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Showing papers on "Noise published in 1988"


Proceedings ArticleDOI
11 Apr 1988
TL;DR: The author presents a self-adapting noise reduction system which is based on a four-microphone array combined with an adaptive postfiltering scheme which produces an enhanced speech signal with barely noticeable residual noise if the input SNR is greater than 0 dB.
Abstract: The author presents a self-adapting noise reduction system which is based on a four-microphone array combined with an adaptive postfiltering scheme. Noise reduction is achieved by utilizing the directivity gain of the array and by reducing the residual noise through postfiltering of the received microphone signals. The postfiltering scheme depends on a Wiener filter estimating the desired speech signal and is computed from short-term measurements of the autocorrelation and cross-correlation functions of the microphone signals. The noise reduction system has been tested experimentally in a typical office room. The system produces an enhanced speech signal with barely noticeable residual noise if the input SNR is greater than 0 dB. The received noise power-measured in the absence of the speech signal-can be reduced by 28 dB. >

370 citations


Journal ArticleDOI
TL;DR: In this paper, a laboratory experiment was carried out to evaluate the relative annoyance of a low frequency noise and reference noises at the same dB(A) levels, and the results indicated that the low-frequency noise is more annoying than the reference noise at levels over 40dB(A).

85 citations


Proceedings ArticleDOI
A. Nadas1, David Nahamoo1, Michael Picheny1
11 Apr 1988
TL;DR: A probabilistic mixture model is described for a frame (the short-term spectrum) of each component of each to be used in speech recognition, which model the energy as the larger of the separate energies of signal and noise in the band.
Abstract: A probabilistic mixture model is described for a frame (the short-term spectrum) of each to be used in speech recognition. Each component of the mixture is regarded as a prototype for the labeling phase of a hidden Markov model based speech recognition system. Since the ambient noise during recognition can differ from the ambient noise present in the training data, the model is designed for convenient updating in changing noise. Based on the observation that the energy in a frequency band is at any fixed time dominated either by signal energy or by noise energy, the authors model the energy as the larger of the separate energies of signal and noise in the band. Statistical algorithms are given for training this as a hidden variables model. The hidden variables are the prototype identities and the separate signal and noise components. A series of speech recognition experiments that successfully utilize this model is also discussed. >

84 citations


Journal ArticleDOI
TL;DR: Vowel identification in quiet, noise, and reverberation was tested with 40 subjects who varied in age and hearing level and the relationship among hearing loss, age, and number and type of errors is discussed in light of acoustic cues available for vowel identification.
Abstract: Vowel identification in quiet, noise, and reverberation was tested with 40 subjects who varied in age and hearing level. Stimuli were 15 English vowels spoken in a /b–t/ context in a carrier sentence, which were degraded by reverberation or noise (a babble of 12 voices). Vowel identification scores were correlated with various measures of hearing loss and with age. The mean of four hearing levels at 0.5, 1, 2, and 4 kHz, termed HTL4, produced the highest correlation coefficients in all three listening conditions. The correlation with age was smaller than with HTL4 and significant only for the degraded vowels. Further analyses were performed for subjects assigned to four groups on the basis of the amount of hearing loss. In noise, performance of all four groups was significantly different, whereas, in both quiet and reverberation, only the group with the greatest hearing loss performed differently from the other groups. The relationship among hearing loss, age, and number and type of errors is discussed in light of acoustic cues available for vowel identification.

79 citations


PatentDOI
Robert W. Chang1
TL;DR: In this paper, a method and system for cancelling noise from sources that are distributed over a region, whereby two sensors are located so that one sensor will detect both voice signals and noise signals, and the other sensor will only detect only the noise signals.
Abstract: A method and system for cancelling noise from sources that are distributed over a region, whereby two sensors are located so that a first sensor will detect both voice signals and noise signals, and a second sensor will detect only the noise signals. The voice signals picked up at the second sensor are negligible, and the noise signals picked up at both sensors are correlated. The signals output from each sensor are connected to a predetermined number of narrowband filters in order to divide each respective signal into a predetermined number of frequencies, such as 15 for example. Thereafter, both signals are combined to cancel effectively the noise component from the signal output having both voice and noise to leave a voice signal that is substantially noise free.

