JPH11133996A - Musical interval converter - Google Patents

Abstract

PROBLEM TO BE SOLVED: To obtain good musical interval conversion even if discontinuity is encountered between frames. SOLUTION: An acoustic signal is supplied to an A/D converting circuit 1 to store the acoustic signal as a digital acoustic signal f(t) on a memory circuit 2. A time window 3 slices L sample to extract an acoustic signal fN(t). The acoustic signal fN(t) is supplied to a fast Fourier transform(FFT) circuit 4 for FFT. The acoustic signal fN(t) is separated into an amplitude component, a real part, and a phase component, an imaginary part. The amplitude component and the phase component are supplied to an amplitude processing circuit 5 and a phase processing circuit 6, respectively. Where, the amplitude processing circuit 5 shifts the array number k0 by pitch shift amount to convert the array number k0 into the sample of the array number k1 . Similarly, the phase processing circuit 6 converts the array number k0 having the phase θN into the sample of the array number k1 . These signals are supplied to an inversion fast Foureir transform(IFFT) circuit 7 for IFFT to obtain an acoustic signal gN(t). In addition, the acoustic signal gN(t) is supplied to the time window 8 to extract only the first ΔL sample. The first ΔL sample as part of the acoustic signal g(t) is supplied to a D/A converting circuit 9 to convert the first ΔL sample into an analog acoustic signal.

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a pitch converter for converting a sound pitch (pitch frequency, fundamental frequency) used in a karaoke apparatus, an audio / video editing apparatus or the like.

[0002]

2. Description of the Related Art Hitherto, a karaoke apparatus or the like has a function called a key control in which the pitch of an accompaniment to be played can be freely changed and set in order to match the range of a singer. In this method, the pitch is changed by changing the reproduction speed of an analog sound signal reproduced as an accompaniment. In recent years, communication data for storing music data at a center, transmitting music data as needed to remote terminal devices connected to a plurality of the center, and reproducing the music at the terminal device. Karaoke is being developed.

[0003] Song data transmitted from the communication karaoke center to the terminal device includes character data for displaying lyrics along with the song and changing the display color and a terminal for reproducing accompaniment of the song. It is composed of a MIDI signal for operating a synthesizer of the apparatus and a compressed audio signal for reproducing a real voice back chorus by a male or female voice on a terminal apparatus. In the communication karaoke terminal device, when changing the pitch of the accompaniment to be played, the pitch of the synthesizer reproduced by the MIDI signal is changed to
By setting so as to raise (lower) as a whole, the pitch can be freely changed and reproduced without changing the reproduction speed.

However, the real voice back chorus is based on MID.
Since it is not an I signal, it does not have pitch related data, does not change the playback speed, does not degrade the sound quality, and it is difficult to convert the sound pitch while retaining the characteristics of the personal voice. there were. In recent years, audio-video editing apparatuses that perform editing operations in the state of digital signals have been developed, but it has been difficult to convert the sound pitch while maintaining high quality.

Conventionally, two methods are considered as a method of converting a sound pitch while maintaining a constant sound reproduction speed. One is a method of operating the acoustic waveform in the time domain. For example, when the pitch frequency is doubled, the acoustic signal is cut out at predetermined time intervals, and data is read out at twice the speed for each cutout section. I have to. In this case, the pitch frequency (the lowest frequency among the peak frequencies) is obtained from the data of the cut-out section, and a waveform having a double pitch frequency is added, so that only the pitch frequency can be doubled without changing the time. it can. Furthermore, pitch conversion can be realized by smoothly connecting the cutout sections thus processed, but in reality, the sound quality is impaired depending on the connection method, and the characteristics of the personal voice are not maintained and are unnatural. Since it becomes acoustic, various improvement methods are still being proposed.

[0006]

Another method is to operate in the frequency domain using a Fourier transform. An audio signal is cut out at predetermined time intervals, and a frequency amplitude component and a frequency phase component are extracted by Fourier transform. next,
This is a method in which the entire frequency band is frequency-shifted and phase-shifted by a desired shift amount, subjected to inverse Fourier transform, and then connected to cutout sections. However, even with this method, discontinuity between frames cannot be avoided, resulting in unnatural sound, and pitch conversion could not be performed well.

SUMMARY OF THE INVENTION It is an object of the present invention to provide a high-quality pitch conversion device which eliminates discontinuity between frames and enables natural pitch conversion.

