JPH07273659A - Method and device for processing digital signal and recording medium - Google Patents

Abstract

PURPOSE:To reduce quantization noise and deterioration in sound quality even from a signal with a large tonality such as the signal of a trumpet sound. CONSTITUTION:The device is provided with band division filters 11, 12 decomposing a digital signal into plural frequency band components, MDCT circuits 13-15 obtaining signal components in a block having a definite time width and a definite frequency width, nonlinear processing circuits 40, 41, 42 applying nonlinear processing to an MDCT coefficient, and an adaptive bit coding circuit 18 quantizing the signal component for each block to implement information compression and providing outputs of both the signal component whose information is compressed and information compression parameters required for compressing the information for each block, and the nonlinear processing circuits 40, 41, 42 make a quantized value zero with a signal to noise ratio as to samples with smaller signal to noise ratio or increase the spectrum value when the signal tonality in the floating block in the high efficiency coding is high.

Description

Detailed Description of the Invention

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a digital signal processing method and apparatus for recording or transmitting compressed data obtained by bit-compressing, for example, a digital audio signal, and recording compressed data compressed by this digital signal processing method or apparatus. In particular, the present invention relates to a digital signal processing method and apparatus for handling a digital audio signal including a signal with high tonality, and a recording medium.

[0002]

2. Description of the Related Art The applicant of the present invention has previously disclosed a technique for bit-compressing an input digital audio signal and recording it on a recording medium in a burst manner with a predetermined amount of data as a recording unit, for example, in Japanese Unexamined Patent Publication (Kokai) No. 105270 publication, US
Appln. SN 08 / 171,263, USP 5,243,588, USP 5,24
Proposed in each specification and drawings of 4,705.

In the specification and drawings of the above-mentioned SN 08 / 171,263, in the lead-in area in which the index data indicating the recording position of the data area is subcoded and recorded, the display data concerning the recorded contents of the data area is displayed. A disc recorded as main data, a disc recording device having a recording unit for recording data on the disc, a reproducing unit for reproducing data from the disc, and a display unit for displaying according to display data obtained by the reproducing unit. And a disc reproducing apparatus having the same. Further, in the above-mentioned Japanese Patent Application Laid-Open No. 4-105270, input data that is continuously input is sequentially written, and the written input data is sequentially read as recording data having a transfer speed higher than the transfer speed of the input data. Memory means,
Rotational drive means capable of switching the speed of rotating the disk-shaped recording medium, recording means for recording the recording data read from the memory means on the disk-shaped recording medium, and input data recorded in the memory means. Memory control means for sequentially reading the recording data by a predetermined amount from the memory means when the data amount exceeds a predetermined amount, and performing memory control so as to secure a write space in the memory means for the predetermined data amount or more; A disk recording device comprising: a recording control means for controlling a recording position so as to continuously record the recording data, which is read out discontinuously from the memory means by the memory control means, on a recording track on the disc-shaped recording medium. , And a disc reproducing apparatus corresponding to this. Also, USP 5,24
In the specification and drawings of 3,588, the storage means for temporarily storing digital data, the digital data from the storage means is clustered into a certain number of sectors, and the interleave length at the time of interleaving processing is applied to the connection part of each cluster. There is described a disk recording apparatus having a long sector for sector connection, a recording means for interleaving digital data and recording it on the disk-shaped recording medium, and a disk reproducing apparatus corresponding thereto. Further, in each specification and drawings of USP 5,244,705, in the disk-shaped recording medium on which compressed audio data and the like are recorded, the inner diameter of the data recording area of the disk-shaped recording medium is set to a predetermined value within the range of 32 mm to 50 mm. When the inner diameter of the data recording area is 32 mm, the outer diameter is 60
By setting the value within the range of mm to 62 mm, and the outside diameter when the inside diameter of the data recording area is 50 mm, the outside diameter is within the range of 71 mm to 73 mm, so that it can be used in a compact portable disk recording / playback device. And, for example, the compression rate is 1/4
By recording compressed audio data of standard 12
It is described that a reproduction time comparable to that of cmCD can be realized.

The technology proposed in the above specifications and drawings uses a magneto-optical disk as a recording medium, and is a so-called compact disk (CD) CD-I (CD-interactive) or CD-ROM.
AD (adaptive difference) PCM audio data specified in the XA audio data format is recorded / reproduced. For example, 32 sectors of this ADPCM audio data and several sectors for linking for interleave processing are recorded as a recording unit. As an example, ADPCM audio data is recorded in a burst form on a magneto-optical disk.

Several modes can be selected for the ADPCM audio data in the recording / reproducing apparatus using this magneto-optical disk. For example, the compression rate is twice as high as that of a normal CD reproducing time. Level A mode with a sampling frequency of 37.8 kHz, level B mode with a compression rate of 4 times and a sampling frequency of 37.8 kHz, sampling frequency of 18.9 kHz with a compression rate of 8 times
There are z level C modes.

That is, for example, in the case of the level B, the digital audio data is compressed to approximately 1/4, and the reproduction time (play time) of the disc recorded in this level B mode is a standard CD. It is four times as large as in the case of the format (CD-DA format). According to this, since a recording / reproducing time of the same size as that of a standard disc having a diameter of 12 cm can be obtained with a smaller disc, the device can be miniaturized.

However, in the recording / reproducing apparatus using this magneto-optical disk, the rotational speed of the disk is the same as that of a standard CD. Therefore, for example, in the case of the level B, the reproducing time is four times as long as the predetermined time. Compressed data will be obtained. For this reason, for example, the same compressed data is read four times in duplicate in time units such as sectors and clusters.
Only the compressed data for one time is sent to the audio reproduction. Specifically, when scanning (tracking) a spiral recording track, a track jump is performed to return to the original track position for each rotation, and the same track is repeatedly tracked four times. Playback operation will proceed. this is,
For example, it is only necessary to obtain normal compressed data at least once out of four times of redundant reading, which is resistant to errors due to disturbances and the like, which is particularly preferable when applied to a small portable device.

Further, in the future, it is considered to use a semiconductor memory as a recording medium, and it is desirable to perform additional bit compression in order to further improve the compression efficiency. Specifically, an audio signal is recorded and reproduced using a so-called IC card in which an IC (integrated circuit) including a semiconductor memory is arranged in the card.
Compressed data that has been bit-compressed is recorded and reproduced on the C card.

IC cards and the like using such a semiconductor memory are expected to have an increased recording capacity and a lower price as the semiconductor technology advances, but at the initial stage when they are supplied to the market. It is thought that the capacity is low and expensive. Therefore, it is sufficiently conceivable to transfer the contents from another inexpensive and large-capacity recording medium such as the above-mentioned magneto-optical disk to an IC card or the like for frequent rewriting and use. Specifically, for example, of a plurality of songs recorded on the magneto-optical disk, a favorite song is dubbed to an IC card, and when it is no longer needed, it is replaced with another song. By frequently rewriting the contents of the IC card in this manner, various songs can be enjoyed outdoors, etc., with a small number of IC cards held in hand.

[0010] The applicant of the present invention has previously proposed that the EUROPEAN PATEN
T APPLICATION publication number: 0 525 809 A2 (Da
te of publication 03.02.93) proposes a suitable encoding method for generating the above-mentioned compressed data.

Further, the applicant of the present application is the EUROPEAN PATENT AP
PLICATION publication number: 0599 719 A1 (Date o
f publication 01.06.94), EUROPEAN PATENT APPLICATI
ONpublication number: 0 601 566 A1 (Date of publi
cation 15.06.94), and International Publication Nu
mber: WO 94/19801 (International Publication Date
: 1 September 1994) proposes a recording / reproducing system suitable for recording / reproducing using the above-mentioned IC card.

[0012]

By the way, when the bit rate of the high efficiency code is lowered for the purpose of extending the recording time, the deterioration of the sound quality becomes gradually noticeable. This is especially noticeable for music signals for which the auditory effect is difficult to work.

Therefore, in view of the above, the present invention does not make the algorithm complicated and makes it feel unnatural when the bit rate of the high efficiency code is lowered for the purpose of extending the recording time. A digital signal processing method and device capable of obtaining sound quality that is easy to listen to,
It is an object of the present invention to provide a recording medium in which compressed data processed by the digital signal processing method or device is recorded.

[0014]

DISCLOSURE OF THE INVENTION A digital signal processing method of the present invention is proposed for achieving the above-mentioned object, and is for transmitting a digital signal, wherein an input digital signal is Converting to a signal component in a plurality of blocks each having a finite time width and a finite frequency width including a plurality of signal components, nonlinearly processing the signal components in at least some of the plurality of blocks, and It is characterized in that the processed signal component is quantized.

Further, the digital signal processing apparatus of the present invention is for transmitting a digital signal, and the input digital signal is stored in a plurality of blocks each having a finite time width and a finite frequency width each including a plurality of signal components. Transforming means for transforming the signal components into a signal component, non-linear processing means for performing non-linear processing on the signal components in at least some of the plurality of blocks, and encoding means for quantizing the non-linearly processed signal components. By having and, the above-mentioned object is achieved.