54 citations


Patent
01 Sep 1988
TL;DR: In this article, the program audio signals and the noise signals are separated into a plurality of frequency bands, for example three bands, the envelope of which are demodulated and the derived demodulation signals are instantaneously compared to then control the transfer characteristics of the respective bands of the equalizer.
Abstract: To improve the fidelity of reproduction of automobile radios during conditions of extraneous in the passenger compartment, noise signals are derived, for example from a microphone installed in the engine compartment of the vehicle The audio program signals are conducted through a multichannel equalizer; the program audio signals and the noise signals are separated into a plurality of frequency bands, for example three bands, the envelope of which are demodulated and the derived demodulation signals are instantaneously compared to then control the transfer characteristics of the respective bands of the equalizer in accordance with the derived comparison between the respective bands of the noise and program signals Preferably, amplitude, change of transfer amplitude and rate of change is controlled by weighting the comparison signals and the rate of change thereof in relation to the frequency bands and the extent of change of the signals

45 citations


PatentDOI
Shogo Nakamura1
TL;DR: In this paper, a noise suppression apparatus has a main microphone for mainly picking up a voice and for outputting an input signal including an audio signal and a first noise component generated from a noise source.
Abstract: A noise suppression apparatus has a main microphone for mainly picking up a voice and for outputting an input signal including an audio signal and a first noise component generated from a noise source, a reference microphone for picking up a second noise component generated from the noise source, a filter bank for band-dividing the input signal from the main microphone and the second noise component from the reference microphone, and a noise cancel circuit for obtaining a phase difference between the input signal and the second noise component with respect to each divided band of the filter bank so as to correct the input signal based on the phase difference and for cancelling the first noise component in the input signal by use of the corrected input signal.

44 citations


Journal ArticleDOI
TL;DR: In this article, the hearing level of 133 railway workers who also hunted for sport was evaluated and compared with that of 82 non-hunters colleagues, both groups were affected by hearing loss, mostly involving the highfrequency range.
Abstract: The hearing level of 133 railway workers who also hunted for sport was evaluated and compared with that of 82 non-hunting colleagues. Both groups were affected by hearing loss, mostly involving the high-frequency range. Hunters were found to differ from non-hunters by having significantly worse hearing threshold in the ear contralateral to the shoulder supporting the firearm. The interaural threshold difference at 4 kHz was related to the number of rounds fired and exposure duration, thus providing an estimate of the adverse effect of gunfire noise to which the hunters had been exposed.

34 citations


Journal ArticleDOI
J.L. Eberhardt1
TL;DR: Road traffic noise during the first hours of a night's sleep tended to disturb sleep more than when it ocurred later in the night, the main effects being a reduction of the total amount of REM sleep during the night and an increased duration of intermittent wakefulness during the hours of exposure.

32 citations


PatentDOI
TL;DR: A conventional voice microphone placed in noncritical spaced relation to a source of intelligible speech sound while exposed to an acoustical field of ambient noise, electrically transmits output signals attenuated under control of a signal processing controller to which a sampled input of noise signals is fed by a reference microphone exposed to the same acoustically noise field as the voice microphone for audio reproduction of the speech sound without background noise by programming of the controller as mentioned in this paper.
Abstract: A conventional voice microphone placed in non-critical spaced relation to a source of intelligible speech sound while exposed to an acoustical field of ambient noise, electrically transmits output signals attenuated under control of a signal processing controller to which a sampled input of noise signals is fed by a reference microphone exposed to the same acoustical noise field as the voice microphone for audio reproduction of the speech sound without background noise by programming of the controller

31 citations


Patent
16 Feb 1988
TL;DR: In this article, an impulse-noise suppression system consisting of a preliminary blanking gate adapted for connecting to and for interrupting the AM-modulated signal path at the input of the IF section is described.
Abstract: To a standard AM radio receiver there is connected an impulse-noise suppression system comprising a preliminary blanking gate adapted for connecting to and for interrupting the AM-modulated signal path at the input of the IF section, and an audio blanking gate adapted for interrupting the audio circuit. Both blanking circuits detect impulse noise at the RF amplifier and with appropriate delays blank both points. Audio blanking masks the audio disturbance caused by the blanking in the AM-modulated-signal path. Audio blanking time is preferably from 2 to 3 times the duration of the blanking of the AM-modulated-signal path and is thus kept very short causing a minimum interruption of the wanted audio signal. Associated with the audio-signal-path blanking circuit is a sample and hold circuit for smoothing the blanked audio signal and virtually eliminating an audio disturbance or noise that is otherwise generated by the audio blanking circuit itself.

