JP2001142497A - Method and device for digital signal processing, method and device for digital signal recording, and recording medium - Google Patents

Abstract

PROBLEM TO BE SOLVED: To make easily performable an processing such as mixing without considering delay correction, an editing influence of a conversion block length, etc. SOLUTION: Adaptive bit assignment decoding circuits 902a and 902b decode a part of efficiently encoded data, which are related to musical pieces a and b, in accordance with adaptive bit assignment encoding. Reverse orthogonal transformation circuits 903a and 903b perform reverse orthogonal transformation of outputs of circuits 902a and 902b. Multipliers 905a and 905b multiply outputs of memories 904a and 904b by a multiplication coefficient values supplied from multiplication coefficient generation circuits 906. An adder 907 adds the outputs of multipliers 905a and 905b in each band. A block determination and orthogonal transformation circuit 908 determines a block size for each band with respect to the output of the adder 907 and performs orthogonal transformation. An adaptive bit assignment encoding circuit 909 converts the output of the circuit 908 to efficiently encoded data.

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 device and a processing method for digital signals such as digital audio data, a digital signal recording device and a recording method, and a recording medium.

[0002]

2. Description of the Related Art As a conventional technique relating to high-efficiency encoding of an audio signal, for example, an audio signal in a time domain is divided into blocks on a unit time basis, and a signal on a time axis for each block is converted into a signal on a frequency axis ( There is known a transform coding method, which is one of the blocking frequency band division methods for dividing the signal into a plurality of frequency bands by performing an orthogonal transform, and encoding each band. In addition, band division coding (sub-band coding), which is one of non-blocking frequency band division methods for dividing and encoding a time domain audio signal into a plurality of frequency bands without blocking the signal every unit time, is described. Coding (SBC: Su
b Band Coding)) A method is known.

[0003] Further, there is also known a high-efficiency coding method combining the above-mentioned band division coding and transform coding. In this method, for example, a signal in each band divided by a band division coding scheme is orthogonally transformed into a signal in a frequency domain by a transform coding scheme, and encoding is performed for each orthogonally transformed band.

Here, as a band division filter used in the above-mentioned band division coding system, for example, QMF (Q
uadrature Mirror filter). QMF
For example, RECrochiere Digital coding
of speech in subbands Bell Syst.Tech.J. Vol. 55,
No. 8 (1976). Also ICASSP 83, BOST
ON Polyphase Quadrature filters-A new subband codi
ng technique JosephH. Rothweiler has a Polyphase Quadrature filter.
An equal bandwidth filter splitting technique and apparatus is described.

As the orthogonal transform, for example, an input audio signal is divided into blocks in a predetermined unit time (frame), and a fast Fourier transform (FFT), a cosine transform (DCT), a modified DCT transform (MD
A method of converting a time axis into a frequency axis by performing CT or the like is known. For MDCT, for example,
ICASSP 1987 Subband / Transform Coding Using Filter
Bank Designs Based on Time Domain Aliasing Cancell
ation JPPrincen ABBradley Univ. of Surrey Roy
al Melbourne Inst. of Tech.

On the other hand, there is known an encoding method which uses a frequency division width in consideration of human auditory characteristics when quantizing each frequency component divided into frequency bands. In other words, a bandwidth called a critical band (critical band) is widely used such that the higher the bandwidth, the wider the bandwidth. An audio signal may be divided into a plurality of bands (for example, 25 bands) using such a critical band. According to such a band division method, when encoding data for each band, encoding is performed by predetermined bit allocation for each band or adaptive bit allocation for each band.
For example, when the MDCT coefficient data generated by the MDCT process is encoded by the above-described bit allocation, an adaptive bit is applied to the MDCT coefficient data of each band generated corresponding to each block. Numbers are allocated, and encoding is performed under such bit number allocation.

[0007] As a known document on such a bit allocation method and an apparatus for realizing the bit allocation method, for example, the following can be cited. First, for example, IEEE Transaction
s ofAccoustics, Speech, and Signal Processing, vol.AS
SP-25, No. 4, August (1977) describes a method for allocating bits based on the magnitude of a signal for each band.
Also, for example, ICASSP 1980 Thecritical band coder--di
gital encoding of the perceptual requirements of
The auditory system MA Kransner MIT describes a method of obtaining a required signal-to-noise ratio for each band and performing fixed bit allocation by using auditory masking.

In encoding for each band, a so-called block floating process for realizing more efficient encoding is performed by normalizing and quantizing each band. For example, when encoding the MDCT coefficient data generated by the MDCT process, the quantization is performed by performing the normalization corresponding to the above-described maximum value of the MDCT coefficient for each band and the like, and then performing the quantization. , More efficient encoding is performed. The normalization processing is performed, for example, as follows. That is, a plurality of types of values that are numbered in advance are prepared, and among the plurality of types of values, a value related to normalization for each block is determined by a predetermined calculation process, and the number assigned to the determined value is determined. Is used as normalization information. Numbering corresponding to a plurality of types of values is performed under a certain relationship, for example, such that an increase or decrease in the number by 1 corresponds to an increase or decrease in the audio level by 2 dB.

The high-efficiency encoded data generated by the above-described method is decoded as follows. First, a process of generating MDCT coefficient data based on encoded data is performed with reference to bit allocation information, normalization information, and the like for each band. A so-called inverse orthogonal transform is performed based on the MDCT coefficient data, thereby generating time-domain data. If band division has been performed by the band division filter in the process of high-efficiency encoding, processing for synthesizing data in the time domain using a band synthesis filter is further performed.