Further, the recording medium of the present invention converts a digital signal into a signal component in a plurality of blocks having a finite time width and a finite frequency width, each of which includes a plurality of signal components,
Non-linear processing of a signal component in at least a part of the plurality of blocks, and recording data generated by quantizing the non-linearly processed signal component is recorded. Is.

[0017]

In the digital signal processing method and apparatus and the recording medium of the present invention, an input digital signal is converted into signal components in a plurality of blocks each having a finite time width and a finite frequency width, each of which includes a plurality of signal components. , The signal components in at least some blocks of the plurality of blocks are non-linearly processed, and the non-linearly processed signal components are quantized, for example, non-linearly processed with respect to a block including a component with high tonality. Data is obtained.

[0018]

Embodiments of the present invention will be described below with reference to the drawings.

First, FIG. 1 shows a schematic configuration of a compressed data recording / reproducing apparatus for recording / reproducing compressed data obtained by bit-compressing a digital audio signal to / from a recording medium as an embodiment for realizing the digital signal processing method of the present invention. Indicates.

The compressed data recording / reproducing apparatus of FIG. 1 includes a magneto-optical disk recording / reproducing unit for recording / reproducing compressed data on / from a magneto-optical disk 1 which is an example of a recording medium of the present invention, and another example. IC card 2 as recording medium
It is configured by assembling two units, an IC card recording unit for writing / reading compressed data with respect to the above, into one system.

When the signal reproduced by the reproducing system on the side of the magneto-optical disk recording / reproducing unit is recorded by the IC card recording unit, first, the data is read from the magneto-optical disk 1 of the reproducing system by the optical head 53. To be
This data is sent to the decoder 71 and EFM (8-1
4) Demodulation, deinterleave processing, error correction processing, etc. are performed to obtain reproduced compressed data. The reproduced compressed data is sent to the memory 85 of the IC card recording unit and temporarily stored therein. Reproduced compressed data read from the memory 85 is subjected to additional processing such as variable bit rate encoding processing by an additional compressor 84 that performs entropy encoding, and then the reproduced compressed data subjected to the additional processing. Via the IC card interface circuit 86
Written to C card 2. In this way, the compressed data reproduced from the magneto-optical disk 1 is the ATC decoder 73.
The above IC in the compressed state before being subjected to the expansion processing by
The IC card 2 is sent to the recording system for the card 2.
Written in.

By the way, during normal reproduction, that is, during reproduction for listening to audio, compressed data is read from the recording medium, that is, the magneto-optical disk 1 in a predetermined data amount unit (for example, 32 sectors + several sectors) intermittently or in bursts. This is expanded and converted into a continuous audio signal. When performing the so-called dubbing as described above, the compressed data on the magneto-optical disk 1 is continuously read and sent to the IC card recording unit. I am recording. This enables high-speed (short-time) dubbing according to the data compression rate.

The specific structure of the compressed data recording / reproducing apparatus shown in FIG. 1 will be described below in detail.

In the magneto-optical disk recording / reproducing unit of the compressed data recording / reproducing apparatus shown in FIG. 1, a magneto-optical disk 1 rotatably driven by a spindle motor 51 is used as a recording medium. At the time of recording data on the magneto-optical disk 1, for example, so-called magnetic field modulation recording is performed by applying a modulation magnetic field according to the recording data with the magnetic head 54 while irradiating the optical head 53 with laser light, thereby performing so-called magnetic field modulation recording. Data is recorded along the recording track of. At the time of reproduction, the recording track of the magneto-optical disk 1 is traced with laser light by the optical head 53 to reproduce the data magneto-optically.

The optical head 53 includes, for example, a laser light source such as a laser diode, a collimator lens, an objective lens,
It is composed of a polarization beam splitter, an optical component such as a cylindrical lens, and a photodetector having a light receiving portion of a predetermined pattern. The optical head 53 is provided at a position facing the magnetic head 54 through the magneto-optical disk 1. When recording data on the magneto-optical disk 1, the head driving circuit 66 of the recording system drives the magnetic head 54 to apply a modulation magnetic field according to the recording data, and at the same time, the optical head 53 drives the magneto-optical disk 1.
By irradiating the target track with laser light, thermomagnetic recording is performed by the magnetic field modulation method. Further, the optical head 53 detects the reflected light of the laser light applied to the target track, detects a focus error by, for example, a so-called astigmatism method, and detects a tracking error by, for example, a so-called push-pull method. When reproducing data from the magneto-optical disk 1, the optical head 53 detects the focus error and the tracking error, and at the same time, detects the difference in the polarization angle (Kerr rotation angle) of the reflected light of the laser light from the target track and reproduces it. Generate a signal.

The output of the optical head 53 is supplied to the RF circuit 55. The RF circuit 55 extracts the focus error signal and the tracking error signal from the output of the optical head 53 and supplies them to the servo control circuit 56, and binarizes the reproduction signal to supply it to the decoder 71 of the reproduction system.

The servo control circuit 56 is composed of, for example, a focus servo control circuit, a tracking servo control circuit, a spindle motor servo control circuit, a sled servo control circuit and the like. The focus servo control circuit controls the focus of the optical system of the optical head 53 so that the focus error signal becomes zero. Further, the tracking servo control circuit controls the tracking of the optical system of the optical head 53 so that the tracking error signal becomes zero. Further, the spindle motor servo control circuit controls the spindle motor 51 so as to rotate the magneto-optical disk 1 at a predetermined rotation speed (for example, a constant linear speed). Further, the sled servo control circuit moves the optical head 53 and the magnetic head 54 to the target track position of the magneto-optical disk 1 designated by the system controller 57. The servo control circuit 56 that performs such various control operations sends information indicating the operating state of each unit controlled by the servo control circuit 56 to the system controller 57.

A key input operation unit 58 and a display unit 59 are connected to the system controller 57. The system controller 57 controls the recording system and the reproducing system in the operation mode designated by the operation input information from the key input operation unit 58. Further, the system controller 57 uses the sector head unit address information reproduced from the so-called header time recorded on the recording track of the magneto-optical disk 1 or the Q data of the sub-code, etc.
3 and the recording position and the reproducing position on the recording track traced by the magnetic head 54 are managed. Further, the system controller 57, based on the bit compression mode information in the ATC encoder 63 switched and selected by the key input operation unit 58 and the bit compression mode information in the reproduction data obtained from the RF circuit 55 via the reproduction system, The bit compression mode is displayed on the display unit 59, and the reproduction time is displayed on the display unit 59 based on the data compression rate in the bit compression mode and the reproduction position information on the recording track. This reproduction time display is performed by compressing the data in the bit compression mode with respect to address information (absolute time information) in sector units reproduced from the header time recorded on the recording track of the magneto-optical disc 1 and the subcode Q data. Reciprocal of rate (eg 1/4
In the case of compression, the actual time information is obtained by multiplying by 4), and this is displayed on the display unit 59. Even at the time of recording, for example, when absolute time information is previously recorded (pre-formatted) on a recording track of a magneto-optical disk or the like, the pre-formatted absolute time information is read to determine the data compression rate. It is also possible to display the current position at the actual recording time by multiplying by the reciprocal.

Next, in the recording system of the recording / reproducing system of this compressed data recording / reproducing apparatus, the analog audio input signal AIN from the input terminal 60 is supplied to the low-pass filter 61.
Is supplied to the A / D converter 62, and the A / D converter 62 quantizes (PCM) the analog audio input signal AIN. The digital audio signal obtained from the A / D converter 62 is an ATC (Adaptive Transform C
oding) is supplied to the encoder 63. Further, the digital audio input signal DIN from the input terminal 67 is supplied to the ATC encoder 63 via the digital input interface circuit 68. The ATC encoder 63 converts the analog audio input signal AIN into the A / D converter 6
A bit compression (data compression) process corresponding to various modes in the ATC system shown in Table 1 is performed on the digital audio signal quantized by 2 or the digital audio signal supplied through the digital input interface circuit 68 at a predetermined transfer rate. The operation mode is designated by the system controller 57. For example, in the B mode, compressed data (ATC audio data) having a sampling frequency of 44.1 kHz and a bit rate of 64 kbps is set in the memory 64.
Is supplied to. The data transfer rate in the B mode stereo mode is 1/8 (9.37) of the data transfer rate (75 sectors / second) of the standard CD-DA format.
5 sectors / second).

[0030]

[Table 1]

Here, in the embodiment of FIG. 1, A / D
The sampling frequency of the converter 62 is, for example, the sampling frequency of the standard CD-DA format described above.
The ATC encoder 63 is fixed at 4.1 kHz.
Also in, it is assumed that the sampling frequency is maintained and bit compression processing is performed. At this time: the signal pass band becomes narrower as the low bit rate mode is set, and accordingly, the cutoff frequency of the low pass filter 61 is also switched and controlled. That is, the cutoff frequency of the low-pass filter 61 of the A / D converter 62 is simultaneously switched and controlled according to the compression mode.

Next, the memory 64 is controlled by the system controller 57 to write and read data, and temporarily stores compressed audio data (hereinafter referred to as ATC audio data) supplied from the ATC encoder 63. It is used as a buffer memory for recording on the disk as needed. That is, for example, in the B mode stereo mode, the ATC supplied from the ATC encoder 63.
The audio data has its data transfer rate reduced to 1/8 of the standard CD-DA format data transfer rate (75 sectors / second), that is, 9.375 sectors / second. Memory 64
Continuously written to. As described above, it is sufficient to record one sector for every eight sectors of this ATC audio data. However, since recording every eight sectors is practically impossible, continuous sector recording as described later is performed. I am trying.

