output to the control panel. [Step 3] When the DSP 4 correctly receives the data (YES), the DSP 4 is subjected to processing A and D
Process B is applied to SP3 and data (5555) is sent from DSP3.
aaa) is generated. [Step 4] The same judgment as in step 2 is performed. If the data do not match, the DSP4 and DSP3 are regarded as an error and the reference number of the IC is output to the control panel. [Step 5] If the data match, DSP3
To Process A, DSP3_1 to Process B, and DSP3_
Process C is performed on 2 to DSP3_4. [Step 6] S / P conversion is performed, the data is read by the CPU, and the data is compared with 5555aaa. If the data do not match (NO), DSP3, DSP3_13
~ DSP3_4 as an error on the control panel I
The reference number of C is output. [Steps 7 to 12] Process B to DSP3_2 and another DS
Process C is applied to P3_1, DSP3_3, and DSP3_4, and the same judgment is performed. And repeat DSP3_3, DS
Process P is applied to P3_4 and the same judgment is made.
If the data do not match, DSP3, DSP3_
1 to DSP3_4 is regarded as an error and the reference number of the IC is output to the control panel.

[Step 13] DSP3_1 to DSP3
_4 for process A, DSP2_1 for process B, DSP2_
Process C is performed on 2 to DSP2_4. [Step 14] The output data is S / P converted, the data is read by the CPU, and the data is read by the CPU (55
55 aaa). If the data do not match, D
SP3_1 to DSP3_4 and DSP2_1 to DSP2
The IC reference number is output to the control panel with _4 as an error. [Steps 15 to 20] Process B to DSP2_2
Process C is applied to SP2_1, DSP2_3, and DSP2_4, and the same judgment is made. And repeat DSP2_3, D
Process B is also applied to SP2_4 and the same judgment is made. If the data do not match, DSP3_1 to D are the same as above.
The SP3_4 and DSP2_1 to DSP2_4 are regarded as an error and the reference number of the IC is output to the control panel.

[Step 21] DSP2_1 to DSP2
_4 for process A, DSP1_1 for process B, DSP1_
Process C is performed on 2 to DSP1_4. [Step 22] The output data is S / P converted, the data is read by the CPU, and the data is read by the CPU (55
55 aaa). If the data do not match, D
SP1_1 to DSP1_4 and DSP2_1 to DSP2
The IC reference number is output to the control panel with _4 as an error. [Steps 23 to 28] Process B is processed by DSP1_2
Processing C is performed on SP1_1, DSP1_3, and DSP1_4, and the same judgment is performed. And repeat DSP1_3, D
Process B is also applied to SP1_4 and the same judgment is made. If data does not match, DSP1_1 to D
The SP1_4 and DSP2_1 to DSP2_4 are regarded as an error and the reference number of the IC is output to the control panel.

[Step 29] Process B is applied to DSP5 and DSP6, Process A is applied to DSP1_1, DSP1_2-DS
Process C is applied to P1_4, and DO1 and DO2 (modulator)
The input of is switched to DSP5 and DSP6. In addition, the routing switcher selects the outputs of DO1 and DO2 for the inputs of DI1 and DI2 (demodulator). [Step 30] The output data is S / P converted, the data is read by the CPU, and the data is compared with (5555aa00). The data to be compared here is (5555aa0
The reason for setting 0) is that when passing through DI and DO, the data becomes 24 bits from the internal bus 32 bit in the ABS / EBU format. If data does not match, DSP1_1, DI1, DI2, DO1, D
The reference number of the IC is output to the control panel as an error in O2 and the routing switcher 2. [Step 31] Process C to DSP1_1, DSP1_
Process A for 2 and Process C for DSP1_3 and DSP1_4
Then, the routing switcher selects the outputs of DO1 and DO2 for the inputs of DI3 and DI4. [Step 32] The same comparison as in step 30 is performed. If the data do not match, the DSP1_2, DI3, and DI4 are regarded as an error and the reference number of the IC is output to the control panel. [Step 33] DSP1_1, DSP1_2 and DS
The process C is performed on P1_4, the process A is performed on the DSP1_3, and the outputs of DO1 and DO2 are selected as the inputs of D15 and DI6 by the routing switcher. [Step 34] The same comparison as described above is performed, and if the data do not match, the reference number of the IC is output to the control panel as an error for DSP1_3, DI5, and DI6. [Step 35] Process C in DSP1_1 to DSP1_3
Then, the DSP 1_4 is subjected to the processing A, and the routing switcher selects the outputs of DO1 and DO2 for the inputs of DI7 and DI8. [Step 36] The same comparison as described above is performed. If the data do not match, the DSP 1_4, DI7 and DI8 are regarded as an error and the reference number of the IC is output to the control panel.

[Step 37] Thus, all D
When the function check of SP, DO, DI and the routing switcher is completed and there is no error, the display indicating no error is displayed on the control panel and the self-diagnosis ends.