[0008]

According to the present invention, as a means for achieving the above object, a digitally input sound signal is moved by a first predetermined sample at a time while a second predetermined sample (the second A predetermined sample >> a first predetermined sample), and convert a sound signal output from the first time window into a frequency domain signal consisting of an amplitude value and a phase from a time domain signal. Fourier transform means, amplitude processing means for shifting the amplitude value of the sound signal output from the Fourier transform means to a higher frequency side or lower frequency side, and a phase for rotating the phase of the sound signal output from the Fourier transform means Processing means, an inverse Fourier transform means for converting an acoustic signal output from the amplitude processing means and the phase processing means into a signal in a time domain, and Is intended to provide a pitch conversion device and having a second time window to be cut out from the first force is the sound signal for said first predetermined sample.

[0009]

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS Assuming that a continuous section is digitized to process a sound signal and one section cut out by a fixed number of samples is defined as a frame, digital processing for performing pitch conversion includes processing within a frame and inter-frame processing. Is divided into two processes.

<Process in Frame> As the process in the frame, a process of pitch-shifting the frequency of the acoustic signal is performed. Now, considering a sine wave signal f (t) as shown in FIG. 2, this signal is represented by f (t) = Aexp (jω ₀ * t). However, the angular velocity of the signal f (t) is ω ₀ . And
When the frequency when transposed on n semitone in the sine wave signal f (t), for example, on the music score and _{ω 1, ω 1 = ω 0} + Δω
Since the frequency is doubled when the frequency increases by one octave (12 sounds), the following equations (1) and (2) are obtained.

2 ^{(n / 12)} = ω ₁ / ω ₀ = (ω ₀ + Δω) / ω ₀ = 1 + Δω / ω ₀ = 1 + Δr Equation (1)

Ω ₁ = ω ₀ (1 + Δr) (2)

Therefore, assuming that a signal waveform transposed (pitch converted) on n semitones is g (t), g (t) is given by the following equation (3).
Can be represented by

G (t) = Aexp {jω₁* T} = Aexp {j (ω₀(1 + Δr) * t} = Aexp ｛j (ω₀* T + ω ₀ Δr * t)} = Aexp {j (θ₀+ Δθ} Equation (3)

Here, since θ ₀ is the initial phase of the signal f (t), the phase of the pitch-converted signal g (t) is rotated by Δθ. In the actual processing, the digitized original signal is subjected to fast Fourier transform (FFT) (more precisely, series expansion) to obtain the amplitude A and the initial phase θ _{0 of} the k ₀ th discrete frequency. Then, the discrete frequency k ₁ th that pitch conversion the k _0-th discrete frequency is a theta ₁ whose phase is rotated by Δθ in the amplitude A. This phase θ ₁ is expressed by the following equation (4).

Θ ₁ = θ ₀ + Δθ = θ ₀ + (ω ₁ −ω ₀ ) * t Expression (4) In addition, Expression (4) is obtained from Expression (2) by the following Expression (5). Become.

Θ ₁ = θ ₀ + ｛ω ₀ (1 + Δr) −ω ₀ ｝ * t = θ ₀ + ω ₀ * Δr * t = θ ₀ (1 + Δr) Equation (5) Further, Equation (1) Thus, the following equation (6) is obtained.

Θ ₁ = θ ₀ * 2 ^{(n / 12)} Equation (6) Therefore, if θ ₀ is known, θ ₁ is also known from Equation (6), so that g (t) is obtained from the signal f (t). ) Can be requested.

<Processing Between Frames> The processing between frames is processing for the purpose of smoothly connecting signals that have been pitch-converted in the above-described processing within a frame between the frames. As shown in FIG. 3, assuming that the amplitude of the signal f (t) in the N-th frame is A _N and the phase is θ _N , the signal f (t) appears periodically by performing an operation by FFT. The amplitude A _N and the phase θ _N can be calculated as a signal. Similarly,
The amplitude A _{N + 1} and the phase θ _{N + 1} in the (N + 1) -th frame can be obtained.

If the sample difference ΔL between the Nth frame and the (N + 1) th frame is larger than the frame length (FFT window length) L as shown in FIG.
_{N + 1} ≠ A _N , θ _{N + 1} ≠ θ _N + tΔLω ₀ , but ΔL is made sufficiently smaller than the frame length L, and the approximate expression A _{N + 1} =
If A _N , θ _{N + 1} = θ _N + tΔLω ₀ is satisfied, it is possible to calculate a signal that smoothly connects between the N-th frame and the N + 1-th frame.