In this recording, a cluster consisting of a plurality of predetermined sectors (for example, 32 sectors + several sectors) is used as a recording unit for the same data transfer rate (75 sectors / second) as the standard CD-DA format through the pause period. ) Is done in a burst. That is, in the memory 64, the ATC audio data in the stereo mode is recorded in the B mode continuously written at the low transfer rate of 9.375 (= 75/8) sectors / second corresponding to the bit compression rate. The data is read in bursts at the transfer rate of 75 sectors / sec. The overall data transfer rate of the read and recorded data, including the recording pause period, is as low as 9.375 sectors / sec, but within the time of the recording operation performed in a burst manner. The instantaneous data transfer rate in the above is the standard 75 sectors / sec. Therefore, the disc rotation speed is a standard CD-D.
When the speed is the same as the A format (constant linear velocity), the C
The same recording density and storage pattern as those in the D-DA format are recorded.

The ATC audio data, that is, the recording data, which is burst-read from the memory 64 at the (instantaneous) transfer rate of 75 sectors / second is recorded by the encoder 65.
Is supplied to. Here, from the memory 64 to the encoder 65
In the data string supplied to the above, the unit of continuous recording in one recording is a cluster composed of a plurality of sectors (for example, 32 sectors) and several sectors for cluster connection arranged in the front and rear positions of the cluster. This cluster connection sector is set longer than the interleave length in the encoder 65 so that the data of other clusters will not be affected even if interleaved.

The encoder 65, regarding the recording data supplied from the memory 64 in bursts as described above,
Encoding processing (parity addition and interleave processing) for error correction, EFM processing, and the like are performed. The recording data encoded by the encoder 65 is supplied to the magnetic head drive circuit 66. The magnetic head drive circuit 66 is connected to the magnetic head 54, and drives the magnetic head 54 so as to apply the modulation magnetic field according to the recording data to the magneto-optical disk 1. Further, the system controller 57 performs the above-mentioned memory control on the memory 64, and continuously records the above-mentioned recording data burst-read from the memory 64 on the recording track of the magneto-optical disk 1 by this memory control. Controls the recording position. The recording position is controlled by controlling the recording position of the recording data which is burst-read from the memory 64 by the system controller 57 and outputting a control signal for designating the recording position on the recording track of the magneto-optical disk 1 to the servo control circuit. By feeding 56. Next, the reproducing system of this magneto-optical disk recording / reproducing unit will be described.

This reproducing system is for reproducing the record data continuously recorded on the recording tracks of the magneto-optical disk 1 by the above-mentioned recording system, and the recording track of the magneto-optical disk 1 is made by the optical head 53. The reproduction output obtained by tracing the laser beam with the laser light is the RF circuit 55.
It is provided with a decoder 71 which is binarized by and supplied. In this reproducing system, not only a magneto-optical disc but also a so-called compact disc (CD: Compact Disc)
The same read-only optical disk can be read.

The decoder 71 corresponds to the encoder 65 in the recording system described above, and performs decoding processing (deinterleave processing or error correction) for error correction on the reproduction output binarized by the RF circuit 55. processing)
ATC audio data in the B mode stereo mode described above is reproduced at a transfer rate of 75 sectors / sec which is faster than the normal transfer rate in the B mode stereo mode by performing processing such as EDM and EFM demodulation processing. To do. The reproduced data obtained by the decoder 71 is supplied to the memory 72.

In the memory 72, writing and reading of data are controlled by the system controller 57, and reproduced data supplied from the decoder 71 at a transfer rate of 75 sectors / second is written in bursts at the transfer rate of 75 sectors / second. Be done. The memory 72 has a normal 9.37 in the stereo mode in which the reproduction data written in burst at the transfer rate of 75 sectors / second is in the B mode.
It is continuously read at a transfer rate of 5 sectors / second.

The system controller 57 writes the reproduced data in the memory 72 at a transfer rate of 75 sectors / second, and also writes the reproduced data from the memory 72 in the above 9.37.
Memory control is performed so that data is continuously read at a transfer rate of 5 sectors / second. Further, the system controller 57 performs the above-mentioned memory control on the memory 72, and continuously reproduces the above-mentioned reproduction data written in a burst in the memory 72 from the recording track of the magneto-optical disk 1 by this memory control. Controls the playback position. This playback position is controlled by the system controller 57.
Manages the reproduction position of the reproduction data read out in burst from the magneto-optical disc 1 by the servo controller, and outputs a control signal from the system controller 57 for designating the reproduction position on the recording track of the magneto-optical disc 1 or the optical disc 1. By feeding 56.

ATC audio data in the B mode stereo mode obtained as reproduction data continuously read from the memory 72 at a transfer rate of 9.375 sectors / second is supplied to the ATC decoder 73. This AT
The C decoder 73 is a recording system ATC encoder 63.
The operation mode is designated by the system controller 57. For example, 16-bit digital audio data is expanded by 8-fold data expansion (bit expansion) of ATC audio data in the B mode stereo mode. Reproduce. This ATC decoder 73
Digital audio data from the D / A converter 7
4 is supplied.

The D / A converter 74 is the ATC decoder 73.
The digital audio data supplied from the converter is converted into an analog signal to form an analog audio output signal AOUT. The analog audio output signal AOUT obtained by the D / A converter 74 is a low-pass filter 75.
Is output from the output terminal 76 via.

Next, the IC card recording unit of this compressed data recording / reproducing apparatus will be described.

The ATC audio data from the decoder 71 is sent to the additional compressor 84 and subjected to processing such as removal of surplus bits and zero word length processing.

Here, in this embodiment, the spectral component which is significantly smaller than the maximum value in the block for the block floating is set to zero. This process is executed while reading / writing data from / to the memory 85. The variable bit rate compression-encoded data from the additional compressor 84, which removes excess bits and performs zero word length processing, is
It is written in the IC card 2 via the card interface circuit 86. Of course, in the present invention, variable bit rate compression such as removal of excess bits and zero word length processing is not performed, but a block for increasing the orthogonal transform size or for block floating on the frequency axis having sub information is used. And / or writing may be performed at a constant bit rate of a lower bit rate by widening the frequency width of a block in which quantization noise occurs.

The compressed data (ATC audio data) from the reproducing system decoder 71 of the magneto-optical disk recording / reproducing unit is directly sent to the memory 85 of the IC card recording unit without being expanded.
This data transfer is performed by the system controller 57 controlling the memory 85 and the like during so-called high-speed dubbing. Writing ATC audio data having a low bit rate from the magneto-optical disk or the optical disk to the IC card 2 is suitable when using an IC card having a high price per recording capacity. The compressed data from the memory 72 may be sent to the memory 85.

Here, a so-called high-speed digital dubbing operation will be described.

At the time of high-speed digital dubbing, the system controller 57 executes the high-speed dubbing control processing operation by operating the dubbing operation keys of the key input operation unit 58. Specifically, the compressed data from the decoder 71 is sent to the memory 85 of the IC card recording unit as it is, and variable bit rate coding is performed by the additional compressor 84 that performs processing such as removal of excess bits and zero word length processing. Then, the data is written to the IC card 2 via the IC card interface circuit 86. If the ATC audio data in the B mode stereo mode is recorded on the magneto-optical disk 1, for example, the decoder 7
From 1, the digital audio data compressed by 8 times is continuously read.

Therefore, at the time of high-speed dubbing, compressed data corresponding to a time which is eight times (in the case of the B mode stereo mode) in real time is continuously obtained from the magneto-optical disk 1. Since excess bits are removed and processing such as zero word length processing is performed on the data and the data having a constant bit rate is written to the IC card 2, eight times faster dubbing can be realized. If the compression mode is different, the magnification of the dubbing speed is also different. Also, dubbing may be performed at a high speed equal to or higher than the compression ratio. In this case, the magneto-optical disk 1 is driven to rotate at high speed at a speed several times the steady speed.

By the way, as shown in FIG. 2, bit-compressed data is recorded on the magneto-optical disk 1, and at the same time, the additional compression / expansion block 3 compresses the data by variable bit rate coding. Information on the amount of data when encoded (that is, the data recording capacity required for writing to the IC card 2) is recorded. By doing so, for example, the number of songs writable in the IC card 2 and the combination of the songs among the songs recorded on the magneto-optical disk 1 can be immediately known by reading the data amount information. . Of course, instead of the variable bit rate mode, conversion to a fixed bit rate lower bit rate mode can be performed by the additional compression / expansion block 3.

On the contrary, by writing not only the data bit-compressed and encoded by the variable bit rate encoding but also the data amount information of the bit-compressed and encoded data in the IC card 2, the IC card 2 It is possible to quickly know the amount of data when transmitting and recording data such as music to the magneto-optical disk 1. Of course, not only the data bit-compressed and encoded by the variable bit rate encoding but also the data in the low bit rate mode of the fixed bit rate can be written in the IC card 2.

