Musical sound processing device, and musical sound processing method
The musical tone processing device addresses the challenge of generating realistic reverberation by modifying tone and noise signals to create a more natural and varied reverberation effect, overcoming the limitations of existing technologies.
Patent Information
- Application Number
- JP2023103480
- Authority / Receiving Office
- JP · JP
- Patent Type
- Applications
- Current Assignee / Owner
- Filing Date
- 2023-06-23
- Publication Date
- 2025-01-09
AI Technical Summary
Existing musical tone processing devices fail to generate suitable sound effects by accurately simulating reverberation, as they do not account for the complex fluctuations and statistical behavior of sound reflections in three-dimensional spaces, resulting in monotonous and unnatural reverberation sounds.
A musical tone processing device that modifies the frequency characteristics and time response of musical tone signals and noise signals to generate a second signal, which is then synthesized with the original tone signal, incorporating elements like envelope deformation and convolution to create a more natural reverberation effect.
The device produces a more suitable and varied reverberation sound by independently processing each musical tone signal, avoiding crackling noise and enabling a wide range of sound transformations, even with multiple instruments, thus enhancing the realism of the sound effects.
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Figure 2025003035000001_ABST
Abstract
Description
[Technical field]
[0001] The present invention relates to a musical sound processing device and a musical sound processing method. [Background technology]
[0002] Conventionally, there are electronic sound processors that create reverberation effects using simulated impulse responses. Conventionally, there is a Schroeder type of technology for providing reverb effects (see, for example, Non-Patent Document 1). [Prior art documents] [Non-patent literature]
[0003] [Non-Patent Document 1] Natural Sounding Artificial Reverberation, MR SCHROEDR, Bell Telephone Laboratories, Incorporated, Murray Hill, New Jersey, JOURNAL OF THE AUDIO ENGINEERING SOCIETY, JULY 1962, VOLUME 10, NUMBER 3. Summary of the Invention [Problem to be solved by the invention]
[0004] An object of the present invention is to provide a musical sound processing device and a musical sound processing method capable of generating a suitable sound effect signal for a musical sound signal. [Means for solving the problem]
[0005] One embodiment of the present invention includes a first generation unit that generates a plurality of first signals by modifying at least one of the frequency characteristics and the time response of each of a plurality of processing unit signals obtained from a musical tone signal; a second generation unit that generates a second signal in which at least one of a frequency characteristic and a time response of a noise signal associated with one or more of the signals of the plurality of processing units is replaced with at least one of a frequency characteristic and a time response of the associated first signal; A musical sound processing device comprising:
[0006] In one embodiment of the present invention, an information processing device generates a plurality of first signals by modifying at least one of the frequency characteristics and the time response of each of a plurality of processing unit signals obtained from a musical tone signal. and generating a second signal in which at least one of the frequency characteristics and the time response of a noise signal associated with one or more first signals among the plurality of first signals is replaced with at least one of the frequency characteristics and the time response of the one or more first signals. [Brief description of the drawings]
[0007] [Figure 1] FIG. 1 shows an example of the configuration of a musical sound processing device. [Diagram 2] FIG. 2 is an explanatory diagram of a musical sound processing device. [Diagram 3] FIG. 3 is a flowchart showing an example of processing performed by the musical sound processing apparatus shown in FIG. [Figure 4] FIG. 4 is an explanatory diagram showing a first configuration example. [Diagram 5] FIG. 5 is a diagram showing a first circuit configuration example applicable to the first configuration example. [Figure 6] FIG. 6 is a diagram showing a second circuit configuration example applicable to the first configuration example. [Figure 7] FIG. 7 is a flowchart showing an example of processing in the first configuration example. [Figure 8] FIG. 8 is a diagram showing the correspondence relationship between a signal of a processing unit and a noise signal. [Figure 9] FIG. 9 is a diagram showing modified examples of the envelope. [Figure 10] FIG. 10 is a diagram showing modified examples of the envelope. [Figure 11] FIG. 11A is an explanatory diagram of an ADSR, and FIG. 11B shows an example of waveform control parameters. [Figure 12] FIG. 12 is a diagram showing an example of the configuration of a musical sound processing device having a convolution section. [Figure 13] FIG. 13 is an explanatory diagram of the convolution unit. [Figure 14] FIG. 14 is an explanatory diagram of the convolution unit. [Figure 15] FIG. 15 is an explanatory diagram of the convolution unit. [Figure 16] FIG. 16 is an explanatory diagram of the convolution unit. [Figure 17] FIG. 17 is a diagram illustrating an example of the operation of the folding unit. [Figure 18] FIG. 18 is an explanatory diagram showing a second configuration example. [Figure 19] FIG. 19 is a diagram showing an example of a circuit configuration applicable to the second example configuration. [Figure 20] FIG. 20 is a flowchart showing an example of processing in the second configuration example. DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS
[0008] A musical sound processing device and a musical sound processing method according to the embodiments will be described below with reference to the drawings. The configurations of the embodiments are merely examples and are not limited to the configurations of the embodiments. FIG. 1 is a diagram showing an example of the circuit configuration of a musical sound processing device (an information processing device that operates as a musical sound processing device) according to the embodiment. The musical sound processing device is, for example, a device called an effector that generates sound effects from musical sounds. However, the configuration according to the embodiment may be implemented in a device other than an effector.