In recent years, there has been a demand for a function of continuing a performance without making a break for each piece of music, that is, when reproducing a plurality of music pieces (tracks). For example, mixing can be performed by decoding two pieces of high-efficiency-encoded music and performing arithmetic processing such as addition processing on digital data as a decoding result, for example. As an example of mixing, cross-fade is performed, that is, a fade-out process is performed so that the playback volume level gradually decreases near the end of a certain song a, and the playback volume level gradually increases near the start position of another song b. There is a process in which the music pieces a and b are continuously reproduced without a break by performing a fade-in process. In this case, the signal levels of the music pieces a and b may be multiplied by a desired coefficient and then added.

[0011] Further, the data obtained by performing arithmetic processing on the decoded signal is subjected to a compensation process for compensating for the delay caused by the encoding and decoding, and then the encoding is performed again. There is a method of creating encoded data that maintains continuity with highly efficient encoded data. According to such a method, it is possible to obtain highly efficient encoded data in which a desired portion of a plurality of encoded data is added. For example, it is possible to create high-efficiency encoded data corresponding to the crossfade described above.

[0012]

However, such a method requires a delay compensation process before decoding the processed data again and converting it into highly efficient encoded data. That is, there is a delay in the encoding and decoding processing between the original high-efficiency encoded data and the highly-efficient encoded data generated by the re-encoding after the arithmetic processing. Due to this delay, a phase shift occurs from the original high efficiency encoded data. To cope with this, in order to maintain continuity between the original high-efficiency encoded data and the high-efficiency encoded data generated by re-encoding after the arithmetic processing,
Delay compensation processing (phase correction) is required.

On the other hand, the high-efficiency coded data is converted into orthogonally transformed data, that is, a spectrum (MDCT) on the frequency axis.
In the method in which decoding is performed up to coefficient data) and editing is performed by calculating the spectrum on the frequency axis, there is no need to perform phase correction. For this reason, there are advantages that the circuit configuration is simplified, the processing time is reduced, and sound quality deterioration due to the processing is small. However, in this method, when there is a certain degree of freedom in the transform block length in the orthogonal transform, it is necessary to consider this.

For example, when orthogonal transformation is performed on so-called audio data having a large attack, that is, audio data in which the volume suddenly increases in a very short time, a technique for shortening the conversion block length is known. As a result, instead of reducing the resolution on the frequency axis, the resolution on the time axis can be increased, so that the influence of noise components peculiar to audio data having a large attack can be reduced. Generally, a normal conversion block length is defined as a long mode, and a short conversion block length is defined as a short mode, and information indicating these two modes is recorded as additional information corresponding to highly efficient encoded data. Is done.

In such a case, for example, in a method in which high-efficiency coded data relating to two music pieces (tracks) is decoded into a spectrum on the frequency axis and processed by arithmetic processing, editing is performed. Editing can be performed if the conversion block lengths of the two songs match, but editing cannot be performed if the conversion block lengths of the two songs do not match in the edited portion. .

Therefore, an object of the present invention is to provide a digital signal processing apparatus and processing method and a digital signal processing method capable of easily performing editing processing such as mixing without considering the effects of delay correction and the influence of a conversion block length. A recording apparatus, a recording method, and a recording medium on which an editing process is recorded.

[0017]

According to the first aspect of the present invention, a plurality of signal components are generated by dividing an input digital signal into a plurality of frequency components, and a time and a frequency are related based on the generated signal components. And determining the bit allocation amount for each of the plurality of blocks specified by referring to the information compression parameter including the determined bit allocation amount, and quantizing the signal component to generate the high-efficiency coding. In a digital signal processing device that performs editing processing on data, a part of high-efficiency coded data is inversely quantized, arithmetic processing is performed based on the result of the inverse quantization, and data generated as a result of the arithmetic processing is processed. A digital signal processing device characterized by generating high-efficiency encoded data by quantizing.

According to a second aspect of the present invention, an input digital signal is divided into a plurality of frequency band components, and a signal component on a frequency axis is generated by orthogonally transforming the band-divided signal. By determining a bit allocation amount for each of a plurality of blocks specified in relation to time and frequency, and by referring to an information compression parameter including the determined bit allocation amount, the signal component is quantized. In a digital signal processor that edits the generated high-efficiency coded data, a part of the high-efficiency coded data is inversely quantized, the result of the inverse quantization is subjected to inverse orthogonal transform, and the result of the inverse orthogonal transform is performed. Perform arithmetic processing on
Orthogonally transform the data generated as a result of the arithmetic processing,
A digital signal processing apparatus characterized in that highly efficient encoded data is generated by quantizing the result of orthogonal transformation.

According to a thirteenth aspect of the present invention, a plurality of signal components are generated by dividing an input digital signal into a plurality of frequency components, and a plurality of signal components are specified in relation to time and frequency based on the generated signal components. Edit processing on the highly efficient encoded data generated by quantizing the signal component by referring to the information compression parameter including the determined bit allocation amount while determining the bit allocation amount for each block of In the digital signal processing method, a part of the highly efficient encoded data is inversely quantized, arithmetic processing is performed based on the result of the inverse quantization, and data generated as a result of the arithmetic processing is quantized. A digital signal processing method characterized by generating efficiency encoded data.

