JP5246095B2 - Acoustic signal processing device - Google Patents

Description

  The present invention relates to an acoustic signal processing apparatus capable of gradually lowering the level of a currently flowing signal in response to a trigger and then gradually raising the level.

  Conventionally, when it is desired to make an announcement during BGM playback in an acoustic signal processing apparatus, processing such as lowering the volume of the BGM and relatively raising the volume of the microphone for announcement is performed manually. In order to simplify this operation, it is conceivable to use a Ducker component provided in the acoustic signal processing apparatus. A time chart showing the operation of the Ducker component is shown in FIG. As shown in FIG. 12, the Ducker component abruptly attenuates the level of the signal that has been passed through without being attenuated until the trigger is supplied, and the condition for lowering the level for a certain period of time is not satisfied. The signal level is gradually increased at a time after ta. In this way, when an announcement is made during BGM playback using the Ducker component, a trigger is generated at the timing of the announcement, so that the volume of the BGM is drastically reduced so that the announcement can be heard clearly. become. However, since the volume of the BGM changes abruptly, the user feels uncomfortable.

  As a technique for solving this, there is a method of using scene recall provided in the acoustic signal processing apparatus. When using scene recall, use scene recall with fade time. A time chart at this time is shown in FIG. As shown in FIG. 13, at a timing when the trigger A is supplied, a specific scene (Scene A) set in advance is recalled. This scene (Scene A) has a fade time, and the level of the recalled scene (Scene A) gradually decreases. Then, at a timing when the trigger B is supplied after a predetermined time, a specific scene (Scene B) set in advance is recalled. This scene (Scene B) also has a fade time, and the recalled scene (Scene B) gradually increases in level. In this way, when an announcement is made during BGM playback using a scene recall with fade time, a trigger A is generated when the announcement is made, and a trigger B is generated when the announcement is finished, thereby making the announcement. It becomes possible to listen clearly without feeling uncomfortable.

Yamaha Corporation PM5D / PM5D-RH V2 DSP5D Instruction Manual, P.95, [online], [Search July 10, 2009], Internet <URL: http://www2.yamaha.co.jp/manual /pdf/pa/japan/mixers/pm5dv2_ja_om_g0.pdf> Yamaha Corporation DME DesignerVersion 3.0 Instruction Manual, P.399-400, [online], [Search July 10, 2009], Internet <URL: http: //www2.yamaha.co.jp/manual/pdf /pa/japan/signal/dme_designer_ja_om_v30f0.pdf>

As described above, when the Ducker component is used when an announcement is made during BGM playback in the acoustic signal processing apparatus, the BGM level sharply drops when the trigger is generated and the announcement is made, and the BGM level gradually increases. There was a problem that it could not be lowered.
Also, when using scene recall with fade time, it is necessary to set such a scene in advance, and a trigger or user operation is required to recall the required scene both when raising and lowering the level. There was a problem of becoming. Furthermore, since “scene recall” is used for a purpose other than the original one, there is a problem that the user operation becomes complicated.

  SUMMARY OF THE INVENTION An object of the present invention is to provide an acoustic signal processing apparatus that can gradually lower the level of a signal that is currently flowing only by turning on a trigger and then gradually increase the level of the signal.

  In order to achieve the above object, an acoustic signal processing device of the present invention mixes acoustic signals supplied from a plurality of input channels and outputs a mixed output, and a mixed output from the mixing bus. An acoustic signal processing apparatus having a plurality of output channels to be output, wherein a time fader module is inserted in the input channel section configured by cascading signal processing modules for performing a plurality of types of acoustic signal processing. When the time fader module detects a trigger, the acoustic signal level output from the input channel is gradually lowered at a slope corresponding to an arbitrarily set attack time, and the trigger is no longer detected. Output from the input channel with a slope according to the release time that can be set arbitrarily after a predetermined time. Are the most important features that it has to increase the acoustic signal level gradually.

  According to the present invention, by inserting a time fader module into an input channel, when an announcement is made, the sound signal level output from the input channel is gradually lowered just by entering a trigger, and then the input channel is The output acoustic signal level can be gradually increased.