Journal ArticleDOI
TL;DR: A major finding in the study was a higher rating of annoyance among men with noise-induced hearing loss compared to men with normal hearing in the activities in which noise interferred with speech.

Patent
13 Jul 1988
TL;DR: In this paper, the authors propose to enable persons in a vehicle to enjoy an optimum sound from an audio equipment without masking by noise by inferring the amount of noise entering the in-vehicle space based on the driving state of the vehicle and the operational state of its autiliary equipments.
Abstract: PURPOSE:To enable persons in a vehicle enjoy an optimum sound from an audio equipment without masking by noise by inferring the amount of noise entering the in-vehicle space based on the driving state of the vehicle and the operational state of its autiliary equipments, and correcting the sound-volume and/or sound-quantity of an on-vehicle audio equipment based on the inferred amount of noise CONSTITUTION:The titled corrector is constituted of the following elements: a first parameter detection means to detect the driving state parameters of the vehicle, a second parameter detection means to detect the operational state parameters of auxiliary equipments aboard which cause a change in the noise in the vehicle when turned on and operating, an inference means to infer the amount of noise corresponding to the driving state parameters and/or operations state parameters of auxiliary equipments of the vehicle, and a correction means to correct the sound volume and/or the quality of sound corresponding to the amount of noise As a result, persons aboard are made able to enjoy an optimum sound from the audio equipment without masking by the noise

Journal ArticleDOI
TL;DR: In this article, the authors describe the temporal and spectral characteristics of the ambient sound heard by workers in open-plan offices, using a computer-based system, and the average A-weighted sound level for the seven offices ranged between 42.9 and 48.4 dB.
Abstract: This letter describes our initial efforts toward providing detailed descriptions of the temporal and spectral characteristics of the ambient sound heard by workers in open‐plan offices. Using a computer‐based system, the ambient sound in seven offices was investigated by sampling the noise in a workstation every 2 min between the hours of 7:00 a.m. and 7:00 p.m. None of the offices had electronic sound masking. It was found that, between the ‘‘core’’ times of 9:00 a.m. and 4:00 p.m., the average A‐weighted sound level for the seven offices ranged between 42.9 and 48.4 dB, with the average sound level for the seven offices being 44.9 dB. On the average, the spectra for the office noise generally had most of their energy at the lower frequencies, with the energy falling off at about 12 dB/decade at frequency bands above 100 Hz. Below 100 Hz, the spectra rolled off faster, at about 28 dB/decade.

Proceedings ArticleDOI
11 Apr 1988
TL;DR: A robust method is described for the determination of pitch in voiced speech that measures the regular harmonic spacing in the spectrum of the speech signal by applying an autocorrelation to the power spectral density.
Abstract: A robust method is described for the determination of pitch in voiced speech. The spectral autocorrelation function measures the regular harmonic spacing in the spectrum of the speech signal by applying an autocorrelation to the power spectral density. It is shown that this method is capable of accurate pitch tracking in additive noise. The preprocessing is critical for good performance, and in this case the speech spectrum is flattened by obtaining the residual from linear prediction analysis and the original speech. Application of suitable postprocessing enables this method to operate over a wide range of human pitch down to very low signal-to-noise ratios. >

Journal ArticleDOI
TL;DR: In 1971, there were no such products on the market for consumers, and there was still debate over what exactly people wanted as discussed by the authors, and two companies determined to solve both problems were Sony Corp. of Tokyo and The Victor Co. of Japan, known as JVC Ltd., Yokohama.
Abstract: Consumer electronics companies worldwide felt sure that the public would be interested in a machine that would tape their favorite television programs in their absence, for replay at home at their leisure. But in 1971 there were no such products on the market for consumers, and there was still debate over what exactly people wanted. Two companies determined to solve both problems were Sony Corp. of Tokyo and The Victor Co. of Japan, known as JVC Ltd., Yokohama.