The self-diagnosis according to the present invention, as is apparent from the above flow chart, returns the output to the input and
A loop check is performed by repeating signal generation, through, and reception, and if there is a defective IC, it is specified and the reference number of the IC is output to the display.

[0063]

EFFECTS OF THE INVENTION The digital mixer of the present invention is capable of (1) phase change without generation of switching noise regardless of the level, phase, frequency, etc. of audio data. (2) When the fader mute switch is turned on or off, or when the fader is moved, the operational feeling and the actual volume response are close to each other. Further, since it is not necessary for the host computer to distinguish between the mute switch and the fader coefficient, the processing speed is increased and the program is simplified. (3) Noise is not generated when the effectors are switched, and a natural scene can be reproduced without any discomfort when reproduced from the snapshot memory (scene data memory). (4) Editing can be performed in full digital format without converting the audio data into analog during insertion. Further, since the operator can arbitrarily select the insertion out level, it is possible to use the effector without clipping the signal with an optimum S / N for the internal and external effectors. (5) Mixer channel parameters can be set for the audio signal source, no matter which source is input to which channel is selected by the routing switcher. Parameters can be changed simply by switching the source. (6) Since the display of the reference level of the bar graph meter can be seen at a glance, the mixing work is facilitated, and the reference level can be changed by the work, which enhances the versatility. (7) For multi-input, multi-output digital audio equipment,
Function inspection of each component (component) becomes easy, self-diagnosis can be performed quickly, and by outputting the component reference number, maintenance becomes easy. Furthermore, maintenance can be performed with external devices connected.

[Brief description of drawings]

FIG. 1 is a configuration diagram of a digital mixer.

FIG. 2 is an explanatory diagram of phase change of the present invention.

FIG. 3 is a block diagram showing an embodiment of phase change of the present invention.

FIG. 4 is a waveform diagram of an audio signal when the phase of the present invention is changed.

FIG. 5 is a block diagram showing an embodiment of fader switching according to the present invention.

FIG. 6 is a diagram illustrating an operation of switching faders.

FIG. 7 is a flowchart of fader switching according to the present invention.

FIG. 8 is an explanatory diagram of effector switching according to the present invention.

FIG. 9 is a block diagram showing an embodiment of effector switching according to the present invention.

FIG. 10 is a block diagram showing another embodiment of effector switching according to the present invention.

FIG. 11 is an explanatory diagram of an operation for switching an effector according to the present invention.

FIG. 12 is a block diagram of the insertion of the present invention.

FIG. 13 is an operation waveform diagram of insertion.

FIG. 14 is a block diagram showing an embodiment of parameter setting of the present invention.

FIG. 15 is a table showing an example of scene data setting of the present invention.

FIG. 16 is a table showing an example of conventional scene data setting.

FIG. 17 is a perspective view of a mixing console.

FIG. 18 is a diagram showing an example of a bar graph meter output level display of the present invention.

FIG. 19 is a diagram showing another example of the bar graph meter output level display of the present invention.

FIG. 20 is a flow chart of the bar graph meter output level display of the present invention.

FIG. 21 is a block diagram of an embodiment of a bar graph meter output level display of the present invention.

FIG. 22 is a block system diagram inside a mixer for performing self-diagnosis of the present invention.

FIG. 23 is a flowchart of the self-diagnosis of the present invention.

FIG. 24 is a flowchart of the self-diagnosis of the present invention.

FIG. 25 is a flowchart of the self-diagnosis of the present invention.

FIG. 26 is a flowchart of the self-diagnosis of the present invention.

FIG. 27 is a flowchart of the self-diagnosis of the present invention.

FIG. 28 is an explanatory diagram of conventional phase change.

FIG. 29 is an explanatory diagram of conventional fader switching.

FIG. 30 is an explanatory diagram of conventional effector switching.

FIG. 31 is an explanatory diagram of a conventional insertion.

FIG. 32 is an explanatory diagram of output data display of a conventional bar graph meter.

[Explanation of symbols]

 1 ・ Mixer input signal 2 ・ Routing switcher 3 ・ ・ Channel mute switch 4 ・ ・ Phase change device 5 ・ ・ Delay device 6 ・ ・ Filter 7 ・ ・ Equalizer 8 ・ ・ Insertion 9 ・ ・ Channel fader 10 ・ ・ Assignment Switch 11 ... Master fader 12 ... Mixer output signal 28 ... Bar graph meter 32, 43, 44 ... Coefficient data interpolator 57, 58 ... Level shifter 59 ... External effector

Claims (1)

[Claims] 1. A plurality of input terminals to which a digital audio signal is input, and N for applying a predetermined process to the input signal.
When the number (N ≧ 1) of processing means, the selection means for selecting one of the signals processed by the processing means and the input signal, and the signal selected by the selection means are switched. , The level of the selected signal is attenuated by a predetermined time constant, and the level of the signal selected this time is boosted from the 0 level to the original level by a predetermined time constant, and both are mixed and mixed. A digital audio signal processing device comprising a crossfader for outputting a signal.