According to a fourteenth aspect of the present invention, an input digital signal is divided into a plurality of frequency band components, and a signal component on the frequency axis is generated by orthogonally transforming the band-divided signal, based on the generated signal component. By determining a bit allocation amount for each of a plurality of blocks specified in relation to time and frequency, and by referring to an information compression parameter including the determined bit allocation amount, the signal component is quantized. In a digital signal processing method for performing editing processing on generated high-efficiency encoded data, a part of the high-efficiency encoded data is inversely quantized, the result of inverse quantization is subjected to inverse orthogonal transform, and the result of inverse orthogonal transform is performed. , Perform orthogonal processing on data generated as a result of the arithmetic processing, and quantize the result of the orthogonal transformation to generate highly efficient encoded data. It is a digital signal processing method according to claim.

According to a fifteenth aspect of the present invention, a plurality of signal components are generated by dividing an input digital signal into a plurality of frequency components, and a plurality of signal components specified in relation to time and frequency are generated based on the generated signal components. Edit processing on the highly efficient encoded data generated by quantizing the signal component by referring to the information compression parameter including the determined bit allocation amount while determining the bit allocation amount for each block of In a digital signal recording device that records the result of the editing process, a part of the high-efficiency coded data is inversely quantized, arithmetic processing is performed based on the result of the inverse quantization, and data generated as a result of the arithmetic processing Digital signal recording apparatus for generating high-efficiency encoded data by quantizing data and recording the generated high-efficiency encoded data A.

According to a sixteenth aspect of the present invention, an input digital signal is divided into a plurality of frequency band components, and a signal component on a frequency axis is generated by orthogonally transforming the band-divided signal. By determining a bit allocation amount for each of a plurality of blocks specified in relation to time and frequency, and by referring to an information compression parameter including the determined bit allocation amount, the signal component is quantized. In a digital signal recording device that performs editing processing on the generated high-efficiency encoded data and records the edited processing result, a part of the high-efficiency encoded data is inversely quantized, and the result of the inverse quantization is subjected to inverse orthogonal transform. By performing an arithmetic operation on the result of the inverse orthogonal transform, orthogonally transforming data generated as a result of the arithmetic process, and quantizing the result of the orthogonal transform, It generates a rate coded data, a digital signal recording apparatus and recording the generated high-efficiency encoded data.

According to a twentieth aspect of the present invention, a plurality of signal components are generated by dividing an input digital signal into a plurality of frequency components, and the plurality of signal components are specified in relation to time and frequency based on the generated signal components. Edit processing on the highly efficient encoded data generated by quantizing the signal component by referring to the information compression parameter including the determined bit allocation amount while determining the bit allocation amount for each block of In the digital signal recording method of recording the result of the editing process, a part of the high-efficiency coded data is dequantized, the arithmetic processing is performed based on the result of the inverse quantization, and the data generated as a result of the arithmetic processing Digital signal recording method comprising generating high-efficiency encoded data by quantizing data and recording the generated high-efficiency encoded data A.

According to a twenty-first aspect of the present invention, an input digital signal is divided into a plurality of frequency band components, and a signal component on a frequency axis is generated by orthogonally transforming the band-divided signal. By determining a bit allocation amount for each of a plurality of blocks specified in relation to time and frequency, and by referring to an information compression parameter including the determined bit allocation amount, the signal component is quantized. In a digital signal recording method for performing editing processing on the generated high-efficiency encoded data and recording the edited processing result, a part of the high-efficiency encoded data is inversely quantized, and the result of the inverse quantization is subjected to inverse orthogonal transform. By performing an arithmetic operation on the result of the inverse orthogonal transform, orthogonally transforming data generated as a result of the arithmetic process, and quantizing the result of the orthogonal transform, Generates a rate encoded data is a digital signal recording method, comprising recording the generated high-efficiency encoded data.

According to a twenty-second aspect of the present invention, a plurality of signal components are generated by dividing an input digital signal into a plurality of frequency components, and the plurality of signal components are specified in relation to time and frequency based on the generated signal components. The bit allocation amount for each block is determined, and with reference to the information compression parameter including the determined bit allocation amount, the highly efficient encoded data generated by quantizing the signal component is edited. In the recording medium on which the result of the editing process is recorded, a part of the high-efficiency encoded data is inversely quantized, the arithmetic processing is performed based on the result of the inverse quantization, and the data generated as a result of the arithmetic processing is quantized. And recording the high-efficiency coded data generated by the recording.

According to a twenty-third aspect of the present invention, an input digital signal is divided into a plurality of frequency band components, and a signal component on a frequency axis is generated by orthogonally transforming the band-divided signal. By determining a bit allocation amount for each of a plurality of blocks specified in relation to time and frequency, and by referring to an information compression parameter including the determined bit allocation amount, the signal component is quantized. On a recording medium on which the edited result of the generated high-efficiency encoded data is edited, a part of the high-efficiency encoded data is inversely quantized, and the result of the inverse quantization is inversely orthogonally transformed. A high-efficiency code generated by performing arithmetic processing on the result of the orthogonal transformation, orthogonally transforming data generated as a result of the arithmetic processing, and quantizing the result of the orthogonal transformation. A recording medium, wherein a data is recorded.

According to the invention described above, the high-efficiency coded data is subjected to the inverse quantization or the inverse quantization and the inverse orthogonal transform, and the arithmetic processing is performed on the data. By being encoded into encoded data, editing processing such as mixing can be performed.

[0028]

DESCRIPTION OF THE PREFERRED EMBODIMENTS One embodiment of the present invention will be described.
Hereinafter, description will be made with reference to the drawings. One embodiment of the present invention uses, for example, an audio PCM (Pull
(Se Code Modulation) The present invention is applied to an MD recorder for recording and reproducing digital data such as data. The present invention can be applied to digital data other than digital audio data, for example, digital video data.