[0009] 1, a musical sound processing device 10 has a CPU (Central Processing Unit) 11 that controls the overall operation of the musical sound processing device 10. The CPU 11 is connected to a ROM (Read Only Memory) 12, a RAM (Random Access Memory) 13, and a DSP (Digital Signal Processor) 15 via a bus 2. The CPU 11 includes a user interface (UI) 16 is connected.
[0010] The RAM 13 is used as a working area for the CPU 11 and a storage area for programs and data. The ROM 12 is used as a storage area for programs and data. The RAM 13 and the ROM 12 are examples of a storage device (storage medium). The CPU 11 is an example of a processor.
[0011] The musical sound processing device 10 has an input terminal 21 to which musical sounds are input. A musical sound signal is input to the terminal 21. The musical sound signal includes the sound of a musical instrument. The musical sound may include human voice.
[0012] The musical sound signal input to the input terminal 21 is converted from analog to digital by the ADC (analog-digital converter) 17 and input to the DSP 15. A sound effect signal is generated from the digital musical sound signal input from the ADC 17, and a signal in which the musical sound signal and the sound effect signal are added is output. The musical sound signal output from the DSP 15 is input to a DAC (digital-analog converter) 18 and converted to an analog signal, which is then input to a power amplifier 19. The power amplifier 19 amplifies the input signal and connects it to a speaker 20. The speaker 20 outputs the musical sound signal with the sound effect added.
[0013] The UI 16 includes input devices such as dials, switches, buttons, and keys for inputting or setting multiple setting values (parameters) for the musical sound processing device 10, and displays (lamps, LEDs, displays, etc.) for displaying the status and setting contents of the musical sound processing device 10.
[0014] The CPU 11 executes a program stored in a storage device or the like to store parameters related to the creation of sound effects set using the UI 16 in the storage device and to set the parameters in the DSP 15. The DSP 15 generates sound effect signals according to the parameters set by the CPU 11 by executing the program.
[0015] 2 is an explanatory diagram of the musical sound processing device 10. By executing a program, the DSP 15 operates as a device including a conversion section 15A, a first generation section 15B, a second generation section 15C, a synthesis and addition section 15D, and a noise signal generation section 15E. By executing a program, the CPU 11 operates as a control section (controller) 11A that controls the operation of the DSP 15 (15A to 15E).
[0016] The conversion unit 15A generates a plurality of divided signals by dividing the input musical tone signal into a predetermined number of processing unit signals. The first generation unit 15B generates a plurality of first signals by modifying at least one of the frequency characteristics and the time response of each of the plurality of processing unit signals input from the conversion unit 15A. Generate.
[0017] The second generating unit 15C receives a plurality of first signals (for each processing unit) and a noise signal. The noise signal is generated by the noise signal generating unit 15E. In this specification, the noise signal includes a signal whose waveform is constantly changing. The noise signal includes a white noise signal (white noise) and a colored noise signal (pink noise, blue noise, etc.). The noise signal may also include a signal having characteristics of both a white noise signal and a colored noise signal. The noise signal may also include a band noise signal. The noise signal may also include a signal including at least one of a stationary irregular random signal and a random signal (random number) based on a pseudorandom number sequence. The noise signal may also include a signal whose waveform on the time axis changes over time but whose ratio of each frequency component is constant. The noise signal may also include a non-attenuating signal that is semi-permanently sustainable, or a signal that always has constant power.