It is a block diagram which shows the structure of the acoustic signal processing apparatus concerning the Example of this invention. It is a block diagram which shows the mixing process algorithm performed in the acoustic signal processing apparatus concerning this invention as an equivalent hardware constitution. It is a figure which shows the detailed structure of the input channel in the acoustic signal processing apparatus of this invention. It is a figure which shows the structure of the time fader module in the acoustic signal processing apparatus of this invention. It is a time chart which shows operation | movement of the time fader module in the acoustic signal processing apparatus of this invention. It is another time chart which shows operation | movement of the time fader module in the acoustic signal processing apparatus of this invention. It is a time chart which shows operation | movement of the other Example of the time fader module in the acoustic signal processing apparatus of this invention. It is a figure which shows the time fader UI screen which can set the parameter of the time fader module in the acoustic signal processing apparatus of this invention. It is a block diagram which shows the 1st usage example of the time fader module in the acoustic signal processing apparatus of this invention. It is a block diagram which shows the 2nd usage example of the time fader module in the acoustic signal processing apparatus of this invention. It is a block diagram which shows the 3rd usage example of the time fader module in the acoustic signal processing apparatus of this invention. It is a time chart which shows operation | movement of the conventional Ducker component. It is a time chart at the time of using the scene recall with the conventional fade time.

FIG. 1 is a block diagram showing a configuration of an acoustic signal processing apparatus according to an embodiment of the present invention.
In the acoustic signal processing device 1 shown in FIG. 1, a central processing unit (CPU) 10 executes a management program (OS: Operating System), and the overall operation of the acoustic signal processing device 1 is controlled on the OS. . The acoustic signal processing apparatus 1 includes a nonvolatile ROM (Read Only Member) 11 in which operation software such as a mixing control program executed by the CPU 10 is stored, and a RAM (Random) in which the work area of the CPU 10 and various data are stored. Access Memory) 12 is provided. The CPU 10 performs a mixing process by executing a mixing control program and applying a sound signal processing to a plurality of input sound signals by a DSP (Digital Signal Processor: Digital Signal Processor) 20. Note that by making the ROM 11 a rewritable ROM such as a flash memory, the operation software can be rewritten, and the version of the operation software can be easily upgraded. The DSP 20 adjusts and mixes the volume level and frequency characteristics of the input acoustic signal based on the set parameters under the control of the CPU 10, and controls the acoustic characteristics such as volume, pan, and effect based on the parameters. Digital signal processing is performed. The effector (EFX) 19 adds effects such as reverb, echo, chorus and the like to the mixed audio signal under the control of the CPU 10.

  The display IF 13 is a display interface that displays various screens related to mixing on the display unit 14 including a display such as a liquid crystal display. The detection IF 15 scans the operation elements 16 such as faders, knobs, and switches provided on the panel of the acoustic signal processing device 1 to detect operations on the operation elements 16, and based on the detected operation signals. The parameter value used for the acoustic signal processing can be changed. The communication IF 17 is a communication interface for communicating with an external device via the communication I / O 18, and is a network interface such as Ethernet (registered trademark). The CPU 10, ROM 11, RAM 12, display IF 13, detection IF 15, communication IF 17, EFX 19, DSP 20 exchange data and the like via the communication bus 21.

  The EFX 19 and the DSP 20 exchange data with the AD 22, DA 23, and DD 24 via the audio bus 25. The AD 22 is a plurality of analog input ports that input analog signals to the acoustic signal processing apparatus 1. The analog input signals input in the AD 22 are converted into digital signals and sent to the audio bus 25. The DA 23 is a plurality of analog output ports for outputting the mixed signal mixed from the acoustic signal processing device 1 to the outside. The digital output signal received via the audio bus 25 in the DA 23 is converted into an analog signal, Output from a speaker placed on the stage. The DD 24 is a plurality of digital input / output ports for inputting a digital signal to the acoustic signal processing apparatus 1 and outputting a digital signal mixed to the outside. The digital input signal input at the DD 24 is sent to the audio bus 25. The digital output signal received via the audio bus 25 is output to a digital recorder or the like. The digital signal sent from the AD 22 and DD 24 to the audio bus 25 is received by the DSP 20 and subjected to the above-described digital signal processing. The DA 23 or DD 24 receives the mixed digital signal sent from the DSP 20 to the audio bus 25.