FIG. 1 shows an overall configuration of an example of an audio signal processing device including an MD recorder. The MD recorder 30 is provided with an interface for inputting and outputting digital data. Using this interface, the MD recorder 30 and the converter 55 are connected via the communication path 50, and the converter 55 and the personal computer 40 are connected via the communication path 50 '. The data communication standard in the personal computer 40 is, for example, RS (Recommended Standard) -232C.
And SCSI (Small Computer System Interface).

The converter 55 is a personal computer 4
It is provided to absorb the difference in the form of the control signal between the data communication standard on the 0 side and the data communication standard on the MD recorder 30 side. The personal computer 40
At the IEEE (Institute of Electrical and Ele
The personal computer 40 and the MD recorder 30 may be directly connected by using the Ctronic Engeneers (1394) as a data communication standard.

Next, referring to FIG. 2, the MD recorder 30
Will be described. The disc 1 is an MD. This disk 1 is stored in a cartridge having a shutter mechanism. A recording or reproducing operation is performed on the disk 1 by applying a magnetic field by the magnetic head 6 and / or irradiating a laser beam by the operation of the optical head 3. The spindle motor 2 drives the disk 1 to rotate in accordance with a command from the servo circuit 9.

The optical head 3 is relatively moved as a whole in the radial direction of the disk 1 by the thread motor 5 which operates according to a command from the servo circuit 9. The optical head 3 includes an objective lens 3a, a biaxial mechanism 4, a semiconductor laser (not shown), and a light receiving unit. The intensity of the laser beam emitted from the semiconductor laser is switched between recording and reproduction. When the light receiving section receives the reflected light obtained by reflecting the laser light by the disk 1,
A read signal relating to recording data and a signal necessary for servo control are read. The two-axis mechanism 4 includes the objective lens 3
and a tracking coil for driving the objective lens 3a in the radial direction of the disk 1 and a focusing coil for driving the objective lens 3a toward and away from the recording surface of the disk 1.

Hereinafter, the configuration and operation of the processing based on the reproduced signal will be described. A detection signal generated by a light receiving unit in the optical head 3 is supplied to the RF amplifier 7. The RF amplifier 7 performs a focus error signal FE, a tracking error signal TE, R based on the supplied signal.
An F (Radio Frequency) signal and a spindle error signal are generated. FE and TE are supplied to the servo circuit 9 and the spindle error signal is supplied to the system controller 11.
Supplied to As the system controller 11, for example, a microcomputer can be used. The system controller 11 performs various controls related to operations such as recording and reproduction. The RF signal is EFM (Eight to Fourteen Modu).
lation) and CIRC (Cross Interleave Reed-Solom
on Coding) are supplied to the encoder / decoder 8 and the address decoder 10.

The servo circuit 9 outputs the output of the RF amplifier 7
Perform phase compensation and gain adjustment. The output of the servo circuit 9 is supplied to a focusing coil and a tracking coil in the two-axis mechanism 4 via a drive amplifier (not shown). Further, an LP (not shown) is provided in the servo circuit 9.
An F (Low Pass Filter) is provided, and a thread error signal is formed by subjecting the tracking error signal TE to LPF filtering. The thread error signal is supplied to the thread motor 5 via a thread drive amplifier (not shown). The sled motor 5 operates according to the sled error signal.

On the other hand, EFM and CIRC encoders
The decoder 8 performs a process of binarizing the RF signal supplied from the RF amplifier 7, and further performs a demodulation process corresponding to the EFM modulation. Then, data obtained as a result of the EFM demodulation processing is subjected to error correction processing based on CIRC coding. Further, address data extracted by the address decoder 10 is supplied to the EFM and CIRC encoder / decoder 8. The EFM and CIRC encoder / decoder 8 generates a spindle error signal based on the supplied address data. The spindle error signal is supplied to the system controller 11. The system controller 11 controls the operation of the spindle motor 2 by issuing a command to the servo circuit 9 based on the spindle error signal. Further, the EFM and CIRC encoder / decoder 8 uses a PLL (Pha) based on the binarized EFM signal.
se Lock Loop) is controlled.

The output of the EFM and CIRC encoder / decoder 8 is transmitted to the memory 1 via the memory controller 12.
3 is written. Writing of data to the memory 13 and reading of data from the memory 13 are controlled by the memory controller 12. Further, the memory controller 12 is controlled by the system controller 11. The signal read from the memory 13 is supplied to an audio compression / decompression encoder / decoder 14. The memory controller 12 controls the amount of processing data using the memory 13. While the transfer rate when data is read from the memory 13 is, for example, 0.3 Mbit / sec, the transfer rate when recording data reproduced from the disk 1 is written to the memory 13 is, for example, 1.4 Mbit. / Sec fast.

Due to such a difference in transfer rate, the memory 13 performs appropriate control to prevent the reproduced sound from being interrupted even when the data read from the disk 1 is interrupted due to disturbance such as vibration. Can be. Audio compression and decompression encoder / decoder 1
4, for example, ATRA applied to the supplied signal
The compression by the C (Acustic TRansferred Adopted Coding) method is decoded. The signal obtained by decoding the compression is converted into an analog audio signal by the D / A converter 15 and supplied to an audio output unit (not shown) via the audio output terminal 15.

The disk 1 is provided with a meandering groove having a predetermined frequency, for example, 22.05 Hz. Thus, address data is recorded by FM (Frequency Modulation) modulation. The address data is based on the RF signal supplied from the RF amplifier 9,
It is extracted by the address decoder 10. That is,
The address decoder 10 is provided with a BPF (Band Pass
Filter) is built in and the supplied RF signal is BPF
The address data is extracted by FM demodulation via the. The extracted address data is supplied to the EFM and CIRC encoder / decoder 8 as described above.

