JPH038035B2 - - Google Patents

Description

TECHNICAL FIELD The present invention relates to an audio signal processing circuit, and more particularly to a circuit that performs signal processing on a read signal from a recording medium in which an audio signal is digitally modulated and recorded as a pulse train. be.

BACKGROUND ART Conventionally, as a circuit for signal processing of a read signal from a recording medium in which an audio signal is digitally modulated and recorded in the form of a pulse train, such as a digital audio disk (hereinafter simply referred to as a disk), the circuit shown in FIG. I have something to show you. In the figure, recorded information on a disk (not shown) is read by a pickup 1. The read signal output from pickup 1 is demodulated by decoder 2,
It is converted into an analog signal by a D/A (digital/analog) converter 3 and sent to a day glitch circuit 4.
Spikes called glitches are removed in L and 4R, resulting in analog signals corresponding to the left (L) and right (R) channels.

The output signal of the day glitch circuit 4L is supplied to LPF (low pass filter) 5L and 6L, and the output signal of the day glitch circuit 4R is supplied to LPF5R and 6R.
is supplied to The cutoff frequency of LPF5L, 5R is set to 2OKHz, and the cutoff frequency of LPF6L, 6R is set to 4OKHz. supply to. Switch 7L, 7
R are interlocked with each other, and during normal disk playback operation, side a is selected, and during high-speed (for example, double speed) playback operation, side b is set.
Switch to the side.

In such a configuration, the spectrum of the demodulated signal during normal reproduction operation is as shown in FIG. 4A.
Here, fs is the sampling frequency when digitizing the audio signal, which is 44.1KHz. By the way, some digital audio disc players have a high-speed (for example, double speed) playback function.
During this double speed playback operation, the spectrum of the demodulated signal expands to twice as much as shown in FIG. 4B. Therefore, by preparing two types of film: LPF5L, 5R for standard mode and LPF6L, 6R for double speed mode, and using switches 7L, 7R according to the operating mode, the audio signal processing circuit can adjust the playback speed. It has also become possible to respond to

There is also another conventional example of an audio signal processing circuit as shown in FIG. In the figure, parts equivalent to those in FIG. 3 are indicated by the same symbols, and the demodulated output of the decoder 2 is supplied to a DSP (digital signal processor) 8. The DSP8 performs so-called oversampling, which converts the sampling frequency fs of the demodulated output to a higher frequency, for example, twice the sampling frequency 2fs, and cuts out 1/2fs to 3/2fs of the output spectrum.
It has an LPF configuration. The output of DSP8 is converted to an analog signal by D/A converter 3, and day glitch circuits 4L and 4R corresponding to the left and right channels
It is supplied to LPF9L, 9R via. LPF9
The cutoff frequency of L and 9R is set to 40KHz.

In such a configuration, the spectrum of the demodulated output during normal playback operation is as shown in FIG.
As shown in Figure B, signals of 1/2 fs to 3/2 fs are removed. In this way, the sampling frequency fs (=
44.1KHz), the LPF 9L, 9
The specifications required for R are simplified, and a filter with a small order can be used.

Here, when double-speed playback is performed, the spectrum of the demodulated signal expands to twice as much as shown in Figure 6C, and when this signal is oversampled by DSP8, the bandwidth reaches 4fs as shown in Figure 6D. will expand. Since the sampling frequency of LPF9L and 9R is set to 40KHz,
Regardless of whether the playback mode is normal playback or double speed playback, only the audio baseband component can be extracted from the outputs of the day glitch circuits 4L and 4R.

In each of the conventional circuits mentioned above, the former (the circuit shown in Figure 3) requires two LPF systems for extracting the audio baseband component, corresponding to the playback mode, which makes the circuit configuration complicated and However, it had the disadvantage of high cost. In addition, in the case of the latter (circuit shown in Figure 5), when playing at double speed,
The oversampling of the DSP8 spreads the signal spectrum to fs, so the drawback was that the DSP8 and all subsequent circuits had to operate at a very high speed of 4fs (176.4KHz).