[0018] The second generating unit 15C generates a second signal in which at least one of the frequency characteristics and the time response of a noise signal associated with one or more first signals among the multiple first signals is replaced with at least one of the frequency characteristics and the time response of the one or more first signals.
[0019] The synthesis and addition unit 15D outputs a signal obtained by adding a musical tone signal to a signal obtained by synthesizing a plurality of second signals generated by the second generation unit 15C. The synthesis and addition unit 15D may be composed of a synthesis unit that synthesizes the second signals and an addition unit that adds the musical tone signal to the signal generated by the synthesis unit. Note that the synthesis unit and the addition unit are optional and are not necessarily required components.
[0020] The noise signal generating unit 15E can generate a plurality of noise signals that can be associated with a plurality of processing unit signals (first signals).
[0021] Fig. 3 is a flowchart showing an example of processing by the musical sound processing device shown in Fig. 2. In step S1, the conversion unit 15A converts a musical sound signal into a plurality of processing unit signals. In step S2, the first generation unit 15B generates a first signal by modifying the frequency characteristics and time response of the processing unit signals.
[0022] In step S3, the second generating unit 15C generates a second signal by replacing the frequency characteristics and time response of one or more noise signals corresponding to the signal of the processing unit with the frequency characteristics and time response of the corresponding first signal.
[0023] In step S4, the synthesis and addition unit 15D synthesizes the second signal, adds the musical sound signal to the synthesized signal (sound effect signal), and outputs the result (step S5). As a result, the musical sound signal and the sound effect are emitted.
[0024] FIG. 4 is an explanatory diagram showing a first configuration example. In the following description, the musical sound signal may be called "signal A" and the noise signal may be called "signal S". In the first configuration example, the conversion unit 15A has a plurality of band pass filters (BPFs). The conversion unit 15A uses the plurality of BPFs to perform band division on the input musical sound signal, and converts (separates) the musical sound signal into a plurality of band signals (a plurality of divided signals). The plurality of divided signals correspond to a plurality of processing unit signals.
[0025] The first generating unit 15B detects the amplitude envelope of each of the multiple band signals and performs processing to modify the envelope. The envelope has at least one of the frequency characteristics and the time response of the signal. For example, the first generating unit 15B includes an envelope generator and can control the attack (rise), decay (decay), sustain (sustain after decay), and release (reverberation) of the envelope. For example, when generating reverberation (reverberation) as a sound effect, the first generating unit 15B lengthens the release (reverberation) time of the envelope. The signal that has undergone such envelope detection and modification processing becomes the first signal. The envelope modification processing does not necessarily need to be performed on all the divided signals, but is performed on one or more bands that require control.
[0026] The noise signal generating unit 15E has multiple BPFs. A single noise signal input to the noise signal generating unit 15E is converted into divided noise signals of multiple bands. The divided bandwidth and the number of bands may be the same as or different from the number of bands and bandwidth for the musical sound signal.
[0027] The second generating unit 15C includes a plurality of voltage controlled amplifiers (VCAs). ) A noise division signal is input to each VCA as an input signal. A first signal having a corresponding relationship with the noise division signal is also input to each VCA as a control signal. This causes the gain (magnitude) of the amplification of the noise division signal to be controlled by the magnitude of the first signal. That is, at least one of the frequency characteristics or the time response of the noise division signal is replaced with at least one of the frequency characteristics or the time response of the corresponding first signal.
[0028] The output signals from each VCA are mixed in the mixer and adder 15D. The mixer and adder 15D adds the original musical tone signal to the mixed signal and outputs the result. As a result, the sound with reverberation (sound effect) added to the musical tone signal is emitted from the speaker 20.
[0029] Fig. 5 is a diagram showing a first circuit configuration example applicable to the first configuration example. Fig. 5 shows a circuit configuration example of a musical sound processing device that performs the operation shown in Fig. 4 on left and right musical sound signals. In Fig. 5, the circuit has a series for the left musical sound signal and a series for the right musical sound signal. Both have the same configuration.
[0030] The "Filter Bank A Left" and "Filter Bank A Right" correspond to the left and right conversion units 15A, respectively, and convert the input musical tone signal into the above-mentioned multiple divided signals using multiple BPFs. The "right" corresponds to the left and right first generating units 15B, which detect the envelope of each of the input divided signals and perform necessary envelope transformation to generate a plurality of first signals.