Next, the configuration and operation of recording will be described. An analog audio signal is supplied to an A / D (Analog to Digital) converter 18 via an input terminal 17. The A / D converter 18 converts the supplied signal into a digital signal, and supplies the digital signal to the audio compression encoder and the expansion decoder 14. Further, the digital audio signal may be directly supplied to the audio compression encoder and the expansion decoder 14 via the terminal 21. The audio compression encoder / decompression decoder 14 compresses the supplied digital audio signal by, for example, the ATRAC method and temporarily stores the digital audio signal in the memory 13 via the memory controller 12 at a transfer rate of, for example, 0.3 Mbit / sec. Is done. The memory controller 12 permits reading from the memory 13 when detecting that a predetermined amount or more of data has been stored in the memory 13.

The digital signal read from the memory 13 is supplied to the EFM and CIRC encoder / decoder 8. The EFM and CIRC encoder / decoder 8 subjects the supplied signal to EFM and CIRC encoding for error correction. The signal generated by such processing is supplied to the magnetic head drive circuit 35. The magnetic head drive circuit 35 controls the magnetic head 6 to apply an appropriate magnetic field for recording the supplied signal to the disk 1. The power of the semiconductor laser in the optical head 3 is made larger than that at the time of reproduction at the timing of application of the magnetic field and palpitations. As a result, the surface of the disk 1 irradiated with the laser beam is heated to the Curie temperature so that a magnetic field reversal can occur, and data is recorded magneto-optically.

Next, the medium format of the disk 1, ie, the MD, will be described with reference to FIG. For example, an information film is attached to a polycarbonate substrate, and a clamping plate 41 made of a magnetic material is attached at the center. The information film is composed of a recording film and a read-only film. The recording film is, in order from the substrate side, a dielectric layer, an MO layer, a dielectric layer, a reflective film,
The protective film is laminated. The read-only film includes a reflective film and a protective film. The area excluding the clamping plate 41 is an information area 42.

The innermost circumference of the information area 42 is a lead-in area 43. Lead-in area 4
3, a read-only film is applied, and information is recorded in advance in the form of pits. A recordable area 44 on which a recording film is applied is provided outside the lead-in area 43, and a lead-out area 45 is provided on the outermost periphery of the disk 1. Further, a program area 47 for recording a program is arranged outside the recordable area 44.

On the inner peripheral side of the recordable 44, U
-U-TO that records TOC (User-Tsble Of Contents)
A C area 46 is arranged, and information about each program related to audio data and the like recorded in the program area 47 is recorded. The U-TOC is read into a memory in the MD recorder 30 when performing operations such as recording and reproduction. The U-TOC is referred to in a reproducing operation or the like, and is rewritten each time editing processing such as recording or erasing of data is performed. The U-TOC in the memory is used when, for example, an ejection command for the disc 1 is issued,
It is written to the U-TOC area 46 at a predetermined timing such as time.

It should be noted that the U-TO
C may be written in the U-TOC area 46. U-TOC7 area lead-in area 43 and UT
Calibration area 4 between OC area 46
8 are provided. A gap area 49 is provided between the U-TOC area 46 and the program area 47. No user data is recorded in the calibration area 48 and the gap area 49. The calibration area 48 is used for adjusting laser power and the like.

The operation by the user via the personal computer 40 will be described. FIG. 4 shows a window 80 as an example of the operation screen. Window 80
Are displayed on the monitor of the personal computer 40 according to the user's request. In a track number column 81, a time display column 82, and a name record column 83, information on a track, which is a data unit corresponding to each song, is displayed. The track number column 81 displays the track number of each track. In the time display column 82, the performance time of each track is recorded. In the name record column 83,
Display and editing of character information such as the title (music name) of each track are performed. The scroll bar 84 is used to display tracks that are not displayed when the number of recorded tracks is large and cannot be displayed on one screen.

In the disk name column 85, the disk name given to the disk 1 is displayed. Also, remaining time display column 8
6 shows the remaining recordable time on the disc 1. Buttons 87 related to various operations are arranged on the left side of the window 80. By operating the buttons 87,
A control signal corresponding to the MD recorder 30 is issued from the personal computer 40. For example, by operating a button labeled “POWER” arranged at the top, the power of the MD recorder 30 can be turned on / off. Similarly, on the left side of the window 80, various icons 88a and 8 for editing audio data recorded in units of tracks, that is, songs.
8b, 88c, 88d and 88e are arranged.

The icon 88a is an icon for instructing the movement of music, that is, the order of performance during sequential reproduction. The icons 88b and 88c are icons for instructing division and combination of music, respectively. Icon 8
8d is an icon for deleting a part of the music. The icon 88e is an icon for instructing deletion of a music piece, that is, deletion of a track as a unit. Using these editing icons, an editing operation can be performed on a plurality of songs displayed in the window in FIG.

Next, with reference to FIG. 5, a high-efficiency encoding process for generating compressed encoded data to be recorded on the disk 1 will be described. The configuration shown in FIG.
It is provided in the audio compression encoder and expansion decoder 14 in FIG. Here, band division coding (SB
C), the adaptive conversion coding (ATC) and the adaptive bit allocation are performed to perform the coding process. For example, when the sampling frequency is 44.1 kHz, a PCM data signal of 0 to 22 kHz is supplied to the band division filter 101 via the input terminal 100. The band division filter 101 converts the supplied signal from 0 to 11 kHz.
It is divided into a z band and an 11 kHz to 22 kHz band. 1
Signals in the 1 to 22 kHz band are MDCT (Modified Discrete
te Cosine Transform) circuit 103 and block determination circuits 109, 110, and 111.

