JPH035597B2 - - Google Patents

Abstract

PURPOSE:To reduce the influence of higher harmonic noise due to the discontinuance of a waveform by providing a variable low-pass filter whose cutoff frequency is controlled by a control signal based upon the processing arithmetic result of a sound piece connection part by an arithmetic processing means. CONSTITUTION:A playback signal reproduced at a different speed from that in recording is equalized by a reproducing equalizer 2 and amplified, and the amplified signal is written in, for example, four BBDs 3-6 and read successively to be outputted through the switching of analog switches 7-10, so that reproduction is carried out through a variable low-pass filter 19. On the other hand, the output of the reproducing equalizer 2 is applied to a variable and-pass dividing filter 20, and levels of a low and a high frequency part are detected by sensors 21 and 22. Both the levels are compared mutually by a level comparator 23; when the level of the low frequency component is higher, a variable low-pass filter 19 operates to cut higher harmonics of the discontinuous part of a signal, and when not, the variable low-pass filter is turned off to mask higher harmonic noise with the signal of the high frequency component.

Description

DETAILED DESCRIPTION OF THE INVENTION The present invention relates to processing of a time axis compression/expansion processing device (hereinafter abbreviated as “time axis compression/expansion device”) used when reproducing an audio signal recorded on a recording tape or the like at a speed different from the recording speed. Concerning signal quality improvement.

In order to listen to the content recorded on a tape recorder or the like more slowly or to understand it in a shorter time, the playback speed is often made variable. In this case, simply changing the playback speed changes the frequency band of the playback signal, resulting in only a playback sound with reduced predictability. Therefore, in order to return the changed frequency band of the reproduced signal to the frequency band of the original signal and obtain reproduced sound with good intelligibility, a time axis compression operation is performed. There are several methods for time axis companding processing, for example
It is known to use an analog storage element such as a BBD and change the write clock signal depending on the playback speed for a constant read clock signal.

FIG. 1 is a block diagram of a time-base companding circuit of a variable speed playback tape recorder using a BBD. The signal picked up by playback head 1 is sent to the playback equalizer (P.Eq)
After being equalized and amplified by 2, there are 4 BBDs, i.e.
BBD-13, BBD-24, BBD-35, BBD
-46 are sequentially written using the write clock (fw). Reading is also performed sequentially from BBD-13 to BBD-24 with the read clock (f _R ). The read signal is an analog switch
The signal is output as one serial signal by opening and closing AS-17 to AS-410, and is applied to the amplifier 11. By periodically writing and reading information using four BBDs, time-axis companding processing becomes possible. However, if this process is simply repeated periodically, when the output signals of four BBDs are combined into a serial signal, discontinuities will occur in the signal waveform at most of the connections. This waveform discontinuity causes harmonic noise, lowers the S/N of the synthesized sound, and reduces clarity.

Therefore, a connection method that does not cause this waveform discontinuity is required, and the arithmetic processing section 13 is responsible for this process. FIG. 2 shows the principle of this process, and is an example when the playback speed is approximately 0.7 times the recording speed. In FIG. 2, T ₁ , T _{2 .} . . are processing cycles, which are usually set to a constant value of 20 to 30 msec.
The reproduced and amplified signals are sequentially written to each BBD,
The signals in the sections t ₁ , t ₂ , t ₃ , and t ₄ are respectively
Incorporated into BBD. This signal is sequentially read out in processing cycles from T ₄ to _{T 7} and becomes a composite output. Here, the intervals t ₁ , t ₂ , t ₃ , t ₄ in which signals are written to each BBD
is determined by the arithmetic processing unit 13, and a signal waveform without discontinuities is synthesized. In other words, the rear end of the signal written to the BBD being read in a certain processing cycle and the BBD to be read next
The correlation with the leading end of the signal being written in is calculated in parallel with the writing process, and the timing to stop the write clock is determined so that the correlation is the highest. Through this processing, a high-quality composite waveform with smooth connections can be obtained. This process divides the reproduced signal into relatively short (20 to 30 msec) signal pieces,
In the case of low-speed playback, a part of the playback signal is repeatedly used, and in the case of high-speed playback, a part of the playback signal is cut out and synthesized, so the redundancy of the target signal is reduced. It can be said that it largely depends on gender. Therefore, depending on the timing of signal acquisition, waveform connection processing may not be performed satisfactorily by controlling the write clock alone. At this time, discontinuity occurs in the waveform at the connection portion of the composite signal waveform, causing harmonic noise as described above and degrading the quality of the composite signal. Looking at this state from the arithmetic processing unit 13, it is nothing but a result of the series of correlation calculations of the waveform patterns described above, and there is no value that should be recognized as an extreme value among the calculated values, and no correlation can be taken. Therefore, if it is possible to set an experimentally confirmed reference value for determining whether or not a certain calculation result is recognized as an extreme value, the arithmetic processing unit 13 can take some action if discontinuity occurs in the composite signal waveform. . An example of a time-base companding circuit using a variable low-pass filter 19 based on this idea is shown in FIG. In FIG. 3, the basic operation of time axis companding is exactly the same as in FIG. Here, a variable low-pass filter (Var.LPF) 19 controlled by the arithmetic processing unit 13 is used for the purpose of reducing the influence of harmonic noise caused by the aforementioned waveform discontinuity.
is provided.