[0031] "Correlation setting", "filter bank S left", and "filter bank S right" correspond to the left and right noise signal generation units 15E. "Correlation setting" sets the correlation (difference) between two noise signals (noise signal 1, noise signal 2). Noise signal 1 for which correlation has been set by "correlation setting" is input to "filter bank S left", and noise signal 2 is input to "filter bank S right". "Filter bank S left" and "filter bank S right" each convert the input noise signal into a predetermined number of noise division signals.
[0032] "Multiplication Left" and "Multiplication Right" correspond to the left and right second generation units 15C, and the corresponding filters It operates as a VCA for each noise-divided signal input from the filter bank. The "left synthesis process" and "right synthesis process" correspond to the left and right synthesis sections, and generate a signal that synthesizes multiple second signals input to them. The left and right adder sections add the synthesis signal output from the corresponding synthesis section to the original musical sound signal (direct sound) and output it. As a result, the left and right musical sound signals and sound effects (e.g. reverberation) are emitted from the left and right speakers.
[0033] FIG. 6 is a diagram showing a second circuit configuration example applicable to the first configuration example. In the first circuit configuration example, a plurality of noise division signals are obtained by dividing one noise signal each on the left and right. In contrast, in the second circuit configuration example, a plurality of noise signals (noise signal 11, noise signal 21, noise signal 31, etc.) corresponding to the noise division signal on the left side are prepared, and a plurality of noise signals (noise signal 12, noise signal 22, noise signal 32, etc.) corresponding to the noise division signal on the right side are prepared. After correlation is set, the plurality of noise signals are input to "multiplication left" or "multiplication right" as input signals to the VCA. Except for this point, the second circuit configuration example is the same as the first circuit configuration example. Note that the number of each of the input terminals and output terminals may be two or more.
[0034] The circuits shown in Figs. 5 and 6 may be actual circuits (hardware), but in this embodiment, the DSP 15 operates as the circuits shown in Figs. 5 or 6 according to a program. In addition, the above-mentioned VCA may be anything other than a VCA as long as it can control the gain according to a control signal. For example, a gain control device or DSP control can be applied. In the circuit configuration examples shown in Figs. 5 and 6, the musical sound signal may be a musical sound signal input from the terminal 21 or a musical sound signal reproduced within the musical sound processing device 10. Also, a musical sound signal previously divided into bands may be reproduced. In this way, band division is not necessarily a required requirement. Also, the noise signal may be generated inside the musical sound processing device 10 using a noise generator or the like, or may be input from outside the musical sound processing device 10 using an input terminal of the musical sound processing device 10.
[0035] 7 is a flowchart showing an example of processing in the first configuration example. In step S11, each of the "filter bank A left" and the "filter bank A right" performs conversion to a plurality of processing unit signals (separation into a plurality of divided signals) by band separation.
[0036] In step S12, "Envelope extraction processing, transformation processing left" and "Envelope extraction processing, transformation processing left" are performed. Each of the "right" and "left" bands detects the envelope of the divided signal of each band, and obtains a first signal by modifying the envelope (for example by lengthening the release time).
[0037] In step S13, each of "Multiply Left" and "Multiply Right" generates a second signal by amplitude-controlling a noise signal with the envelope of the corresponding first signal. In step S15, the adder adds the synthesized signal and the musical tone signal and outputs the result.
[0038] FIG. 8 is a diagram showing the correspondence relationship between a signal of a processing unit and a noise signal. For example, a musical sound signal (signal A) and a noise signal (signal S) may correspond to the same band, and the envelopes of the same band may be replaced (FIG. 8A). The band of signal A may correspond to a different band of signal S (FIG. 8B). Furthermore, signal A and signal S may correspond 1:1, or the band of the musical sound signal and the band of the noise signal may correspond 1:n (n is a natural number of 2 or more) as shown in FIG. 8C. Furthermore, the band of the musical sound signal and the band of the noise signal may correspond n:n (n is a natural number of 2 or more) as shown in FIG. 8D. In other words, the envelope of one or more noise signals can be controlled using the envelope of one or more first signals.
[0039] When controlling multiple bands of a noise signal, the same band as the musical signal and one octave above are used. By controlling the frequency band one octave lower, an effect called shimmer reverb can be created. By controlling the same frequency band as the musical signal and the two adjacent frequency bands at the same time, sidebands can be increased. The envelope of the noise signal can also be controlled by using a band division set of the musical signal that is completely different from the band division set of the noise signal.