Here, FIG. 3 shows a front appearance of the compressed data recording / reproducing apparatus 5 having the structure shown in FIG. 1, in which an inserting portion 6 for a magneto-optical disk or an optical disk and an IC card inserting slot 7 are provided. ing. Of course, the magneto-optical disk recording / reproducing unit and the IC card recording unit may be set separately and may be connected by a cable.

Next, the high efficiency coding in the ATC encoder 63 will be described in detail. That is, a technique of high efficiency encoding of an input digital signal such as a digital audio signal using band division encoding (SBC), adaptive transform encoding (ATC) and adaptive bit allocation techniques is described.
The description will be made with reference to the drawings starting from FIG.

The ATC encoder 63 (hereinafter, referred to as a high efficiency coding device) for specifically realizing the high efficiency coding processing in the digital signal processing method of the present invention divides the input digital signal into a plurality of frequency bands. , The bandwidths of the two lowest bands are the same, and in the higher frequency band, the higher the frequency band, the wider the bandwidth is selected, and the orthogonal transform is performed for each frequency band to obtain the spectrum data of the obtained frequency axis. In the low frequency range, each so-called critical bandwidth (critical band) that takes into account human auditory characteristics, which will be described later, is adaptively adjusted, and in the middle and high frequencies, the critical bandwidth is subdivided in consideration of block floating efficiency. Bits are allocated and encoded. Usually, the block for the above-mentioned orthogonal transformation is a unit in which quantization noise occurs. Further, in the present embodiment, the block size (block length) is adaptively changed according to the input digital audio signal before the orthogonal transformation, and the floating process is performed for each block.

Specifically, referring to FIG. 4, the input terminal 10
For example, when the sampling frequency is 44.1 kHz,
A digital audio signal of 0 to 22 kHz is supplied. This input digital audio signal is supplied to a band division filter 11 composed of a filter such as a so-called QMF (Quadrature Mirror filter) for 0 to 11 kHz.
Band and 11 to 22 kHz band divided into 0 to 11
A signal in the kHz band is also supplied to a band-dividing filter 12 which is also a so-called QMF filter or the like and has a frequency of 0 to 5.5 kHz.
It is divided into a band and a band of 5.5 to 11 kHz. A signal in the 11 kHz to 22 kHz band from the band division filter 11 is sent to the MDCT circuit 13 which is an example of an orthogonal transformation circuit, and 5.5 kHz to 11 kHz from the band division filter 12.
The band signal is sent to the MDCT circuit 14, and the 0 to 5.5 kHz band signal from the band division filter 12 is MDCT.
The MDCT processing is performed by being sent to the circuit 15.

As a method of dividing the above-mentioned input digital signal into a plurality of frequency bands, for example, the above QM is used.
There is a division method using a filter such as F. This segmentation method is described in the document "Digital coding of speech in subba".
nds "RE Crochiere, Bell Syst.Tech. J., Vol.55, N
o.8 1976).

In addition, the document "Polyphase Quadrature Filters-New Band Division Coding Technology"("Po
lyphase Quadrature filters -A new subband coding t
echnique ", Joseph H. Rothweiler ICASSP 83, BOSTON)
Describes an equal bandwidth filter partitioning technique.

As the above-mentioned orthogonal transform, for example, the input audio signal is divided into blocks in a predetermined unit time, and the fast Fourier transform (FFT), discrete cosine transform (DCT), and modified discrete cosine transform (MDC) are performed for each block.
There is an orthogonal transformation in which the time axis is transformed into the frequency axis by performing T) or the like. Regarding the above-mentioned MDCT, refer to the document "Filters based on time domain aliasing cancellation.
Subband / Transform Coding Using Bank Design "(" Subban
d / Transform Coding Using Filter Bank Designs Based
on Time Domain Aliasing Cancellation, "JPPrincen
ABBradley, Univ. Of Surrey Royal Melbourne Ins
t. of Tech. ICASSP 1987).

Next, the MDCT circuits 13, 1 in each mode for a standard input digital audio signal.
FIG. 5 shows a specific example of the blocks at 4 and 15.

In the specific example of FIG. 5, the three filter output signals from the band division filters 11 and 12 of FIG. 4 each have M orthogonal transform block sizes.
The DCT circuits 13, 14 and 15 can switch the time resolution depending on the time characteristic of the signal. Also MD
The CT circuits 13, 14 and 15 increase the time length of the maximum processing block and narrow the signal pass bandwidth in the mode in which the bit rate is lower.

That is, in this embodiment, in the A mode, the orthogonal transform block size is increased to 11.6 ms when the signal is quasi-stationary in time, and shown in FIG. 5 when the signal is non-stationary. In the band of 11 kHz or less, the orthogonal transform block size is further divided into 4
In the band of Hz or higher, the orthogonal transform block size is divided into eight.

In the B mode, the time length of the orthogonal transform block is twice as long as that in the A mode, which is 23.2 ms, and the signal pass bandwidth is narrowed to 13 kHz. When the signal is quasi-stationary in time, the orthogonal transform block size is increased to 23.2 ms, and when the signal is non-stationary, it is divided into 2 to 11.6 ms. Further, when the non-stationarity of the signal becomes stronger, as shown in FIG. 5, the orthogonal transform block size is further divided into four in the band of 11 kHz or less to be a total of eight, and the orthogonal transform block size is further increased in the band of 11 kHz or more. Eight divided into a total of 16 divided.

In the C mode, the time length of the orthogonal transform block is set to 34.8 ms. Pass band is 5.5k
Limit to Hz.

In the D mode, the time length of the orthogonal transform block is set to 46.4 ms.

Here, each MDCT circuit 13, 14, 15
In the above, the bit length conversion from the A mode to the B mode is facilitated by limiting the time length of the orthogonal transform block to be twice as long as the low frequency band. That is,
The signal orthogonally transformed on the low frequency side of the A mode is inversely orthogonally transformed,
The obtained signal is subjected to orthogonal transform with a double orthogonal transform block size. This is easier than performing inverse orthogonal transform on signals in a plurality of bands forming the entire band and then performing orthogonal transform again for each band. Further, this is convenient for executing high-speed transfer from the magneto-optical disk to the IC memory card while converting from the A mode to the B mode. This is based on the fact that the acoustic signal in the high frequency range has a larger temporal variation and the signal-to-noise ratio may be smaller than that in the low frequency range.

At this time, the signal pass bandwidth is 13
Up to kHz. In this case, 11 kHz to 22 kHz
By sub-sampling the filter output signal before orthogonal transformation in the z-band signal by ½ or ¼, it is possible to avoid unnecessary signal processing for a band equal to or higher than the signal pass band.

In the following, the length of the orthogonal transform block becomes longer and the signal pass bandwidth can be made narrower in the C mode and the D mode. Of course, the length of the orthogonal transform block and the signal pass bandwidth do not have to be different between all modes, and the same value may be used in some cases.

Even if the length of the orthogonal transform block is longer in the low bit rate mode, for the purpose of reducing the time delay, the orthogonal transform block size among the plural orthogonal transform block sizes is included in the mode. The object can be achieved by selectively using a short orthogonal transform block size to perform orthogonal transform.

Referring again to FIG. 4, each M in the A mode
In the spectrum data (spectral components) on the frequency axis or the MDCT coefficient data obtained by the MDCT processing in the DCT circuits 13, 14 and 15, low frequencies are grouped into so-called critical bands (critical bands), and medium high The region is divided into the critical bandwidth in consideration of the effectiveness of the block floating, and the nonlinear processing circuit 40, which will be described later,
After passing through 41 and 42, the adaptive bit allocation encoding circuit 18
Have been sent to. In addition, with this critical band,
It is a frequency band divided in consideration of human auditory characteristics, and is a band of noise when a pure tone is masked by a narrow band noise of the same strength near the frequency of the pure tone. The critical band has a wider bandwidth in a higher frequency range, and the entire frequency band of 0 to 22 kHz is divided into, for example, 25 critical bands.

In the B mode, when the signal that does not double the orthogonal transform block size in the A mode is non-stationary, the frequency width of the block having the sub information is
For example, by adopting a frequency width twice as wide as that of the A mode, the number of blocks is halved and sub information is reduced. In this way, the orthogonal transform block size is doubled in the low band, and the frequency width of the block having sub information is increased in the other bands, thereby reducing the sub information in the entire band. You can

Next, the bit allocation calculating circuit 43 calculates the energy or peak value of each divided band in consideration of the critical band and the block floating based on the spectrum data divided in consideration of the critical band and the block floating. Furthermore, the number of allocated bits is calculated for each band based on the energy or peak value of each divided band in consideration of the masking amount, and this information is sent to the adaptive bit allocation encoding circuit 18. The adaptive bit allocation encoding circuit 18 normalizes and quantizes each spectrum data (or MDCT coefficient data) according to the number of bits allocated for each band. The data encoded in this way is taken out via the output terminal 19.

Next, FIG. 6 shows the bit allocation calculating circuit 43.
It is a block circuit diagram which shows schematic structure of one specific example. In FIG. 6, the input terminal 21 is supplied with spectrum data on the frequency axis from each of the nonlinear processing circuits 40, 41 and 42.