A signal in the 0 kHz to 11 kHz band is supplied to the band division filter 102. The band division filter 102 converts the supplied signal from 5.5 kHz to 11 kHz.
It is divided into a z band and a 0 to 5.5 kHz band. 5.5-
The signal in the 11 kHz band is supplied to the MDCT circuit 104 and the block decision circuits 109, 110, 111. Also, the signal in the 0-5.5 kHz band is transmitted to the MDCT circuit 10.
5 and the block determination circuits 109, 110, and 111. The band division filters 101 and 102 can be configured using, for example, a QMF filter or the like. The block determination circuit 109 determines a block length based on the supplied signal, and outputs information indicating the determined block length to the MD.
It is supplied to the CT circuit 103 and the output terminal 113.

The block determining circuit 110 determines the block length based on the supplied signal, and outputs information indicating the determined block length to the MDCT circuit 104 and the output terminal 115.
To supply. The block determining circuit 111 determines a block length based on the supplied signal, and outputs information indicating the determined block length to the MDCT circuit 105 and the MDCT circuit 105. And output terminal 11
7 Block length block determination circuit 110, 1
11 and 112 adaptively set the block length (block size) according to the time characteristic and frequency distribution of the supplied signal.

The MDCT circuits 103, 104, 105
MDCT processing is performed based on the supplied signal, and MDC
Generate T coefficient data or spectrum data on the frequency axis. High-frequency MDCT generated by MDCT circuit 103
The coefficient data or the spectrum data on the frequency axis is supplied to the adaptive bit allocation encoding circuit 106 and the bit allocation calculating circuit 118 after being subjected to a process of subdividing the critical bandwidth in consideration of the effectiveness of block floating. . The mid-range MDCT coefficient data or the spectrum data on the frequency axis generated by the MDCT circuit 104 is subjected to a process of subdividing the critical bandwidth in consideration of the effectiveness of block floating, and then the adaptive bit allocation encoding circuit 1
07 and the bit allocation calculation circuit 118.

The low-frequency MD generated by the MDCT circuit 105
The CT coefficient data or the spectrum data on the frequency axis is subjected to a process of summarizing for each critical band (critical band), and then supplied to the adaptive bit allocation encoding circuit 108 and the bit allocation calculating circuit 118. Here, the critical band is a frequency band divided in consideration of human auditory characteristics, and when the pure sound is masked by narrow band noise of the same strength near the frequency of a certain pure sound, the narrow band is It is the band of band noise. The critical band has a property that the bandwidth increases as the frequency increases. 0 to
The entire frequency band of 22 kHz is divided into, for example, 25 critical bands.

Based on the supplied MDCT coefficient data or spectrum data on the frequency axis and the block length information, the bit allocation calculation circuit 118 takes the above-described critical band and block floating , The masking amount, energy and / or peak value, etc., of each divided band are calculated, and a scale factor indicating the state of block floating and the number of allocated bits are calculated for each band based on the calculation result. The calculated number of allocated bits is supplied to adaptive bit allocation coding circuits 106, 107, and 108. In the following description, each divided band which is a unit of bit allocation is referred to as a unit block.

The adaptive bit allocation encoding circuit 106 determines the MDCT according to the block length information supplied from the block determination circuit 109, the number of allocated bits supplied from the bit allocation calculation circuit 118, and the scale factor information as normalization information. A process for requantizing (normalizing and quantizing) the spectrum data or MDCT coefficient data supplied from the circuit 103 is performed. As a result of such processing, highly efficient encoded data is generated. This high efficiency coding is supplied to the arithmetic unit 120. Adaptive bit allocation encoding circuit 10
7 is an MDC according to the block length information supplied from the block determination circuit 110, the allocated bit number and the scale factor information supplied from the bit allocation calculation circuit 118.
Spectrum data supplied from the T circuit 104 or M
A process of requantizing the DCT coefficient data is performed. As a result of such processing, highly efficient encoded data is generated. This highly efficient encoded data is supplied to the arithmetic unit 121.

The adaptive bit allocation encoding circuit 108 is supplied from the MDCT circuit 105 according to the block length information supplied from the block determination circuit 110, the number of allocated bits and the scale factor information supplied from the bit allocation calculation circuit 118. Spectral data or MDCT
Requantize the coefficient data. As a result of such processing,
Highly efficient encoded data is generated. This highly efficient encoded data is supplied to the arithmetic unit 122. The normalization information change circuit 119 and the computing units 120, 121, 122 will be described later.

FIG. 6 shows the MDCT circuits 103, 104, 1
5 shows an example of data supplied for each band, which is supplied to the network 05. By the operation of the block decision circuits 109, 110, and 111, the orthogonal transform block length can be set independently for each band for a total of three data output from the band division filters 101 and 102, and the signal time It is possible to switch the time resolution according to characteristics, frequency distribution, and the like. That is, when the signal is quasi-stationary in time, Long Mo that increases the orthogonal transform block length to, for example, 11.6 ms as shown in FIG. 6A.
de is used.

On the other hand, when the signal is non-stationary, a mode is used in which the orthogonal transform block length is divided into two or four as compared with that in the long mode. More specifically, Sho is divided into four parts, for example, 2.9 ms.
rt Mode (see FIG. 6B), or Middle Mode-a (FIG. 6) in which one part is divided into two parts, for example, 5.8 ms, and the other part is divided into four parts, for example, 2.9 ms.
C) or Middle Mode-b (FIG. 6D
) Is used. By setting the time resolution variously in this way, it is possible to adapt to an actual complicated input signal.