[0040] FIG. 9 is a diagram showing a modified example of the envelope. When the envelope at the top left is the original envelope of the musical tone signal, a delayed envelope can be generated as shown on the right side. At this time, by subtracting the delayed envelope multiplied by a coefficient from the original envelope, a gated reverb-like envelope (sound effect) can be obtained. That is, by utilizing the fact that the attenuation characteristics of the release are similar regardless of which part is cut out, the envelope of the detected musical tone signal is delayed, multiplied by a coefficient, and subtracted, the value of the envelope can be set to 0 after the delay time. By changing the settings depending on the band, it is possible to obtain reverberation sounds that have not been obtained before. Conventional gated reverbs are operated by detecting the input threshold value, so they can only be applied to a single musical instrument sound. In contrast, in the method according to the embodiment, even if multiple musical instrument sounds are input, they operate individually for each processing unit, so a natural effect can be obtained. In addition, by shortening the gate time so that it cannot be perceived as a reverberation sound, a tone that has not been obtained before can be obtained.
[0041] Fig. 10 is a diagram showing a modified example of the envelope. As shown in Fig. 10A, when the attenuation coefficient is set to 1 (freeze on) when detecting the envelope of the input signal (musical sound signal), the reverberation sound at that time can be frozen (the level at that time can be maintained until the freeze is turned off).
[0042] Also, as shown in FIG. 10B, if you start counting the time during decay and set the decay coefficient to 1 until the counted value reaches the count value indicating the set time, an automatic freezing effect can be obtained.
[0043] Furthermore, if the damping coefficient during the freeze operation is set slightly larger than 1, a reverberation sound with a reverse rotation wind can be obtained (see FIG. 10(C)). If the damping coefficient during the freeze operation is set smaller than 1, a gated reverb-like effect can be obtained (see FIG. 10(D)). Furthermore, by controlling the damping coefficient, it is possible to control the damping in two stages, for example, so that the envelope damps at a predetermined speed (slope) from the start of counting until a predetermined count value is reached, and then the damping becomes slower (or faster) when the predetermined count value is reached.
[0044] Combining envelope deformation and freezing operations makes it possible to create time changes like those of ADSR. Figure 11A is an explanatory diagram of ADSR, and Figure 11B shows an example of waveform control parameters. As shown in Figure 11A, Attack is a parameter that sets the time it takes to reach that amplitude when the input amplitude increases. If it is set to 0 seconds, the input amplitude will immediately follow. Decay is a parameter that sets the time it takes to move from the amplitude reached in Attack to Sustain. Sustain (maintenance after decay) is the time for which the level is maintained. Release is a parameter that sets the time from when Sustain ends until the sound stops.
[0045] As shown in the table of Fig. 11B, the parameters set by the control unit 11A (CPU 11) to the DSP 15 include, when the effect is reverberation, character selection that determines the characteristics of the reverb tone, reverb length, attack speed, frequency at which the reverb starts to attenuate high frequencies, etc. The parameters can also include parameters related to gated reverb, such as a switch to enable / disable gated reverb. Such parameters are set in advance before the input of the musical tone signal.
[0046] FIG. 12 shows a convolution unit for setting the correspondence between the signal of the processing unit and the noise signal. 12 is a diagram showing an example of the configuration of a musical sound processing device for generating a first generation signal. In the example shown in FIG. 12, a conversion unit 15A of the musical sound processing device includes a plurality of BPFs and a convolution unit 15F. The convolution unit 15F includes an enveloper that detects the envelope of the amplitude of the musical tone signal and performs the necessary transformation. However, the convolution unit 15F may be a part of the first generation unit 15B, and the transformation It may be configured independent of the converting unit 15A and the first generating unit 15B.
[0047] As shown in Fig. 13, the convolution unit 15F includes a plurality of input terminals and a plurality of output terminals, and the plurality of input terminals and the plurality of output terminals are fully connected (all elements are convoluted). In Fig. 13, the illustration of the multipliers is omitted. By setting the coefficient (weight) of an unused tap among the plurality of taps extending from the input terminal to 0, it is possible to achieve a state in which each input terminal and each output terminal are connected 1:1, as shown in Fig. 14, that is, a state in which no convolution is performed (effectively, the convolution unit 15 does not exist).
[0048] Also, by setting the coefficient for the tap to 0, the band can be shifted to the desired output end, as shown in Fig. 15. This type of shift configuration can also be applied to octave shifts (changing the octave position up or down).