Next, FIG. 2 is a block diagram showing the mixing processing algorithm executed in the acoustic signal processing apparatus 1 as an equivalent hardware configuration.
In FIG. 2, a plurality of analog signals input to a plurality of analog input ports (AD 22) are converted into digital signals and input to an input patch 30. The plurality of digital signals input to the plurality of digital input ports (DD24) are input to the input patch 30 as they are. In the input patch 30, any one of a plurality of input ports that are signal input sources is used as an N-channel (N is an integer equal to or greater than 1). .., 31-N are selectively patched (connected). Each input channel 31-1 to 31 -N has an audio signal In. From the input port patched by the input patch 30. 1, In. 2, In. 3, ..., In. N is supplied respectively.

  The input channel signals in the input channels 31-1 to 31-N in the input channel unit 31 are adjusted to M (M is an integer equal to or greater than 1) by adjusting the characteristics of the acoustic signal and controlling the delivery level by an equalizer or a compressor. ) Mixed bus 35 and L, R stereo cue bus 36. In this case, the N input channel signals output from the input channel unit 31 are selectively output to one or more of the M mixing buses 35. In the mixed bus 35, one or a plurality of input channel signals selectively input from any input channel among the N input channels are mixed in each of the M buses, and a total of M mixed outputs are output. Is done. The mixed outputs from each of the M mixing buses 35 are output channels (Output Channels) 32-1, 32-2, 32-3,. -M is output respectively. In each of the output channels 32-1 to 32-M, the characteristics of the acoustic signal such as the frequency balance are adjusted by an equalizer or a compressor, and the output channel signal Mix. 1, Mix. 2, Mix. 3, ..., Mix. M, and this M output channel signal Mix. 1-Mix. M is output to an output patch 34. In the L and R cue bus 36, there is a cue / monitor signal in which one or a plurality of input channel signals selectively inputted from any of the N input channels are mixed. The data is output to a cue / monitor unit (Cue / Monitor) 33. The cue / monitor output (Cue / monitor) output from the cue / monitor unit 33 is output to the output patch 34.

  In the output patch 34, the M output channel signal Mix. 1-Mix. M and any of the cue / monitor outputs from the cue / monitor unit 33 can be selectively patched (connected) to any of a plurality of output ports (DA23 and DD24). Each output port has an output patch. The output channel signal patched at 34 is provided. At the output port of the DA 23, the digital output channel signal is converted into an analog output signal, amplified by an amplifier, and emitted from a plurality of speakers arranged in the venue. Further, this analog output signal is supplied to an in-ear monitor worn by a musician or the like on the stage or reproduced by a stage monitor speaker placed in the vicinity thereof. A digital audio signal output from a digital output port section (DD24) having a plurality of digital output ports is supplied to a recorder, an externally connected DAT, etc., and can be digitally recorded. Also, the cue / monitor output is converted into an analog output signal at the output port of the DA 23 assigned by the output patch 34, and is output from a monitor speaker arranged in the operator room, headphones worn by the operator, etc. become able to.

  The N channel input channels 31-1 to 31-N in the input channel section 31 have the same configuration, and FIG. 3 is a circuit block diagram showing the detailed configuration of the jth input channel 31-j. As shown in FIG. 3, the input channel 31-j includes a high-pass filter (HPF) 40, an equalizer (EQ) 41, a compressor (COMP) 42, a delay (DELAY) 43, and a fader 44 connected in cascade. Has been. The HPF 40 cuts unnecessary low frequency components, and the EQ 41 adjusts the frequency characteristics of the input acoustic signal. The COMP 42 narrows the dynamic range of the input sound signal to prevent the input sound signal from being saturated, and the DELAY 43 is input so as to correct the distance between the sound source and the microphone. The sound signal is time delayed. The fader 44 is a level variable means for controlling the delivery level to the mixing bus 35, and is an electric type. An insertion point (Insert Point) 46 is provided between the EQ 41 and the COMP 42, and a time fader module (Time Fader) 45 can be inserted at the insertion point 46.