Next, this spectrum data on the frequency axis is sent to the energy calculation circuit 22 for each band, and the energy of each divided band considering the critical band and block floating is, for example, each spectrum component in the band. It is obtained by calculating the sum of the amplitude values of. Instead of the energy for each band, a peak value, an average value, or the like of the amplitude value may be used. As the output from the energy calculating circuit 22, for example, the spectrum of the sum value of each band is shown as SB in the drawing of FIG. However, in FIG. 7, in order to simplify the illustration,
Considering the masking amount, critical band and block floating, the number of divided bands is 12 bands (B1
~ B12).

Here, in order to take into account the influence of the spectrum SB on so-called masking, a convolution process is performed such that the spectrum SB is multiplied by a predetermined weighting function and added. Therefore, the output of the energy calculation circuit 22 for each band, that is, each value of the spectrum SB is sent to the convolution filter circuit 23. The convolution filter circuit 23 includes, for example, a plurality of delay elements for sequentially delaying input data and a plurality of multipliers (for example, 25 corresponding to each band) for multiplying outputs from these delay elements by a multiplication coefficient (weighting function). Multipliers)
And a sum total adder that sums the outputs of the respective multipliers. By this convolution processing, the sum total of the portion indicated by the dotted line in FIG. 7 is obtained. Note that the masking is a phenomenon in which one signal is masked by another signal and becomes inaudible due to human auditory characteristics, and this masking effect includes a time axis masking by an audio signal on the time axis. There are an effect and a simultaneous masking effect by a signal on the frequency axis. Due to these masking effects, even if there is noise in the masked portion, this noise cannot be heard. Therefore, in the actual audio signal, the noise within the masked range is regarded as an acceptable noise.

Here, a specific example of the multiplication coefficient (filter coefficient) of each multiplier of the convolution filter circuit 23 will be described. When the coefficient of the multiplier M corresponding to an arbitrary band is 1, the multiplier M-1 gives a coefficient of 0.15, multiplier M-2 gives a coefficient of 0.0019, and multiplier M-3 gives a coefficient of 0.0000.
086, multiplier M + 1 gives a coefficient of 0.4, multiplier M + 2
By multiplying the output of each delay element by a coefficient of 0.06 with a coefficient of 0.007 with a multiplier M + 3, the convolution processing of the spectrum SB is performed. However, M is an arbitrary integer of 1 to 25.

Next, the output of the convolution filter circuit 23 is sent to the subtractor 24. The subtractor 24 obtains a level α corresponding to an allowable noise level described later in the convoluted area. The level α corresponding to the permissible noise level (hereinafter referred to as the permissible noise level) becomes the permissible noise level for each band of the critical band by performing the inverse convolution process as described later. It is a good level. Here, the subtractor 24 is supplied with an allowance function (function expressing a masking level) for obtaining the level α. The level α is controlled by increasing or decreasing this allowance function. The permissible function is supplied from the (n-ai) function generating circuit 25 described below.

That is, the level α corresponding to the allowable noise level can be obtained by the following equation (1), where i is the number given in order from the low band of the critical band.

Α = S- (n-ai) (1) In this equation (1), n and a are constants, a> 0, S is the intensity of the convolved Bark spectrum, and ( 1)
In the formula, (n-ai) is the allowable function. In this embodiment, n = 38,
Since a = 1, there was no sound quality deterioration at this time, and good encoding was possible.

In this way, the level α is obtained, and this data is transmitted to the divider 26. The divider 26 is for inverse convolution of the level α in the convolved area. Therefore, by performing this inverse convolution process,
The masking spectrum can be obtained from the level α. That is, this masking spectrum becomes the allowable noise spectrum. Although the above-mentioned inverse convolution processing requires a complicated operation, in the present embodiment, the inverse convolution is performed using the simplified divider 26.

Next, the masking spectrum is transmitted to the subtractor 28 via the synthesizing circuit 27. Here, the subtractor 28 includes an energy detection circuit 22 for each band.
From the output signal, that is, the spectrum SB described above is supplied via the delay circuit 29. Therefore, the masking spectrum and the spectrum SB are obtained by the subtractor 28.
By performing the subtraction operation with and, the spectrum SB is masked below the level indicated by the level of the masking spectrum MS, as shown in FIG.
Hereinafter, the output from the subtractor 28, in which the masked spectrum SB is referred to as an allowable noise level, is taken out from the output terminal 31 via the allowable noise correction circuit 30, and, for example, the ROM or the like in which the allocated bit number information is stored in advance (see FIG. Sent (not shown). This ROM, etc., according to the output obtained from the subtraction circuit 28 via the allowable noise correction circuit 30,
Information on the number of allocated bits for each band is output. This allocation bit number information is the adaptive bit allocation encoding circuit 1 described above.
By being sent to the channel 8, the spectrum data on the frequency axis from the MDCT circuits 13, 14, and 15 are quantized by the number of bits assigned to each band.

In summary, the adaptive bit allocation encoding circuit 18 is to quantize the spectrum data for each band with the number of bits allocated according to the energy of each divided band in consideration of the masking amount. Become.
The delay circuit 29 is provided to delay the spectrum SB from the energy detection circuit 22 in consideration of the delay amount in each circuit before the synthesis circuit 27.

By the way, at the time of synthesizing in the synthesizing circuit 27 described above, data indicating a so-called minimum audible curve RC which is a human auditory characteristic as shown in FIG. The masking spectrum MS may be combined. In this minimum audible curve RC, if the absolute level of noise is less than or equal to this minimum audible curve RC, the noise cannot be heard. Even if the coding is the same, the minimum audible curve RC is different due to the difference in the reproduction volume at the time of reproduction. However, in a realistic digital system, for example, how to enter music into a 16-bit dynamic range is not so much. Since there is no difference, if, for example, the quantization noise in the most audible frequency band near 4 kHz is not heard, it is considered that the quantization noise below the level of the minimum audible curve is not heard in other frequency bands.

Therefore, for example, assuming that the noise of around 4 kHz cannot be heard, the minimum audible curve RC
And the masking spectrum MS are combined together to obtain an allowable noise level. In this case, the allowable noise level can be up to the shaded portion in the drawing of FIG. In this embodiment, the minimum audible curve of 4k
The Hz level is adjusted to the lowest level equivalent to 20 bits, for example. Further, FIG. 9 shows the signal spectrum SS
Are also shown at the same time.

The allowable noise correction circuit 30 corrects the allowable noise level in the output from the subtractor 28 based on the information of the equal loudness curve sent from the correction information output circuit 33. Here, the equal loudness curve is a characteristic curve relating to human auditory characteristics, for example, a curve obtained by obtaining the sound pressure of sound at each frequency that sounds the same as a pure tone of 1 kHz, and connecting the curves. Also called sensitivity curve. Further, this equal loudness curve draws a curve substantially the same as the minimum audible curve RC shown in FIG. In this equal loudness curve, for example, in the vicinity of 4 kHz, even if the sound pressure drops by 8 to 10 dB from 1 kHz, it sounds as loud as 1 kHz. It doesn't sound the same. Therefore, it is preferable that the allowable noise correction circuit 30 has an allowable noise level having the same frequency characteristic as the equal loudness curve. From this, it can be seen that correcting the permissible noise level in consideration of the equal loudness curve is suitable for human auditory characteristics.

Further, in the correction information output circuit 33, the error between the output information amount (data amount) at the time of quantization in the adaptive bit allocation encoding circuit 18 and the bit rate target value of the final encoded data. Correction data for correcting the allowable noise level is output on the basis of the information of. This is because the total number of bits obtained by performing temporary adaptive bit allocation to all bit allocation unit blocks in advance is a fixed number of bits (target value) determined by the bit rate of the final encoded output data. May have an error, and bit allocation is performed again so that the error becomes 0. That is, when the total number of allocated bits is smaller than the target value, the difference bit number is allocated and added to each unit block, and when the total number of allocated bits is larger than the target value, the difference bit number is set in each unit. Allocate to blocks and delete.

Specifically, an error of the total allocated bit number from the target value is detected, and the correction information output circuit 33 outputs correction data for correcting each allocated bit number according to the error data. . Here, the case where the error data indicates a shortage of the number of bits is a case where the number of bits per unit block is used and the amount of data is larger than the target value. Further, when the error data is data indicating the remainder of the number of bits, the number of bits per unit block is small, and the amount of data is smaller than the target value. Therefore, from the correction information output circuit 33,
The correction data for correcting the allowable noise level from the subtractor 28 according to the error data is output based on, for example, the information data of the equal loudness curve. By transmitting the correction data as described above to the allowable noise correction circuit 30, the allowable noise level from the subtractor 28 is corrected.

In the high-efficiency coding apparatus described above, the quantized spectrum data is output as the main information, and the scale factor indicating the block floating state and the word length indicating the word length are output as the sub information. To be done. Of course, the word length information is not essential, and can be obtained from the scale factor information in the ATC decoder 73.

Here, the bit allocation calculation circuit 43
It is also possible to adopt a configuration as shown in FIG. The following effective bit allocation method different from the bit allocation method described above will be described with reference to FIG.