[0049] In addition, the input terminals at both ends have two taps, and each input terminal between the two ends is combined with three taps (the corresponding output terminal and the taps on either side of it) for convolution, thereby enabling peak emphasis.
[0050] The graph on the left side of Fig. 17 shows each band (BPF) and the power of each band. The envelope curves of each band and its adjacent bands are correlated (-0.5, 2.0,-0.5) and convolution was performed. The graph on the right shows the result of the convolution. The convolution emphasizes the peaks, meaning that the output result has more definition. Note that in the example shown in Figure 17, the results from one side were used for both ends of the band (numbers 1 and 20).
[0051] In this way, by providing the convolution section 15F, it becomes possible to create a tone color, etc. Also, by a simple operation of changing the coefficients, it is possible to eliminate the convolution section 15F.
[0052] 18 is an explanatory diagram showing a second configuration example of the musical sound processing device 10. In the second configuration example, an input signal (musical sound signal) is converted into a time-varying frequency domain signal by short-time Fourier transform. That is, the signal is divided into frames according to a predetermined window size by window function processing, and a Fourier transform (FFT, time-frequency transform) is performed on each frame, and each frame is converted from a time domain signal to a frequency domain signal. In this way, a plurality of frequency domain signals that are time-varying are obtained.
[0053] If we arrange the divided frames in order, we can observe the time evolution of each spectrum. And The amplitude of each spectrum (band) is detected for each frame, and the necessary modification is applied to the time change of the amplitude of each detected spectrum. The amplitude of each spectrum (band) is given to white noise (an example of a noise signal). As a result, white noise having a modified amplitude of each spectrum is generated. In other words, the source of the sound is white noise, and a spectrum is generated whose amplitude corresponds to the amplitude modified for the musical tone signal. It is not necessarily required that the amplitude of all spectra is modified. It is sufficient that the amplitude of one or more desired spectra from among a plurality of spectra is modified.
[0054] Any white noise with such a spectrum can be transformed into a time domain signal by the inverse short-time Fourier transform (IFFT). Then, a time domain signal is generated in units of frames. These time domain signals are synthesized according to the output window function. The original input signal (musical tone signal) is then added to the synthesized signal, and the result is output as an output signal.
[0055] FIG. 19 is a diagram showing an example of a circuit configuration applicable to the second configuration example. In FIG. 19, "frame separation / window processing" (frame separation unit 151) and "FFT" (FFT unit 152) are , corresponds to the conversion unit 15A. Also, the "Envelope detection / envelope conversion of each frame amplitude" shown in FIG. "Transformation" (envelope detection unit 153) corresponds to the first generation unit 15B. "Providing real and imaginary values of each frequency bin as a noise signal (e.g., white noise)" (processing unit 154) and "IFFT" (IFFT unit 155) correspond to the second generation unit 15C. The "frame synthesis" (synthesizing unit 156) and adder 157 correspond to the synthesis and addition unit 15D.
[0056] 20 is a flowchart showing an example of processing in the second configuration example. In step S21, the frame separator 151 converts a musical sound signal, which is an input signal, into a frame signal by using a window function.
[0057] In step S22, the FFT unit 152 performs an FFT on each frame and converts it into a plurality of processing units. In step S23, the envelope detection unit 153 detects the amplitude envelope of each frame and generates a first signal by modifying the envelope (for example, by lengthening the release).
[0058] In step S24, the processing unit 154 generates a second signal by replacing the amplitude of the frequency bin of the noise signal (e.g., white noise) with the amplitude of the corresponding first signal. In step S25, the IFFT unit 155 performs IFFT on each second signal. In step S26, the frame synthesis unit 156 synthesizes the signal obtained by IFFT using an output window (window function). In step S27, the adder 157 adds the synthesized signal and the original musical sound signal and outputs the result.
[0059] In the above embodiment, musical sounds are used, but natural sounds may be used. In the embodiment, bandpass filter banks and FFT are shown as the conversion unit, but other methods may be used. For example, "wavelet analysis" or "mode analysis" may be used. In addition, the first generation unit generates all of the amplitude ADSRs, but it may also be possible to generate any one of A, D, S, and R. It is also possible to combine multiple ADSRs to create a complex waveform outline. It may also be for the purpose of generating shapes.