Summary of the Invention The present invention was made to eliminate the drawbacks of the conventional ones as described above, and provides an audio signal processing circuit that requires only one LPF and can operate the DSP and subsequent circuits at 2fs. The purpose is to provide

The audio signal processing circuit according to the present invention has a structure in which the means for converting the sampling frequency of the demodulated output of the read signal read from the recording medium to a higher frequency is configured such that the conversion operation is performed only during normal playback and is prohibited during high-speed playback. It's summery.

Embodiments Hereinafter, embodiments of the present invention will be described in detail based on the drawings.

FIG. 1 is a block diagram showing one embodiment of the present invention, in which parts equivalent to those in FIG. 5 are designated by the same reference numerals. In the figure, the demodulated output of decoder 2 is transferred to DSP by switch 10A, 10B.
8 or directly to the D/A converter. The changeover switches 10A and 10B are on the a side during normal playback, and during high-speed playback, for example double speed playback, they are controlled by a changeover signal output from the control circuit 11 in response to a command signal supplied from an operation section (not shown). Switched to side b. Changeover switch 10A
The circuit configuration before and after the changeover switch 10B is the same as that shown in FIG. 5, and the cutoff frequency of the LPFs 9L and 9R is set to 40 KHz.

In this configuration, the spectrum of the demodulated output during normal playback operation is as shown in FIG. 2A,
Further, by performing, for example, double oversampling in the DSP 8, the DSP output becomes as shown in FIG. 2B. In addition, in Figure 2 A, 3/2f
In FIG. 2B, the high range above 5/2 fs is omitted.

Here, for example, when double speed reproduction is performed, the spectrum of the demodulated output becomes as shown in FIG. 2C. In other words, the normal audio baseband component from 0Hz to 20KHz becomes 0Hz to 40KHz, and fs
(=44.1KHz) sampling frequency is also 2fs (=
88.2KHz). Since the cutoff frequency of the LPFs 9L and 9R is set to 40 KHz, only the audio baseband components can be extracted from the outputs of the day glitch circuits 4L and 4R, regardless of whether the playback mode is normal playback or double speed playback.

In the above embodiment, the DSP 8 is bypassed by the changeover switches 10A and 10B during high-speed playback, but if the DSP 8 is configured to perform oversampling when selected,
It is also possible to prevent the oversampling operation of the DSP 8 from being performed by the signal from the control circuit 11, and the same effect as in the above embodiment can be achieved. Furthermore, the present invention is not limited to application to devices that play back digital audio discs, but can also be applied to devices that play back digital audio tapes.

Effects of the Invention As explained above, according to the audio signal processing circuit according to the present invention, the DSP converts the sampling frequency of the demodulated output of the read signal read from the recording medium to a higher frequency only during normal playback. Since the configuration is such that the operation is allowed and prohibited during high-speed playback, only one LPF is required, and
The operation of the DSP and subsequent circuits can be performed at speeds up to the speed of high-speed reproduction operation (for example, 2 fs in the case of double-speed reproduction).

[Brief explanation of the drawing]

FIG. 1 is a block diagram showing one embodiment of the present invention;
2 is a characteristic diagram for explaining the circuit operation of FIG. 1, FIG. 3 is a block diagram showing a conventional example, FIG. 4 is a characteristic diagram for explaining the circuit operation of FIG. 3, and FIG. 5 is a characteristic diagram for explaining the circuit operation of FIG.
The figure is a block diagram showing another conventional example, and Figure 6 is a block diagram showing another conventional example.
FIG. 3 is a characteristic diagram for explaining the operation of the circuit shown in the figure. Explanation of symbols of main parts 1...Pickup,
3...D/A converter, 4L, 4R...day glitch circuit, 5L, 5R, 6L, 6R, 9L, 9
R...Low pass filter, 8...Digital signal processor.

Claims (1)

[Claims] 1. An audio signal processing circuit that performs signal processing of a read signal from a recording medium in which an audio signal is digitally modulated and recorded as a pulse train, and converts the sampling frequency of the demodulated output of the read signal to a higher frequency. 1. An audio signal processing circuit comprising sampling frequency converting means for performing a conversion operation by said sampling frequency converting means only during normal playback of said recording medium and prohibiting said conversion operation during high speed playback.