Each non-linear processing circuit 40 in FIG.
The outputs of 41 and 42 are energy calculation circuit 30 for calculating energy for each band via the input terminal 301 of FIG.
Sent to 3. In the energy calculation circuit 303 for each band, the energy for each band obtained by further dividing the critical band (critical band) or the critical band in the high band is calculated, for example, the square root of the root mean square of each amplitude value in the band. It is required by doing. Instead of the energy for each band, the peak value or average value of the amplitude values may be used. Further, the output from the energy calculation circuit 303 may be, for example, the Bark spectrum SB which is the sum of the spectral components of each band obtained by further dividing the critical band shown in FIG. 7 or the critical band in the high range.

In this embodiment, the number of bits that can be used for transmitting or recording MDCT coefficients is 100, for example.
If Kbps is set, a fixed bit allocation pattern using the 100 Kbps is created in this embodiment. In this embodiment, a plurality of bit allocation patterns for the fixed bit allocation are prepared, and the pattern can be selected according to the characteristics of the signal. In the present embodiment, the fixed bit allocation circuit 305 has various patterns in which the bit amount of the short time corresponding to 100 Kbps is distributed to each frequency. The fixed bit allocation circuit 305 has a plurality of patterns in which the bit allocation ratios for the middle low band and the high band are different. Then, as the signal size is smaller, a pattern with a smaller amount of allocation to the high frequency band is selected. By doing so, the loudness effect in which the sensitivity in the high frequency range decreases as the signal becomes smaller can be used. In addition, as the magnitude of the signal at this time,
It is possible to use the signal magnitudes in all bands, but it is also possible to use, for example, a filter output such as QMF or an output obtained by MDCT processing. The number of bits that can be used to transmit or record the MDCT coefficient (the number of usable bits, 100 Kbps) is set by, for example, the total available bit number output circuit 302. The total number of usable bits is
It is also possible to input from the outside.

The energy-dependent bit allocation is performed by weighting the dB value of energy for a short time corresponding to 100 Kbps by multiplying a predetermined coefficient for each block, and the value thus obtained. Is performed in proportion to. Here, by setting the weighting coefficient so that it has a large value in the low frequency band, more bits are allocated to the low frequency band. The energy-dependent bit allocation is performed by the energy-dependent bit allocation circuit 304 to which the output of the energy calculation circuit 303 is supplied.

That is, in the energy-dependent bit allocation circuit 304, a plurality of patterns of weighting coefficients are prepared in the same manner as the fixed bit allocation circuit 305, and the plurality of patterns can be switched by the input signal, or, for example, two weights can be weighted. An energy-dependent bit allocation is calculated using a weighting pattern obtained by interpolating a pattern of coefficients with an input signal. As described above, in the present embodiment, by changing the weighting coefficient according to the input signal, it becomes possible to perform bit allocation more suited to the sense of hearing and improve the sound quality.

In FIG. 10, the division ratio between the above-mentioned allocation to the fixed bit allocation pattern and the bit allocation depending on, for example, the Bark spectrum (spectrum SB) is determined by the index indicating the smoothness of the signal spectrum. . That is, in the present embodiment, the output of the energy calculation circuit 303 is sent to the spectrum smoothness calculation circuit 308, and the spectrum smoothness calculation circuit 308
A value obtained by dividing the sum of the absolute values of the differences between adjacent values of the signal spectrum data by the sum of the signal spectrum data is calculated as an index, and this index is sent to the bit division rate determination circuit 309 for obtaining the division rate of the bit allocation. .

The division ratio data from the bit division ratio determining circuit 309 is supplied to the multiplier 312 to which the output of the fixed bit allocation circuit 305 is supplied and the multiplier 311 to which the output of the energy dependent bit distribution circuit 304 is supplied. Sent to. The outputs of these multipliers 312 and 311 are sent to the sum calculation circuit 306. That is, the sum of the fixed bit allocation and the critical band (critical band) for each band or the bit allocation value depending on the energy of the spectrum for each band obtained by further dividing the critical band in the high band is calculated by the sum calculation circuit 306. Then, the calculation result is sent from the output terminal (bit allocation amount output terminal of each band) 307 to the adaptive bit allocation encoding circuit 18 and used for quantization.

The state of bit allocation at this time is shown in FIGS. The states of quantization noise corresponding to this are shown in FIGS. Note that FIGS. 11 and 12 show the case where the spectrum of the signal is flat, and FIGS.
4 shows the case where the signal spectrum shows high tonality. Further, Q _{S in} the diagrams of FIGS. 11 and 13 indicates the bit amount for energy dependence, and Q _F in the diagrams indicates the bit amount for fixed bit allocation. 12 and 14, L represents a signal level, N _S represents a noise reduction amount due to energy dependence, and N _F represents a noise level due to fixed bit allocation in the diagrams.

In FIGS. 11 and 12, which show the case where the spectrum of the above signal is proportionally flat, bit allocation with a large amount of fixed bit allocation is usually useful for obtaining a large signal to noise ratio over the entire band. .
However, in the case of FIGS. 11 and 12, a relatively small number of bit allocations are used in the low band and the high band. This is because this band is acoustically less important. Further, at this time, as shown by Q _S in the drawing of FIG. 11, the noise level in the band in which the signal magnitude is large is selectively lowered by the amount (bits) that is slightly energy-dependent bit allocation. Therefore, if the spectrum of the signal is relatively flat, this selectivity also works over a relatively wide band.

On the other hand, when the signal spectrum exhibits a high tonality as shown in FIGS. 12 and 14, a large amount of energy-dependent bit allocation is performed as indicated by Q _S in the diagram of FIG. Quantization noise reduction by (bits) is used to reduce noise in a very narrow band (band indicated by N _S in the drawing of FIG. 14). As a result, the improvement of the characteristics of the quantization noise with respect to the input signal having the isolated spectrum component is achieved. At the same time, a bit is allocated by a bit of fixed bit allocation (bit)
As a result, the noise level in a wide band is reduced non-selectively.

The block selection circuit 20 detects a block for which a sufficient signal-to-noise ratio cannot be obtained, and the non-linear processing circuit 4
Numerals 0, 41 and 42 perform the following non-linear signal processing on the blocks detected by the block selection circuit 20 to reduce the quantization noise. That is, the spectrum data on the frequency axis, which is the MDCT transform output, has a smaller spectrum data size than the maximum spectrum data for each divided band considering the masking amount, the critical band, and the block floating. Or a conversion process to zero is performed.

This will be described with reference to FIG.

FIG. 15 shows the case where five spectrum data (components) are present in each frequency block ni (i is an integer) like the frequency blocks n and n + 1 for floating a certain block. There is. In the case of the frequency block n, the size of each spectrum component is similar, so that the signal-to-noise ratio of each spectrum component when quantized with a common word length for the block floatin and each spectrum component. Are substantially the same, and even if the block floating information and the word length information common to the spectrum components in the frequency block n are used, a high signal-to-noise ratio can be efficiently given to each spectrum component.

On the other hand, in the case of the frequency block n + 1, the size of each spectral component is not similar, and in particular, when the few spectral components are by far larger than many other spectral components, it is sufficient. The number of spectral components from which the signal-to-noise ratio can be obtained is small. The remaining large number of spectral components will have a significantly lower signal to noise ratio. In this case, a masking effect by a spectral component having a high level is expected to be expected, but it is known that such a masking effect by an isolated spectral component is significantly smaller than a masking effect by a noise component. As a result, the spectrum component having a small signal-to-noise ratio causes deterioration of the sound quality as a whole.

According to the present invention, the effect of masking is judged for such a spectral component in which a large signal-to-noise ratio cannot be obtained, and if masking is difficult,
Spectral components for which a large signal-to-noise ratio cannot be obtained are assigned zero bits so that quantization noise does not occur, and the quantized value becomes zero, or the signal-to-noise ratio is set when bit allocation is performed. The adaptive bit allocation encoding circuit 18 performs normalization and quantization processing after transforming the spectral component so that it becomes larger.

The operation of the non-linear processing circuits 40, 41 and 42 will be described with reference to FIG. The frequency block n + 1
Are selected as blocks to be subjected to nonlinear processing by the block selection circuit 20.

In FIG. 16, the spectral component A of the frequency block n + 1 for the block floating,
Consider B, C, D, and E. In this case, since the spectral component B gives the maximum value, the level of normalization is determined by this spectral component B.

Next, the first comparison level is set as a level that is approximately 12 dB lower than the normalization level, and the second comparison level is set as a level that is 18 dB lower. Then, with respect to the spectrum component of the level between the first comparison level and the second comparison level, the size of the spectrum component is increased in order to increase the signal-to-noise ratio. As a method of increasing the magnitude of the spectral component, the level is set to 6 dB lower than the normalization level.

As shown in FIG. 16, at this time, since the spectral component A has a value smaller than the second comparison level, it is further reduced so that the quantized output becomes zero like the spectral component A '. It Since the spectral component B has the maximum value, it is not changed at all. Since the spectral component C is between the first comparison level and the second comparison level, the spectral component C is increased to change like the spectral component C ′. Hereinafter, since the spectral components D and E are smaller than the second comparison level, they are reduced and the quantized output becomes zero. Although not shown in FIG. 16, no particular processing is performed on the spectrum component larger than the first comparison level, since the signal-to-noise ratio can be sufficiently obtained even with the value as it is.