  The mixed processing algorithm shown in FIG. 2 is realized by the DSP 20 executing a microprogram. Since the audio signal processing in the input channels 31-1 to 31 -N is also realized by the DSP 20 executing the microprogram, the time fader module 45 is also realized by the DSP 20 executing the microprogram. An equivalent hardware configuration of the time fader module 45 is shown in FIG. The time fader module 45 shown in FIG. 4 includes an input terminal (Input) 45a to which an acoustic signal is input, a key-in terminal (Key In) 45b to which a key-in signal is input, and an output terminal (Output) from which the acoustic signal is output. 45c. The operation of the time fader module 45 will be described with reference to the time chart shown in FIG. The horizontal axis of the time chart is time.

  As shown in FIG. 5, an acoustic signal having a level of, for example, 0 dB is input to the input terminal (Input) 45a, and a pulsed key-in signal is input to the key-in terminal (Key In) 45b at timing t1. Suppose. The time fader module 45 compares the level of the key-in signal with a threshold (Detect Level), and detects that a trigger is entered when the level of the key-in signal exceeds the threshold. Here, since the key-in signal exceeds the threshold (Detect Level) at timing t1, it is detected that the trigger has entered the time fader module 45 at timing t1. When the trigger is entered, the time fader module 45 gradually lowers the level of the acoustic signal output from the output terminal (Output) 45c with an inclination corresponding to the attack time. When the level of the lowered acoustic signal reaches the level (Level R) set by the range parameter (Range), the level of the output acoustic signal is held at the level (Level R) from timing t2.

  The period from the timing t1 to t3 until the key-in signal falls below the threshold (Detect Level) is set to “Detecting” and the trigger is on. At timing t3, the trigger is stopped, and a holding period (Hold Time) is started from timing t3. The holding period (Hold Time) is “time for guaranteeing operation” after the key-in signal falls below the threshold (Detect Level), and ends at timing t4, for example. When the holding period (Hold Time) ends, the level of the acoustic signal output from the output terminal (Output) 45c is gradually increased with an inclination corresponding to the release time (Release Time). When the raised level reaches the original level of 0 dB, the level of the acoustic signal output from the reached timing t5 is maintained at 0 dB. As described above, the time fader module 45 has an inclination that can arbitrarily set the level of the output acoustic signal according to the attack time (attack time) when the trigger is entered, and is based on the range parameter (Range). Gradually lower to a level (Level R) that can be set arbitrarily. Then, when the hold time (Hold Time) that can be set arbitrarily has elapsed from the timing when the trigger was stopped, the slope can be set arbitrarily according to the release time (Release Time), gradually to the original level Go up to.

As described above, by appropriately setting the attack time, the holding time, and the release time set in the time fader module 45, an acoustic signal output in the case of an erroneous operation, etc. You will be able to quickly restore your level. The time chart at this time is shown in FIG.
As shown in FIG. 6, an acoustic signal with a level of, for example, 0 dB is input to the input terminal (Input) 45a, and a key-in signal having a pulse-like width is received at the timing t1 at the key-in terminal (Key In) 45b. Suppose that it is input at. The time fader module 45 compares the level of the key-in signal with a threshold (Detect Level), and detects that the trigger has entered the time fader module 45 at timing t1. When the trigger is activated, the level of the acoustic signal output from the output terminal (Output) 45c gradually decreases with a slope corresponding to the attack time, but the pulse width of the key-in signal is narrowed, The “Detecting” period ends at timing t3 ′ before the attack time immediately after t1 ends. Then, the holding period (Hold Time) is started from the timing t3 ′, but the timing t4 ′ before the end of the attack time (before the level of the output acoustic signal reaches the level (Level R)). At the end of the holding period (Hold Time). At the timing t4 ′, the level of the acoustic signal output from the output terminal (Output) 45c gradually increases with a slope corresponding to the release time (Release Time), and returns to the original level at the timing t5 ′. To come. As described above, even if the key-in signal is erroneously input at the timing t1, the level of the acoustic signal output from the output terminal (Output) 45c can be restored to the original level at the timing t5 ′ after a short time. .