When the restrictions on the circuit scale and the like are small,
Obviously, by making the division of the orthogonal transform block length more complicated, the actual input signal can be more appropriately processed. The block length as described above is determined by the block determination circuits 109, 110, 111, and information on the determined block length is stored in the MDCT circuits 103, 10, 10.
4 and 105 and a bit assignment calculation circuit 118 and output via output terminals 113, 115 and 117.

Next, the bit allocation calculating circuit 118 will be described in detail with reference to FIG. Through the input terminal 301, spectrum data or MDCT coefficients on the frequency axis from the MDCT circuits 103, 104, and 105 and block length information from the block determination circuits 109, 110, and 111 are supplied to the energy calculation circuit 302. The energy calculation circuit 302 calculates the energy of each unit block by, for example, calculating the sum of the amplitude values in the unit block. Note that a configuration for calculating a peak value, an average value, or the like of the amplitude value may be provided instead of the energy calculation circuit 302, and the bit allocation processing may be performed based on the calculated value of the peak value, the average value, or the like of the amplitude value. .

FIG. 8 shows an example of the output of the energy calculation circuit 302. In FIG. 8, the spectrum SB of the total value of each band is indicated by a vertical line segment with a circle at the tip. Here, the horizontal axis indicates frequency, and the vertical axis indicates signal strength. Note that, in order to avoid complicating the drawing, in FIG.
B12), and the code “SB” is assigned to only the spectrum of B12.
Is attached.

The energy calculation circuit 302 performs a process of determining a scale factor value which is normalization information indicating a block floating state of a unit block.
Specifically, for example, some positive values are prepared in advance as scale factor value candidates, and among those taking values equal to or more than the maximum value of the absolute value of the spectral data or MDCT coefficient in the unit block, among them, The smallest one is adopted as the scale factor value of the unit block. The scale factor value candidates may be numbered using, for example, several bits in a form corresponding to the actual value, and the number may be stored in a ROM (Read Only Memory) (not shown) or the like. At this time, it is defined that the candidates for the scale factor value have values at intervals of, for example, 2 dB in numerical order. The number assigned to the scale factor value adopted for a certain unit block is used as sub-information, and is used as the scale factor information for the unit block.

The output of the energy calculation circuit 302, that is, each value of the spectrum SB is sent to the convolution filter circuit 303. The convolution filter circuit 303 includes, for example, a plurality of delay elements for sequentially delaying input data, a plurality of multipliers for multiplying outputs from these delay elements by a filter coefficient (weighting function), and a sum of outputs of the respective multipliers. And a total sum adder. The convolution filter circuit 303 performs convolution (convolution) such that the spectrum SB is multiplied by a predetermined weighting function and added in order to consider the influence on the masking of the spectrum SB.
Perform processing. By this convolution processing, the sum of the parts shown by the dotted lines in FIG. 8 is calculated.

Returning to FIG. 7, the output of the convolution filter circuit 303 is supplied to the arithmetic unit 304. The arithmetic unit 304 includes:
In addition, a tolerance function (a function that expresses the masking level)
Is supplied from the (n-ai) function generation circuit 305. The arithmetic unit 304 calculates a level α corresponding to an allowable noise level in the area convolved by the convolution filter circuit 303 according to the allowable function. here,
The level α corresponding to the allowable noise level (allowable noise level) is a level which becomes an allowable noise level for each band of the critical band by performing inverse convolution processing as described later. is there.
The calculated value of the level α is controlled by increasing or decreasing the allowable function.

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

Α = S− (n−ai) (1) In the equation (1), n and a are constants, a> 0, and S is the intensity of the convolved spectrum. n-
ai) is an allowable function. As an example, n = 38, a = 1
It can be.

Level α calculated by arithmetic unit 304
Is transmitted to the divider 306. The divider 306 performs a process of deconvolving the level α, and generates a masking spectrum from the level α as a result. This masking spectrum becomes an allowable noise spectrum. In general, when performing inverse convolution processing, complicated operations need to be performed. However, in one embodiment of the present invention, inverse convolution is performed using a simplified divider 306. ing. The masking spectrum is supplied to the synthesis circuit 307. The synthesizing circuit 307 is further supplied with data indicating the minimum audible curve RC as described later from the minimum audible curve generation circuit 312.

The synthesizing circuit 307 generates a masking spectrum by synthesizing the masking spectrum output from the divider 306 with the data of the minimum audible curve RC. The generated masking spectrum is subtracted by the subtractor 30.
8 is supplied. The output of the energy detection circuit 302, that is, the spectrum SB for each band, is supplied to the subtracter 308 after the timing is adjusted by the delay circuit 309. The subtractor 308 performs a subtraction process based on the masking spectrum and the spectrum SB.

As a result of this processing, a portion of the spectrum SB for each block which is lower than the level of the masking spectrum is masked. FIG. 9 shows an example of the masking. It can be seen that the portion below the level (denoted as MS) of the masking spectrum in the spectrum SB is masked. In order to avoid complicating the drawing, in FIG. 9, the symbol “SB” is assigned to the spectrum and the symbol “MS” is assigned to the level of the masking spectrum only at B12.

If the absolute noise level is below the minimum audible curve RC, the noise is inaudible to humans. The minimum audible curve differs depending on, for example, the reproduction volume at the time of reproduction even if the coding is the same. However, in an actual digital system, for example, there is not much difference in how music data enters a 16-bit dynamic range. It is considered that quantization noise below the level of the minimum audible curve is not audible in other frequency bands.