Generally, the frequency spectrum of audio signals is 300Hz~
It has a peak around 500Hz, and in the higher frequency range, the level decreases as the frequency increases. On the other hand, the noise caused by the discontinuity of the waveform has a spectral distribution that extends to a high frequency region, and is audibly unpleasant.
A variable low-pass filter 19 is used to reduce the influence of such harmonic noise. The timing at which the variable low-pass filter 19 is driven is when, as a result of the correlation calculation, no calculated value greater than a certain reference value is obtained, as described above. Variable low pass filter 1
9. The cutoff frequency is normally set to a value (about 6 KHz) that does not affect the audio quality, but at the time of driving, it is lowered to about 2 to 3 KHz, which provides the effect of reducing high frequency noise components. With the configuration shown in FIG. 3, the influence of noise generated at the connection can be effectively reduced when correlation cannot be established. By the way, when paying attention to the spectral distribution of the reproduced signal, if the reproduced signal is dominated by low frequency components, even if the variable low-pass filter 19 is operated as described above to reduce the influence of noise generated at the connection part, It is thought that the effect on the signal itself is relatively small. However, if the reproduced signal contains many frequency components higher than the cutoff frequency when the variable low-pass filter 19 is operating, operating the variable low-pass filter 19 will affect the signal itself, or in other words, reduce the effect on some of the signals. The problem is that some ingredients are missing. Furthermore, if the level of a frequency component higher than the cutoff frequency during operation of the variable low-pass filter 19 (referred to as a high-frequency component for convenience) is sufficiently high, the high-frequency component will have a masking effect on harmonic noise. In the case of a reproduced signal containing many high frequency components, there is room for further improvement in operating the variable low-pass filter 19 described above.

FIG. 4 shows an embodiment of the time-base companding circuit of the present invention, which has been made with this point in mind.

In the configuration shown in FIG. 4, the time axis companding process is the same as in FIGS. 1 and 3. The reproduction signal from the reproduction equalizer 2 is passed through a variable band division filter 20 whose crossover frequency (fcr) can be changed by the reproduction speed control input (Vs).
is input. In normal playback, the crossover frequency (fcr) of the variable band division filter 20 is as follows:
fcrn=2~3KHz, but as the playback speed changes, if the playback speed (m) is m=variable playback speed/normal playback speed...(1), then fcr=m・fcrn... (2) It is controlled by the arithmetic processing unit 13 according to.