Next, the operation of another embodiment in the time fader module 45 will be described with reference to the time chart shown in FIG. The horizontal axis of the time chart is time.
As shown in FIG. 7, an acoustic signal of a predetermined level (Level A) is input to the input terminal (Input) 45a, and a pulsed key-in signal is input to the key-in terminal (Key In) 45b at timing t11. Suppose that The time fader module 45 compares the level of the key-in signal with a threshold (Detect Level), and detects that the trigger has entered the time fader module 45 at timing t11. When the trigger is entered, the time fader module 45 gradually increases the level of the acoustic signal output from the output terminal (Output) 45c with an inclination corresponding to the attack time. When the raised level reaches the level set by the range parameter (Range), for example, 0 dB, the level of the acoustic signal output from the reached timing t12 is held at 0 dB.

  The key-in signal becomes lower than the threshold (Detect Level) at timing t13, and the trigger is stopped at timing t13 when the “Detecting” period ends. The holding period (Hold Time) is started from timing t13. For example, the holding period (Hold Time) ends at timing t14. When the holding period (Hold Time) ends, the level of the acoustic signal output from the output terminal (Output) 45c gradually decreases at an inclination corresponding to the release time (Release Time). When the lowered level reaches the original level (Level A), the level of the acoustic signal output from the reached timing t15 maintains the level (Level A). In the time fader module 45 of this embodiment, when a trigger is entered, the level of the sound signal to be output can be arbitrarily set with a slope that can be arbitrarily set according to the attack time (Attack Time). Gradually raise to the level set by the range parameter (Range). Then, when the hold time (Hold Time) that can be set arbitrarily has elapsed from the timing when the trigger was stopped, the slope can be set arbitrarily according to the release time (Release Time), gradually to the original level Go down. In this case, the level set by the range parameter (Range) needs to be set within a level that does not destroy the speaker when the acoustic signal from the output channel is output from the speaker.

When setting each parameter of the time fader module 45, the time fader UI screen 50 is displayed on the display unit 14 and set. FIG. 8 shows the time fader UI screen 50.
The time fader UI screen 50 shown in FIG. 8 is divided into a “Key In” area 51 on the left side of the page and a remaining “Mode” area 52. In the “Key In” area 51, a knob 51b for setting a threshold level (Detect Level), a display unit 51a for displaying the set threshold level, and a knob 51d for setting a holding time (Hold Time) are set. And a display unit 51c for displaying the time of the hold time. In the “Mode” area 52, a key 52b for setting the operation mode shown in FIG. 5, a key 52a for setting the operation mode shown in FIG. 7, a knob 52e for setting the attack time (Attack Time), and A display unit 52c for displaying the set attack time (Attack Time), a knob 52f for setting the release time (Release Time), and a display unit 52d for displaying the set release time (Release Time) And are provided. Further, a knob 52h for setting the level of the range parameter (Range) and a display unit 52g for displaying the level of the set range parameter (Range) are provided.

Next, usage examples of the time fader module are shown in FIGS.
In the first usage example shown in FIG. 9, BGM is input to the input terminal (Input) of the time fader module 60, and a tally signal is input to the key-in terminal (Key In). The sound signal output from the output terminal (Output) of the time fader module 60 and the sound signal of the announcement are mixed and output by a mixer 61. The tally signal corresponds to a “cue” signal used in a broadcasting station or the like, and various broadcasting devices are reset to a preset mode based on the tally signal. In the first usage example, the time fader module 60 operates as shown in FIG. 5 using the tally signal as a trigger signal. That is, when the tally signal is supplied, the BGM level in the mix signal output from the mixer 61 is gradually lowered, and the mixed announcement can be heard clearly. When the announcement ends (the tally signal is stopped), the level of BGM output from the mixer 61 is gradually increased so that the BGM can be heard.

  In the second usage example shown in FIG. 10, BGM is input to the input terminal (Input) of the time fader module 60, and an announcement signal is input to the key-in terminal (Key In). The sound signal output from the output terminal (Output) of the time fader module 60 and the sound signal of the announcement are mixed by a mixer 61 and output. In the second usage example, a trigger signal is created from the announcement signal, and when the announcement signal exceeds the threshold, the time fader module 60 operates as shown in FIG. That is, when the announcement signal exceeds the threshold, the BGM level in the mix signal output from the mixer 61 is gradually lowered, and the mixed announcement can be heard clearly. When the announcement is finished, the level of the announcement signal is lowered and the trigger is stopped, so that the level of BGM output from the mixer 61 is gradually raised so that BGM can be heard.