[0021]

BRIEF DESCRIPTION OF THE DRAWINGS FIG. 1 is a block diagram showing an embodiment of a pitch conversion apparatus according to the present invention. FIG. 1 is a block diagram showing one embodiment of the pitch conversion device of the present invention. FIG. 4 is a diagram for explaining the processing contents. For example, consider a case in which an analog sound signal obtained by synthesizing a sine wave of 250 Hz and 315 Hz as shown in FIG. This acoustic signal is A /
Sampling frequency 44.1 supplied to D conversion circuit 1
It is temporarily stored in the storage circuit 2 as a digital audio signal f (t) of kHz. Then, for example, by the time window 3, FIG.
As shown in (A), 32768 (L) samples are cut out from time t _N to extract a digital sound signal f _N (t). Then, the 32768-sample sound signal f
_N (t) is supplied to the FFT circuit 4 and subjected to fast Fourier transform (FFT) to convert a signal in the time domain into a signal in the frequency domain.

At this time, each sample corresponding to the time domain corresponds to each frequency, and the sample number corresponds to the frequency. That is, when the audio signal data of the sampling frequency fs is cut out and processed every N samples, the frequency pH of the signal output from the FFT circuit 3 is
The sample number indicating z is the (p × N / fs) -th sample number. In the case of the present embodiment, the sampling frequency is 44.1 kHz.
Since the audio signal data of z is cut out every 32768 samples, the sample number indicating the frequency pHz is the (p × 32768/44100) th (rounded down below the decimal point).

The signal is separated into an amplitude component and a phase component and supplied to an amplitude processing circuit 5 and a phase processing circuit 6, respectively. Here, as shown in FIG. 4B, the array number k ₀ having the amplitude value A _N is moved by the pitch shift amount (three and a half) by the amplitude processing circuit 5, and the sample of the array number k ₁ (the amplitude Value A _N ). Similarly, the array number k ₀ having the phase θ _N is changed by the phase processing circuit 6 to the pitch shift amount (three and a half)
And converted into a sample of array number k ₁ having a phase θ ₁ further rotated by Δω * t _N. These signals are supplied to an IFFT circuit 7 where an inverse fast Fourier transform (IFFT) is performed to obtain an acoustic signal g _N (t). Further, only the first 16 (ΔL) samples supplied to the time window 8 are extracted and supplied to the D / A conversion circuit 9 as a part of the pitch-shifted audio signal g (t), and are converted into an analog audio signal. Is output.

Then, the sound signal of the next frame is one time smaller than the sound signal of the previously processed frame due to the time window 3.
An audio signal of length L samples after 6 (ΔL) samples is extracted, the same processing as above is performed, and an audio signal of the next ΔL sample is obtained and output. As a result of performing such processing 60 times within 20 msec, an acoustic signal as shown in FIG. 6 can be obtained. Note that the processing section L in the above embodiment is 3
Although the extraction interval ΔL is 16 samples in 2768 samples, if L >> ΔL, it is a matter of course that any other number of samples may be used.

[0025]

According to the pitch conversion apparatus of the present invention, since there is no discontinuity between frames, there is an effect that natural and high-quality pitch conversion with little deterioration of sound quality can be performed.

[Brief description of the drawings]

FIG. 1 is a block diagram showing an embodiment of a pitch conversion device according to the present invention.

FIG. 2 is a diagram for explaining the concept of pitch conversion.

FIG. 3 is a diagram for explaining clipping in a time window in one embodiment of the pitch conversion device of the present invention.

FIG. 4 is a diagram for explaining the processing steps of an embodiment of the pitch conversion device of the present invention.

FIG. 5 is a graph showing an example of an input signal of an embodiment of the pitch converter of the present invention.

FIG. 6 is a graph showing an example of an output signal of an embodiment of the pitch converter of the present invention.

[Explanation of symbols]

 Reference Signs List 1 A / D conversion circuit 2 Storage circuit 3 Time window (first time window) 4 FFT circuit (Fourier transform means) 5 Amplitude processing circuit (Amplitude processing means) 6 Phase processing circuit (Phase processing means) 7 IFFT circuit (Reverse Fourier transform means) 8 time window (second time window) 9 D / A conversion circuit

Claims (1)

[Claims] 1. A first time window for cutting out a digitally input audio signal by a second predetermined sample (here, a second predetermined sample >> a first predetermined sample) while moving the audio signal by a first predetermined sample. A Fourier transform unit for transforming an audio signal output from the first time window from a time domain signal into a frequency domain signal comprising an amplitude value and a phase; and an amplitude of the audio signal output from the Fourier transform unit. Amplitude processing means for shifting the value to the high frequency side or low frequency side; phase processing means for rotating the phase of the sound signal output from the Fourier transform means; output from the amplitude processing means and the phase processing means Inverse Fourier transform means for converting an audio signal into a signal in a time domain; and cutting off the first predetermined sample from the beginning of the audio signal output from the inverse Fourier transform means Pitch conversion device and having a second time window out.