Journal ArticleDOI
TL;DR: The effect of speaking over background noise on the normal voices of thirty adult female subjects was investigated in this article, where each subject performed a speaking-in-quiet task followed by 20 minutes vocal rest, before speaking for the same period of time over 80dBA white noise.
Abstract: The effect of speaking over background noise on the normal voices of thirty adult female subjects was investigated. Subjects were randomly assigned into three groups of 10 in order to study the effect of speaking over noise and duration of speaking on fundamental frequency and fundamental frequency variability. Each subject performed a speaking-in-quiet task followed by 20 minutes vocal rest, before speaking for the same period of time over 80dBA white noise. Using a sound level meter to monitor their voice level, subjects were required to speak as loudly as would be necessary to be understood by another person in 80dBA background white noise. The subjects' productions of /i:/, /u:/ and /a:/ were recorded before and after the speaking-in-quiet task, and before and after the speaking-in-noise task. Fundamental frequency and fundamental frequency variability were examined acoustically using the Fast Fourier Transform (Connor, 1975). Fundamental frequency was found to be significantly elevated after speaking...

Proceedings ArticleDOI
12 Jun 1988
TL;DR: An equipment-independent echo performance measure and an effective, subjective assessment system for overall speech quality evaluation is proposed for hand-free audio teleconference services.
Abstract: A discussion is presented of a quality assessment procedure for hand-free audio teleconference services, and an effective, subjective assessment system for overall speech quality evaluation is proposed. Simulated transmission impairments can be independently controlled due to the construction of the assessment system without voice processing equipment. Test results show the effectiveness of a mean opinion score for the services and the influence of room noise and talker echoes on speech quality. For quality design, the authors propose an equipment-independent echo performance measure. The conventional opinion equivalent SNR is available for quantization noise assessment. >

Patent
30 Jan 1988
TL;DR: In this article, the authors proposed a data type detector of a simple constitution capable of effectively discriminating in a short time by using error detecting data added to main data and discriminating the type of the main data.
Abstract: PURPOSE: To obtain the data type detector of a simple constitution capable of effectively discriminating in a short time by using error detecting data added to main data and discriminating the type of the main data. CONSTITUTION: An initial value used at the time of forming the error detecting data DRC added to reproduced transmitted audio DATAAD is selected to a different value so as to match to the audio data DATAAD. Thereby, at the time of reproducing mode, an error detecting operation is used to detect the error in the short time. This time is completely short to a time for forming a sound based on a reproduced audio signal and generates no noise like a sounding 'BARI' in the convention. When the format of audio data DATA is switched, the initial value is automatically switched to a value suitable for reproducing data DFB to completely shorten a period obtaining no error detection, so that the type of the audio data can be decided by sharing an error detecting circuit without disposing an exclusive circuit. COPYRIGHT: (C)1989,JPO&Japio

Patent
16 Feb 1988
TL;DR: In this article, the authors proposed a circuit arrangement for coupling alternating voltage signals having frequencies within the audio frequency range between a low noise audio frequency source and a zero ohm amplifier, wherein the audio source has a known source impedance.
Abstract: A circuit arrangement for coupling alternating voltage signals having frequencies within the audio frequency range between a low noise audio frequency source and a zero ohm amplifier, wherein the audio frequency source has a known source impedance. The circuit arrangement includes an output connected to the zero ohm amplifier and a highly resistant input for receiving the alternating voltage signals from the audio frequency source. The input is more resistant by at least one power of ten than the source impedance of the audio frequency source and the arrangement has a voltage amplification factor which is so low that no overshooting of the output signal will happen at the maximum occcurring voltage of the signals received at its input. The output of the arrangement is configured as a short-circuit resistant alternating current source which furnishes a maximum effective current of at least 10 mA. The equivalent noise resistance of the arrangement is maintained to be at most 20% of the equivalent noise resistance of the audio frequency. Any branch circuit included in the arrangement and connected to its output is so highly resistant that the internal gain of the subsequently connected zero ohm amplifier is not raised significantly, thereby causing no significant increase in noise in the zero ohm amplifier.