  In the third usage example shown in FIG. 11, the announcement signal is input to the input terminal (Input) of the time fader module 62, and the tally signal is input to the key-in terminal (Key In). The announcement signal and BGM output from the output terminal (Output) of the time fader module 62 are mixed by the mixer 61 and output. In the third usage example, the tally signal becomes a trigger signal and the time fader module 62 operates as shown in FIG. That is, when the tally signal is supplied, the level of the announcement signal in the mix signal output from the mixer 61 is gradually increased so that the mixed announcement can be heard clearly. When the announcement is finished (the tally signal is stopped), the level of the announcement signal output from the mixer 61 is gradually lowered so that BGM can be heard.

  In the above description, the acoustic signal processing device has been described as a mixer. However, the present invention is not limited to this, and an acoustic device that has two or more input channels and mixes and outputs the acoustic signals input to the input channels. That's fine. Further, the time fader module is not limited to be inserted between the EQ 41 and the COMP 42, and may be inserted before or after any module. Since this time fader module can be realized by a microprogram, it may be incorporated as a plug-in into an acoustic signal processing apparatus as required.

DESCRIPTION OF SYMBOLS 1 Acoustic signal processing apparatus, 10 CPU, 11 ROM, 12 RAM, 13 Display IF, 14 Display part, 15 Detection IF, 16 Operator, 17 Communication IF, 18 Communication I / O, 19 EFX, 20 DSP, 21 Communication bus , 22 AD, 23 DA, 24 DD, 25 Audio bus, 30 input patch, 31 input channel section, 31-1 to 31-N input channel, 32 output channel section, 32-1 to 32-M output channel, 33 queues / Monitor unit, 34 output patch, 35 mixing bus, 36 cue bus, 40 high pass filter, 41 equalizer, 42 compressor, 43 delay, 44 fader, 45 time fader module, 46 insertion point, 50 time fader UI screen, 51 “Key In "area, 51a display, 51b knob, 5 c Display section, 51d knob, 52 “Mode” area, 52a key, 52b key, 52c Display section, 52d display section, 52e knob, 52f knob, 52g Display section, 52h knob, 60 time fader module, 61 mixer, 62 time Fader module

Claims (3)

An acoustic signal processing apparatus having a plurality of mixing buses that output a mixed output by mixing acoustic signals supplied from a plurality of input channels, and a plurality of output channels that output the mixed output from the mixing bus,
A time fader module can be inserted between any of the signal processing modules in the input channel section configured by cascading signal processing modules for performing a plurality of types of acoustic signal processing, and the time fader module When a trigger is detected, the acoustic signal level output from the input channel is gradually lowered to the holding level with an inclination according to an arbitrarily settable attack time and holding level, and the trigger is no longer detected. An acoustic signal processing apparatus characterized in that the acoustic signal level output from the input channel is gradually increased with a release time that can be arbitrarily set after a predetermined time from the time and an inclination corresponding to the holding level. An acoustic signal processing apparatus having a plurality of mixing buses that output a mixed output by mixing acoustic signals supplied from a plurality of input channels, and a plurality of output channels that output the mixed output from the mixing bus,
A time fader module can be inserted between any of the signal processing modules in the input channel section configured by cascading signal processing modules for performing a plurality of types of acoustic signal processing, and the time fader module When a trigger is detected, the sound signal level output from the input channel is gradually increased to the hold level with an inclination according to an arbitrarily settable attack time and hold level, and the trigger is no longer detected. An acoustic signal processing apparatus, wherein the acoustic signal level output from the input channel is gradually lowered with a release time that can be arbitrarily set after a predetermined time from the time and an inclination corresponding to the holding level.   3. The acoustic signal processing device according to claim 1, wherein the trigger is detected while the level of the acoustic signal input to the time fader module exceeds a predetermined level.

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