The low frequency component from the variable band division filter 20 is detected by a level detector 121 that detects the level of the low frequency component.
The high frequency component is input to a level detector 222 which detects the level of the high frequency component. As mentioned above, looking at the spectral distribution of the audio signal, the level of the high-frequency signal is lower than the level of the low-frequency signal, but it is a band that is audible to the ear. Therefore, the gains of the level detectors 2 and 22 have been subjected to this audible correction. The low level signal from the level detector 121 is input to the non-inverting input terminal of the level comparator 23, and the high level signal from the level detector 222 is input to the inverting input terminal of the level comparator 23. The output of the level comparator 23 becomes one input of an AND gate 24, which gates the variable low-pass filter operation control signal from the arithmetic processing section 13. AND gate 24
The output of is input to the control input terminal of the variable low-pass filter 19. The operation of this configuration will be explained. The variable band division filter 20 is controlled so that its crossover frequency changes according to the playback speed as shown in equation (2) above, so even if the frequency band of the playback signal changes depending on the playback speed, The low frequency component and high frequency component can be accurately divided and extracted. Therefore, the level signals from the level detector 121 and the level detector 222 are the same as the low level signal and the high level signal during normal reproduction. The level comparator 23 compares the low level signal and the high level signal, and if the low level signal is higher than the high level signal, the output of the level comparator 23 is "H" and the arithmetic processing unit 13 The variable low-pass filter operation control signal from is applied to the variable low-pass filter 19 without being gated by the AND gate 24. That is, in this case, the low-frequency signal component is sufficiently dominant, and the influence of the operation of the variable low-pass filter 19 is small. Conversely, if the high-frequency level signal is larger than the low-frequency level signal,
The output of the level comparator 24 becomes "L", gates the variable low-pass filter operation control signal from the arithmetic processing section 13, and the variable low-pass filter 19 does not operate. That is, in this case, the quality of the signal is significantly degraded due to the loss of high-frequency components due to the operation of the variable low-pass filter 19, and good results can be obtained by not operating the variable low-pass filter 19.

In addition, in the above-mentioned embodiment, an example was explained in which a BBD was used as an analog storage element, but the present invention can also be applied to an A-D converter, a RAM (random memory element), etc.
It can be applied in exactly the same way to the case of digital processing using an access memory) or a DA converter.
Furthermore, it goes without saying that if a digital filter is used and controlled by an arithmetic processing section, level detection of band components can also be processed by software.

As described above, according to the configuration according to the present invention,
Compared to the case where the variable low-pass filter is activated immediately based on the correlation calculation result of the waveform pattern of the connection part, it is possible to realize a time-base companding apparatus that can obtain a processed signal of higher quality overall.

[Brief explanation of the drawing]

Figure 1 is a circuit diagram showing a conventional time-base companding circuit.
Fig. 2 is a signal processing time chart showing signal processing to prevent discontinuities from occurring at the connecting parts of speech segments, and Fig. 3 is a time axis compression/expansion circuit of the present invention using a variable low-pass filter. FIG. 4 is a block circuit diagram showing another embodiment of the present invention. 3, 4, 5, 6... storage means, 7, 8, 9, 1
0...Analog switch, 11...Amplifier, 12
...A-D converter, 13... Arithmetic processing circuit, 14
... Clock control circuit, 15 ... Voltage controlled oscillator, 16, 18 ... Frequency dividing circuit, 17 ... Oscillator,
19... Variable low-pass filter, 20... Variable band division filter, 21, 22... First and second level detector, 23... Level comparator, 24... AND gate.

Claims (1)

[Claims] 1. When performing a connection process on speech segments extracted by temporally dividing a speech signal, the degree of correlation between a value near the rear end of a preceding speech segment and a value near the leading end of a subsequent speech segment. arithmetic processing means for calculating and controlling the leading end of the subsequent speech piece to be connected to the rear end of the preceding speech piece at the point when the degree of correlation is highest; 1. A time-base compression/expansion circuit for an audio signal, comprising: a variable low-pass filter whose cutoff frequency is controlled by a control signal based on the result. 2. When connecting the speech segments extracted by temporally dividing the speech signal, calculate the degree of correlation between the value near the rear end of the preceding speech segment and the value near the leading end of the succeeding speech segment, and calculate the correlation. an arithmetic processing means for controlling the leading end of the subsequent speech piece to be connected to the rear end of the preceding speech piece at the highest point in time, and a control signal based on the result of the processing calculation of the connection part by the arithmetic processing means; a variable low-pass filter whose cut-off frequency is controlled; a variable band division filter whose crossover frequency can be changed according to the time-domain companding ratio of the audio signal; and a variable band division filter to which the input audio signal is supplied; and from the variable band division filter. a first level detector that detects the signal level of the low frequency signal from the variable band division filter; and a second level detector that detects the signal level of the high frequency signal from the variable band division filter.
It is characterized by comprising a level detector, and a level comparator that compares the output signals of both the level detectors and gates the control signal of the variable low-pass filter when the output of the second level detector is larger. A time-base compression/expansion circuit for audio signals.