Proceedings ArticleDOI
28 Nov 1988
TL;DR: A system of high-quality, multichannel sound broadcasting by PCM (pulse code modulation) is described and it has been confirmed that no degradation of sound quality can be ensured when the receiving C/N ratio is above 8 dB.
Abstract: A system of high-quality, multichannel sound broadcasting by PCM (pulse code modulation) is described. In this system, 12-16 PCM-coded high-quality stereo audio signals are multiplexed by time division. This multiplexed signal digitally modulates a carrier which is sent to a broadcast satellite. Then, using one satellite broadcast channel in the 12 GHz band, the signal is broadcast throughput the country. For the preservation of sound quality, a BCH-DEC-TED (63, 50) error-correction code and an error-correction system which uses range transmission have been adopted. The improvement due to the use of range transmission was uniform over a wide range of received C/N (carrier/noise) ratios. The characteristics of the system have been tested by experiments using the broadcast satellite BS-2. It has been confirmed that no degradation of sound quality can be ensured when the receiving C/N ratio is above 8 dB. >

Patent
22 Dec 1988
TL;DR: In this article, a signal detection circuit 60 operates in combination with the high pass filter circuit to provide for automatic adjustment of the noise gate to control the release time, quickly for short notes and slowly for held notes.
Abstract: An audio noise gate has a main audio circuit (30) including a filter with a control terminal (32) for determining frequency response and having at least two different roll-off frequencies including one that rolls-off high end content audio signals and one that passes high end content audio signals. The main (30) is controlled from two secondary circuits one including a high pass filter for determining high frequency content of the audio signal and the other comprising a signal detection circuit (60) including a peak detector for determining absence or presence of a note. The signal detection circuit 60 operates in combination with the high pass filter circuit to provide for automatic adjustment of the noise gate to control the release time, quickly for short notes and slowly for held notes. The output of the high pass filter circuit (50) also includes a peak detector and a gain stage for controlling the frequencing response of the main circuit.

01 Jan 1988
TL;DR: In this article, the authors compare the resultats obtenus par refendage sur different aciers etires avec the results obtained by l'analyse du bruit ferromagnetique.
Abstract: Description du principe de la methode. Comparaison des resultats obtenus par refendage sur differents aciers etires avec ceux obtenus par l'analyse du bruit ferromagnetique

Proceedings ArticleDOI
17 Oct 1988
TL;DR: A digital audio system for business and regional air carrier aircraft reduces installation complexity and noise pickup, and provides significant performance enhancements to reduce pilot workload and stress.
Abstract: A digital audio system for business and regional air carrier aircraft reduces installation complexity and noise pickup, and provides significant performance enhancements. To meet the market place requirements, a new electrical data bus concept was developed to permit connecting stations freely at any points along an electrically long bus. The same protocol and bus management circuits may be used for a fiber optic version of the data bus. With most of the aircraft audio signals available in digital form, digital signal processing and computation techniques make possible both audio quality improvements and operational improvements to reduce pilot workload and stress.

Book ChapterDOI
01 Jan 1988
TL;DR: In this article, the authors discuss the need to increase the recording density of the wire recorder. But they do not discuss the benefits of increasing the number of recordings on the same record.
Abstract: After he had listened to his voice on the wonderful wire recorder of the late 1930s, Dr. Vagtborg, director of Armour Research Foundation asked, “Why must the wire move so fast? Why can’t we record slower and longer on the same record?” In other words, why not increase the recording density?

Patent
23 Jul 1988
TL;DR: In this paper, a stationary head records the analog audio signal as a consecutive track in the direction the same as the tape signal direction and reproduces the analog sound signal at queue/review.
Abstract: PURPOSE:To obtain a reproducing sound without having noise or abnormal sound by recording an analog audio signal at recording and at normal reproduction to form a consecutive track in a direction the same as the tape advancing direction and reproducing the analog sound signal at queue/review. CONSTITUTION:As soon as a rotary head 20 records a PCM signal at recording, a stationary head records the analog sound signal as a consecutive track in the direction the same as the tape signal direction and reproduces the analog sound signal at queue/review. In order to attain the similar queue/review to a tape recorded by a rotary head digital sound tape recorder without providing a stationary head 22 and an analog signal processing section 23, the analog sound signal 25 reproduced and outputted by the rotary head 20 and the digital signal processing section 21 is recorded as a consecutive track in the same direction as the tape running direction by the stationary head. Thus, the reproducing sound without having noise or abnormal sound is obtained.