At this time, as another method, the first and second
It is also possible to make the comparison level of the variable according to the value of the maximum spectral component in the frequency block. As the method, the first comparison level decreases as the value of the maximum spectral component in the frequency block increases, or the second comparison level increases as the value of the maximum spectral component in the frequency block increases. Let them do it.
Further, the first comparison level may decrease and the second comparison level may increase as the value of the maximum spectral component in the frequency block increases. In this way, by making the first and / or second comparison level variable according to the value of the maximum spectral component in the frequency block, it is possible to make a selection more suitable for hearing. Further, although the change in sound quality becomes large, each spectrum component may be set to a smaller value so that the quantized value becomes zero for all spectrum components having a value smaller than the first comparison level. .

In the block selection circuit 20, as a method of determining whether or not the above-described nonlinear processing is performed in the frequency block, the block selection circuit 20 has the same configuration as that of the bit allocation calculation circuit 43 described above, and the MDCT circuit 13 is used. , 14 and 15 may be used to calculate the provisional bit allocation and selection may be made based on the word length of each frequency block determined by the provisional bit allocation. Specifically, only the frequency block having a quantization noise level of 24 dB or less from the normalization level, that is, the frequency block having a word length of 4 bits or less is subjected to the non-linear processing.

In the block selection circuit 20, another method of determining whether or not the above-mentioned nonlinear processing is performed in the frequency block is a method of using the tonality of each frequency block. For example, a method is used in which the ratio between the effective value of at least one spectral component and the effective value of the remaining spectral components is determined as the tonality from the largest spectral component, and a determination method is used.

Here, in the present embodiment, as the ratio of the effective value at the time of this determination, the effective value of the spectral component having the maximum magnitude of the spectral component, that is, the effective value of the spectral component having the maximum signal-to-noise ratio and the remaining When the ratio of the spectrum component to the effective value is 10 dB or more and when the value of the maximum spectrum component in the frequency block is equal to or higher than a certain level, the frequency block is selected as the frequency block to be subjected to the non-linear processing. In this embodiment, −40 from the peak level
dB is at this level. As a result, it is possible to avoid unnecessary processing with a low-level signal, which is less likely to cause a sense of discomfort in terms of hearing.

Further, the frequency band in which such non-linear processing is performed can be limited to a specific frequency band. Especially, by limiting the band in which the non-linear processing is performed to the high frequency band, the change in the sound quality can be minimized. After such non-linear processing is performed, the actual bit allocation is executed by the bit allocation calculation circuit 43. The final bit allocation is determined in consideration of the spectral component increased by the non-linear processing and the spectral component zeroed.

As described above, in the present embodiment, among the signal components excluding the maximum value in the block, the component having a large value is subjected to the non-linear processing so as to increase the value, so that the signal pair The masking effect can be increased by increasing the noise ratio.

Further, among the signal components excluding the maximum value in the block, the component having a small value is subjected to the non-linear processing such that the value is made smaller so that the quantized value becomes zero. It is possible to prevent noise from being generated from a signal having a low signal-to-noise ratio.

Further, by making only the blocks whose word lengths determined by the tentative bit allocation are less than or equal to a certain length as the processing targets of the above-mentioned non-linear processing, it is possible to minimize the sound quality deterioration.

Furthermore, by selecting the block for performing the above-mentioned non-linear processing based on the tonality of each block, it is possible to process only the necessary blocks and minimize the change in sound quality. You can In addition, the tonality at this time is defined as the component having at least the maximum signal-to-noise ratio among the signal components in the block,
By obtaining the value obtained from the intra-block signal component excluding the component, for example, the ratio of the effective value of each component, it is possible to select only the block for which the masking effect cannot be expected from the viewpoint of hearing.

The present invention is not limited to the above-mentioned embodiments, and is applicable not only to digital audio signals, but also to signal processing devices such as digital audio (speech) signals and digital video signals. . Further, the above-described minimum audible curve synthesizing process may not be performed. In this case, the minimum audible curve generating circuit 32 and the synthesizing circuit 27 are unnecessary, and the output from the subtractor 24 is inversely convolved by the divider 26 and then immediately transmitted to the subtractor 28. become. Also,
There are various kinds of bit allocation methods, and the simplest is to use fixed bit allocation, simple bit allocation by each band energy of a signal, or bit allocation combining fixed and variable components. In addition, the magneto-optical disk 1
By driving at a rotational speed faster than the steady speed,
The dubbing may be performed at a speed higher than the bit compression rate. In this case, high-speed dubbing can be performed within the range permitted by the data transfer rate.

Next, FIG. 17 shows a high-efficiency decoding apparatus for specifically realizing the high-efficiency decoding process corresponding to the high-efficiency encoding in the digital signal processing method of the present invention.

In FIG. 17, the input terminals 152, 1
54 and 156 are supplied with the coded data that is the main information that has been subjected to the high-efficiency coding process described above, and the decoding circuits 146 and 14 to which these coded data correspond, respectively.
Sent to 7,148. In addition, each decoding circuit 146, 1
47 and 148 have terminals 153 and 15 respectively corresponding thereto.
Information compression parameters, which are sub-information, are also supplied via 5, 157. Each of these decoding circuits 146, 147, 1
At 48, the encoded data is decoded using the information compression parameter to restore the spectrum data on the frequency axis.

Each of the above decoding circuits 146, 147, 148
The output data from each is sent to the corresponding IMDCT circuits 143, 144, 145. These IMDCT
The circuits 143, 144, and 145 perform IMDCT processing, which is an inverse transformation corresponding to the above-described MDCT processing. That is, for the spectrum data in the 0 to 5.5 kHz band among the spectrum data from the decoding circuits 146, 147 and 148, the IMDCT circuit 145 and the spectrum data in the 5.5 to 11 kHz band are used for the IMDCT circuit. In 144, further 11-22kH
The IMDCT circuit 143 performs IMDCT processing on the spectrum data in the z band.

Further, the output of the IMDCT circuit 143 is sent to a band synthesizing filter (IQMF) circuit 141 which performs a process reverse to that of the band dividing filter 11. Also,
The outputs of the IMDCT circuits 144 and 145 are output to a band synthesis filter (IQ) that performs a process reverse to that of the band division filter 12.
MF) circuit 142. The output of the band synthesis filter circuit 142 is also sent to the band synthesis filter circuit 141. Therefore, the band synthesis filter circuit 14
From 1, a digital audio signal obtained by synthesizing the signals divided into the respective bands is obtained. This digital audio signal is output from the output terminal 140.

[0120]

As is apparent from the above description, in the digital signal processing method and apparatus and the recording medium of the present invention, when the signal-to-noise ratio is not sufficient due to the lack of the bit allocation amount. However, if the difference is small due to the difference in size from the spectral component with the maximum value in the block for block floating, change the size of the spectral component to be large, or if the difference is large By setting the quantization value to zero, it is possible to reduce the influence of the quantization noise on the sound quality. Therefore, in the present invention, it is possible to reduce the quantization noise for a signal of a sound having a large tonality in a floating block in high-efficiency coding, such as a signal of a trumpet sound, and reduce sound quality deterioration.

[Brief description of drawings]

FIG. 1 is a block circuit diagram showing a configuration example of a compressed data disk recording / reproducing apparatus as an embodiment for realizing a digital signal processing method of the present invention.

FIG. 2 is a diagram showing recorded contents of a magneto-optical disk and an IC card.

FIG. 3 is a schematic front view showing an example of the external appearance of the apparatus of this embodiment.

FIG. 4 is a block circuit diagram showing a specific example of a high-efficiency compression encoding apparatus that can be used for bit rate compression encoding according to the present embodiment.

FIG. 5 is a diagram showing a data structure of a processing block in each mode of bit compression.

FIG. 6 is a block circuit diagram of a specific example for performing a bit allocation calculation.

FIG. 7 is a diagram showing spectra of bands divided in consideration of each critical band and block floating.

FIG. 8 is a diagram showing a masking spectrum.

FIG. 9 is a diagram in which a minimum audible curve and a masking spectrum are combined.

FIG. 10 is a block circuit diagram of a configuration that realizes a second bit allocation method.

FIG. 11 is a diagram showing a noise spectrum when the signal spectrum is flat in the second bit allocation method.

FIG. 12 is a diagram showing bit allocation when the signal spectrum is flat in the second bit allocation method.

FIG. 13 is a diagram showing a noise spectrum when the tonality of a signal spectrum is high in the second bit allocation method.

FIG. 14 is a diagram for explaining bit allocation when the tonality of a signal spectrum is high in the second bit allocation method.

FIG. 15 is a diagram for explaining a difference in signal-to-noise ratio caused by a difference in signal tonality.

[Fig. 16] Fig. 16 is a diagram for explaining non-linear conversion adapted to a block having a low signal to noise ratio.

FIG. 17 is a block circuit diagram showing a specific example of a high-efficiency compression encoding apparatus that can be used for bit rate compression encoding of this embodiment.