[0060] According to the musical sound processing device 10 according to the embodiment, in either the first or second configuration example, a second signal is generated by replacing the envelope of the noise signal with the envelope of the modified musical sound signal. Then, a signal obtained by synthesizing the second signal is added to the original musical sound signal and output. In this way, a desired sound effect is obtained.
[0061] As an example of sound effects, it is possible to generate suitable reverberation of a musical sound signal. Actual reverberation is the result of reflections from walls and other surfaces overlapping with subtle air fluctuations (sound speed modulation) in three-dimensional space, resulting in sidebands with statistical behavior over time. Existing methods for generating reverberation ignore the above-mentioned fluctuation elements or reproduce reverberation using only the characteristics of a single instant. For this reason, even if additional fluctuations are added to the reproduced reverberation, only monotonous sidebands are obtained.
[0062] According to the musical sound processing device 10, in the configuration described above, a signal is output that is a synthesis of a second signal in which the envelope of the musical sound signal with a longer release (reverberation) is replaced with the envelope of a noise signal, so that a more suitable reverberation sound can be output than in the past. The signal may be output alone (separate from the musical tone signal).
[0063] In principle, it does not generate the so-called crackling noise that is often a problem with existing reverberation sounds. By independently setting the level of each band, and the attack time and release time when detecting the envelope of the input signal, it is possible to create a wide range of sounds and easily modify them. This makes it easier to realize sounds that were previously difficult to create.
[0064] In the existing method, the same input signal always produces the same output signal. In contrast, in the musical sound processing device 10 according to the embodiment, the waveform of the noise signal is constantly changing, so that even if the same input signal is input, the output signal will be different according to the change in the noise signal. The configurations of the embodiments can be appropriately combined within the scope of the invention. [Explanation of symbols]
[0065] 11···CPU 12···ROM 13···RAM 15···DSP
Claims
1. a first generating unit for generating a plurality of first signals by modifying at least one of the frequency characteristics and the time response of each of a plurality of processing unit signals obtained from a musical tone signal; a second generating unit configured to generate a second signal in which at least one of a frequency characteristic and a time response of a noise signal associated with one or more first signals among the plurality of first signals is replaced with at least one of a frequency characteristic and a time response of the one or more first signals; 23. A musical sound processing device comprising:
2. The musical tone processing apparatus according to claim 1 , further comprising a synthesis section for synthesizing the plurality of second signals generated by said second generation section.
3. The signal synthesis unit further includes an adder unit that outputs a signal obtained by adding the musical tone signal to the signal synthesized by the synthesizer unit.
3. The tone processor according to claim 2.
4. The noise signal is one or more selected from a white noise signal, a colored noise signal, a signal having characteristics of both a white noise signal and a colored noise signal, and a band noise signal.
2. The tone processor according to claim 1.
5. The noise signal includes at least one of a stationary irregular random signal and a random signal based on a pseudorandom number sequence.
2. The tone processor according to claim 1.
6. The noise signal includes a signal whose waveform on the time axis changes over time but whose ratio of each frequency component is constant.
2. The tone processor according to claim 1.
7. The noise signal includes a non-attenuating signal that is semi-permanently sustainable.
2. The tone processor according to claim 1.
8. the plurality of processing unit signals are a plurality of divided signals obtained by dividing the musical tone signal into bands, The noise signal is a plurality of noise signals that can be associated with one or more of the plurality of divided signals.
2. The tone processor according to claim 1.
9. the plurality of processing unit signals are a plurality of frequency domain signals obtained by dividing the musical tone signal according to a window size and performing a time-frequency transform of the divided signals; The noise signal is a plurality of noise signals that can be associated with one or more of the signals in the plurality of frequency domains.
2. The tone processor according to claim 1.
10. The first generating unit transforms an envelope of each of the signals of the plurality of processing units.
2. The tone processor according to claim 1.
11. The first generating unit extends a release time of an envelope of a signal of a processing unit. The tone processor according to claim 10.
12. The first generating unit performs a convolution process on the signals of the plurality of processing units.
2. The tone processor according to claim 1.
13. An information processing device, generating a plurality of first signals by modifying at least one of the frequency characteristics and the time response of each of a plurality of processing unit signals obtained from a musical tone signal; generating a second signal in which at least one of a frequency characteristic and a time response of a noise signal associated with one or more first signals among the plurality of first signals is replaced with at least one of a frequency characteristic and a time response of the one or more first signals; A musical sound processing method.
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