Patent
28 Sep 1988
TL;DR: In this paper, the authors proposed a method to obtain the calling of good quality even in a place where a surrounding noise is loud by enlarging a received sound volume when the surrounding noise was loud, and returning it to an ordinary level when the noise was not present.
Abstract: PURPOSE:To obtain the calling of good quality even in a place where a surrounding noise is loud by enlarging a received sound volume when the surrounding noise is loud, and returning the received sound volume to an ordinary level when the surrounding noise is quiet and a transmitter input is present. CONSTITUTION:An M 11 is installed at a transmitter mouth piece side, and the M 12 is installed at a noise collector opening side on a contrary side, and the noise is collected from the noise collector opening. At the ordinary calling in the quiet place, a positive feedback quantity to an AMP 22 by an amplification control circuit GC 23 is prohibited. Thus, a received voice to a receiver REC 13 goes to the voice amplified ordinarily by the AMP 22. Next, at the calling in the place where the noise is loud, the amplification degree of the AMP 22 is elevated by adding the positive feedback quantity by the GC 23 to the AMP 22. Thus, the AMP 22 is amplified, for instance, 6 dB higher than ordinary time, and the received voice to the REC 13 can be heard clearly even in the place where the noise exists.

Patent
11 Oct 1988
TL;DR: In this article, the authors propose to automatically prevent the generation of a strange sound at the joint of audio signals by starting to record the signal to be recorded, or setting a pause mode or a stop mode, after a PCM audio signal corresponding to 0 level is recorded in a prescribed frame.
Abstract: PURPOSE:To automatically prevent the generation of a strange sound at the joint of audio signals by starting to record the signal to be recorded,or setting a pause mode or a stop mode, after a PCM audio signal corresponding to 0 level is recorded in a prescribed frame CONSTITUTION:A control circuit 40 controls respective circuits and means and so on, correspondingly to an input from an inputting means 41, and makes them execute prescribed operations When a record mode is inputted, before the signal to be recorded is recorded, the PCM audio signal corresponding to 0 level is automatically recorded and the recording of the signal to be recorded is executed subsequently Besides, at the time of the recording mode, when the pause mode or the stop mode is inputted, the PCM audio signal corresponding to 0 level is recorded automatically in the prescribed frame, and the pause mode or the stop mode is executed after the said recording is finished Thus, the generation of a noise at the joint can be prevented

Journal ArticleDOI
TL;DR: In this paper, the authors obtained threshold thresholds for sinusoidally amplitude-modulated (SAM) broadband noise for modulation frequencies from 2 to 512 Hz using a two-interval, forced-choice adaptive procedure.
Abstract: Modulation thresholds for sinusoidally amplitude‐modulated (SAM) broadband noise were obtained for modulation frequencies from 2 to 512 Hz using a two‐interval, forced‐choice adaptive procedure. The noise carrier was on continuously throughout a block of trials, and was modulated for 500 ms in one of the two observation intervals. Thresholds were obtained in quiet and in the presence of a SAM broadband noise masker. In the masking conditions, the same noise carrier, presented at a spectrum level of 15 dB SPL, was used for the signal and the masker. The masker was modulated in both of the 500‐ms observation intervals. The modulation frequency of the masker was 4, 16, or 64 Hz; its modulation depth (m) was 0 (no modulation), 0.5, or 1.0. For a given masker modulation frequency, the modulation masking patterns generally were bandpass, with the greatest amount of masking occurring when the signal and masker modulation frequencies were the same. With a few consistent exceptions, there was a monotonic relation ...

Patent
23 Mar 1988
TL;DR: In this article, the authors proposed a method to prevent forgery or illegal use of a magnetic card by using a Magnetic card in which a magnetic medium is formed on a magnetic film and recording random noise data to mask true data on said magnetic medium.
Abstract: PURPOSE:To prevent forgery or illegal use of a magnetic card by using a magnetic card in which a magnetic medium is formed on a magnetic film and recording random noise data to mask true data on said magnetic medium. CONSTITUTION:On the lower magnetic film 12 of a magnetic card 1, a bit of true data 12a such as money amount is recorded by a magnetic head that generates a strong recording magnetic field. At the time of this recording, the true data is recorded simultaneously on the upper magnetic medium 13, however, later, senseless noise data 13a to mask the true data is recorded on the magnetic medium 13 with a weak magnetic field that does not erase the true data 12a recorded on the magnetic film 12. At the time of recording/ reproducing the true data, the recording/reproducing of true data is performed with a data magnetic head 2 after erasing only the random noise erasing head 4, thereafter, the random noise data is recorded on the upper layer with a noise magnetic head 3. Thus the true data is masked again not to be read.