[Explanation of symbols]

 1 Magneto-optical disk 2 IC card 3 Additional compression / expansion block 5 Recording / reproducing device 6 Magneto-optical disk slot 7 IC card slot 11, 12 Band division filter 13, 14, 15 Orthogonal transform (MDCT) circuit 18 Adaptive bit allocation encoding circuit 20 Block selection circuit 22 Energy detection circuit for each band 23 Convolution filter circuit 27 Synthesis circuit 28 Subtractor 30 Allowable noise correction circuit 32 Minimum audible curve generation circuit 33 Correction information output circuit 40, 41, 42 Non-linear processing circuit 43 Bit allocation calculation circuit 53 optical head 54 magnetic head 56 servo control circuit 57 system controller 62, 83 A / D converter 63 ATC encoder 64, 72, 85 memory 65 encoder 66 magnetic head drive circuit 71 decoder 73 ATC decoder 74 D / A conversion 84 Residue bit removal and word length zero processing circuit 85 RAM 121 Low pass filter 122, 123 Band division filter 124 High frequency signal processing circuit 125 Down sampling circuit 126, 127, 128 MDCT circuit 129 Bit allocation circuit 141 142 Band synthesis filter 143 144 144 145 Inverse orthogonal transformation circuit 146 147 148 Decoding circuit 301 Energy calculation circuit for each band 302 Spectrum smoothness calculation circuit 304 Bit division rate determination circuit 305 Total number of usable bits 306 Energy-dependent bit allocation circuit 307 Fixed bit allocation Circuit 308-bit sum operation circuit

Claims (39)

[Claims] 1. A digital signal processing method for transmitting a digital signal, wherein an input digital signal is converted into a signal component in a plurality of blocks each having a finite time width and a finite frequency width, each signal component including a plurality of signal components, A digital signal processing method characterized by nonlinearly processing a signal component in at least a part of the plurality of blocks and quantizing the nonlinearly processed signal component. 2. The digital signal processing according to claim 1, wherein the signal component is a spectral component, and the non-linear processing largely processes the spectral component excluding the spectral component which gives at least the maximum value in the block. Method. 3. The non-linear processing processes signal components excluding a signal component having at least a maximum signal-to-noise ratio in the block such that a quantized value by the quantization of the signal component becomes zero. The digital signal processing method according to claim 1, wherein 4. The signal component is a spectral component, and a process of normalizing the signal component is also performed, and the non-linear process includes a first comparison level smaller than a normalization level in the normalization and the first comparison level. For a spectral component having a magnitude between the second comparison levels smaller than the comparison level, the spectral component is increased or the quantized value by the quantization of the spectral component is set to zero. 2. The digital signal processing method according to claim 1, wherein a spectral component having a value smaller than the second comparison level is processed so that the quantized value by the quantization of the spectral component becomes zero. 5. The digital signal processing method according to claim 4, wherein the first comparison level and the second comparison level are variable according to the value of the maximum spectral component in the block. 6. The larger the value of the maximum spectral component in the block, the lower the first comparison level and / or the second comparison level increases. The described digital signal processing method. 7. The word length determined by bit allocation obtained based on the signal component before the non-linear processing is:
2. The digital signal processing method according to claim 1, wherein a block shorter than a preset word length is selected as a block for performing the non-linear processing. 8. The signal component is a spectral component, and based on the value of the maximum spectral component in each block,
2. The digital signal processing method according to claim 1, wherein a block on which the non-linear processing is performed is selected. 9. The digital signal processing method according to claim 8, wherein when the value of the maximum spectral component of the block is equal to or larger than a predetermined value, the block is selected as a block for performing the non-linear processing. 10. The digital signal processing method according to claim 1, wherein a block to be subjected to the non-linear processing is selected based on the tonality of each block. 11. The signal component is a spectral component, and the tonality is a component having at least a maximum signal-to-noise ratio among spectral components in a block.
11. The digital signal processing method according to claim 10, wherein the determination is performed based on the second component, which is a spectral component in the block excluding the first component. 12. The tonality is a ratio of a first value obtained from the first component and a second value obtained from the second component. The described digital signal processing method. 13. The digital signal according to claim 12, wherein the first value is an effective value of the first component, and the second value is an effective value of the second component. Processing method. 14. A digital signal processing device for transmitting a digital signal, wherein the input digital signal is converted into a signal component in a plurality of blocks having a finite time width and a finite frequency width, each of which includes a plurality of signal components. Means, a non-linear processing means for performing non-linear processing on a signal component in at least a part of the plurality of blocks, and an encoding means for quantizing the non-linearly processed signal component. A digital signal processing device characterized by: 15. The digital signal processing according to claim 14, wherein the signal component is a spectral component, and the non-linear processing means increases the spectral component excluding the spectral component which gives at least the maximum value in the block. apparatus. 16. The non-linear processing means reduces a signal component excluding a signal component having at least the maximum signal-to-noise ratio in the block so that the quantized value of the signal component by the quantization of the encoding means becomes zero. 15. The digital signal processing device according to claim 14, wherein: 17. The signal component is a spectral component, the encoding unit normalizes the signal component, and the non-linear processing unit includes a first comparison level smaller than a normalization level in the normalization and the first comparison level. For a spectral component having a magnitude between the second comparison levels less than the comparison level of, the spectral component is increased or the quantized value of the quantization of the spectral component is zero, 15. The digital signal processing device according to claim 14, wherein for a spectral component having a value smaller than the second comparison level, the quantized value by the quantization of the spectral component becomes zero. 18. The digital signal processing apparatus according to claim 17, wherein the first comparison level and the second comparison level are variable according to the value of the maximum spectral component in the block. 19. The larger the value of the maximum spectral component in the block, the lower the first comparison level,
19. The digital signal processing device according to claim 18, wherein the second comparison level is increased. 20. A block having a word length determined by a bit allocation obtained based on the signal component before the non-linear processing shorter than a preset word length is selected as a block for performing the non-linear processing. 15. The digital signal processing device according to claim 14, further comprising block selecting means. 21. The signal component is a spectral component, and based on the value of the maximum spectral component in each block,
15. The digital signal processing device according to claim 14, further comprising block selecting means for selecting a block for performing the non-linear processing. 22. The digital device according to claim 21, wherein the block selection means selects the block as a block for performing the linear processing when the value of the maximum spectral component of the block is equal to or larger than a predetermined value. Signal processing device. 23. The digital signal processing apparatus according to claim 14, further comprising block selecting means for selecting a block to be subjected to the non-linear processing based on the tonality of each block. 24. The signal component is a spectral component, and the tonality is a first component which is a component having at least a maximum signal-to-noise ratio among spectral components in a block, and the first component. 24. The digital signal processing device according to claim 23, wherein the digital signal processing device is obtained on the basis of a second component which is a spectral component in a block other than 25. The tonality is the ratio of a first value obtained from the first component to a second value obtained from the second component. The described digital signal processing device. 26. The digital signal according to claim 25, wherein the first value is an effective value of the first component and the second value is an effective value of the second component. Processing equipment. 27. A digital signal is converted into a signal component in a plurality of blocks each having a finite time width and a finite frequency width, and a signal component in at least a part of the plurality of blocks is nonlinearly processed, A recording medium, wherein recording data generated by quantizing the non-linearly processed signal component is recorded. 28. The recording medium according to claim 27, wherein the signal component is a spectral component, and the non-linear processing largely processes a spectral component excluding a spectral component which gives at least a maximum value in the block. . 29. The non-linear process processes a signal component excluding a signal component having at least a maximum signal-to-noise ratio in the block such that a quantized value by the quantization of the signal component becomes zero. 28. The method according to claim 27, wherein
The recording medium described. 30. The signal component is a spectral component, the signal component is normalized when the recording data is generated, and the non-linear processing is performed by a first comparison smaller than a normalization level in the normalization. For a spectral component having a magnitude between a level and a second comparison level smaller than the first comparison level, the spectral component is increased or the quantized value by the quantization of the spectral component is zero. 28. The spectral component having a value smaller than the second comparison level is processed so that the quantized value by the quantization of the spectral component becomes zero. Recording medium. 31. The recording medium according to claim 30, wherein the first comparison level and the second comparison level are variable according to the value of the maximum spectral component in the block. 32. The larger the value of the maximum spectral component in the block, the lower the first comparison level,
32. The recording medium according to claim 31, wherein the second comparison level is increased. 33. In the recording data, a block having a word length determined by a bit allocation obtained based on the signal component before the non-linear processing is shorter than a preset word length is subjected to the non-linear processing. 3. The block is formed by performing a process of selecting as a block for performing.
7. The recording medium according to 7. 34. The signal component is a spectral component, and the recording data is formed by performing a process of selecting a block to be subjected to the non-linear processing based on a value of a maximum spectral component in each block. 28. The recording medium according to claim 27, wherein: 35. The recording medium according to claim 34, wherein when the value of the maximum spectral component of the block is equal to or larger than a predetermined value, the block is selected as a block for performing the non-linear processing. 36. The recording medium according to claim 35, wherein the recording data is formed by performing a process of selecting a block on which the non-linear process is performed based on the tonality of each block. 37. The tonality comprises a first component which is a component having at least a maximum signal-to-noise ratio among spectral components in a block and a spectral component in the block excluding the first component. The recording medium according to claim 36, wherein the recording medium is obtained based on the second component. 38. The tonality is the ratio of a first value obtained from the first component to a second value obtained from the second component. The recording medium described. 39. The recording medium according to claim 38, wherein the first value is an effective value of the first component, and the second value is an effective value of the second component. .