Therefore, for example, in the case of using the system so that noise around 4 kHz of the word length of the system cannot be heard, if an allowable noise level is obtained by synthesizing the minimum audible curve RC and the masking spectrum MS, The permissible noise level in this case is indicated by the hatched portion in FIG. Here, the 4 kHz level of the minimum audible curve is adjusted to the lowest level corresponding to, for example, 20 bits. In FIG. 10, SB is shown as a horizontal solid line in each block, and MS is shown as a horizontal dotted line in each block. However, in order to avoid complicating the illustration, in FIG. FIG.
At 0, the signal spectrum SS is indicated by a dashed line.

Referring back to FIG. 7, the output of the subtractor 308 is supplied to the allowable noise correction circuit 310. Allowable noise correction circuit 31
A value of 0 corrects the allowable noise level at the output of the subtractor 308 based on, for example, data of an equal loudness curve. That is, the permissible noise correction circuit 310 uses the various parameters, such as the above-described masking and auditory characteristics,
The allocated bits for each unit block are calculated. The output of the allowable noise correction circuit 310 is output via an output terminal 311.
It is output as final output data of the bit allocation calculation circuit 118. Here, the equal loudness curve is a characteristic curve relating to human auditory characteristics. For example, the loudness curve is obtained by calculating the sound pressure of sound at each frequency that sounds as loud as the pure tone of 1 kHz, and connecting the curves with each other. Also called a sensitivity curve.

The equal loudness curve is shown in FIG.
The same curve as the minimum audible curve RC shown in FIG. In this equal loudness curve, for example, around 4 kHz, even if the sound pressure falls by 8 to 10 dB from the place of 1 kHz, it is 1
It sounds the same size as kHz, and conversely, it does not sound the same at around 50 Hz unless it is about 15 dB higher than the sound pressure at 1 kHz. Therefore, if noise exceeding the level of the minimum audible curve RC (allowable noise level) has a frequency characteristic along the equal loudness curve, the noise can be inaudible to humans.

It can be seen that the correction of the allowable noise level in consideration of the equal loudness curve is suitable for human auditory characteristics. As described above, the bit allocation calculation circuit 1
In step 18, data obtained by processing the orthogonal transform output spectrum with the sub-information as main information, and a scale factor indicating a block floating state and a word-lens indicating a word tone as the sub-information are obtained. Based on these pieces of information, the adaptive bit encoding circuit 10 shown in FIG.
6, 107 and 108 perform requantization to generate highly efficient encoded data according to the encoding format.

Returning to FIG. 5, the normalization information change circuit 119 will be described. As described above, by operating the scale factor information determined by the energy calculation circuit 302, it is possible to perform level adjustment for every 2 dB, for example. The normalization information change circuit 119 generates a value related to the change of the scale factor information, and supplies the generated values to the computing units 120, 121, and 122, respectively. The arithmetic unit 120 adds the value supplied from the normalization information change circuit 119 to the scale factor information in the encoded data supplied from the adaptive bit allocation encoding circuits 106, 107 and 108, respectively. I do. However, when the value output from the normalization information change circuit 119 is negative, the arithmetic units 120, 121, and 122 operate as subtractors. The addition result at this time is restricted so as to be within the range of the scale factor value defined in the format.

When the normalization information change circuit 119 outputs the same value to all unit blocks as a value to be added to the scale factor information, the level adjustment processing is performed. If the unit 119 outputs a different value for each unit block, for example, a filtering process or the like can be realized. When performing filter processing, etc.
The normalization information change circuit 119 outputs a set of a value to be added to the scale factor information and a unit block number related to the scale factor information to which the value is to be added. The above-described normalization information adjustment processing can also be realized in the case of decoding described later.

Next, the encoding format of the highly efficient encoded data will be described with reference to FIG. Numerical values 0, 1, 2,..., 211 shown on the left side represent the number of bytes. In this example, 212 bytes are used as a unit of one frame. Block length information of each band determined by the block determination circuits 109, 110, and 111 in FIG. Information on the number of unit blocks to be recorded is recorded at the next byte position. For example, the bit allocation calculation circuit 1
In many cases, the bit allocation is set to 0 by 18 and recording is not required. Therefore, by setting the number of unit blocks so as to cope with such a situation, the number of unit blocks is increased in the middle and low range where the influence on the auditory sense is large. Bit has been made to be distributed. At the same time, in the position of the first byte, the number of unit blocks in which bit allocation information is double-written and the number of unit blocks in which scale factor information is double-written are recorded.

The double writing is a method of recording the same data as the data recorded at a certain byte position in another location for error correction. The greater the amount of double-written data, the higher the strength against errors. However, the smaller the amount of double-written data, the greater the data capacity available for spectrum data. In one example of this encoding format, the number of unit blocks to be double-written is set independently for each of the bit allocation information and the scale factor information, so that it is used to record the strength against an error and the spectrum data. The number of bits is made appropriate. Note that for each piece of information, the correspondence between the number of codes and unit blocks in the prescribed bits is predetermined as a format.

FIG. 12 shows an example of the recorded contents of 8 bits at the position of the first byte. Here, the first three bits are information on the number of unit blocks to be actually recorded, the subsequent two bits are information on the number of unit blocks in which bit allocation information is double-written, and the last three bits are information. This is information on the number of unit blocks for which double writing of scale factor information is performed.

In FIG. 12, bit allocation information of a unit block is recorded at a position from the second byte. For recording bit allocation information, for example, 4 bits are used per unit block. As a result, bit allocation information for the number of unit blocks recorded in order from the 0th unit block is recorded. After the data of the bit allocation information, the scale factor information of each unit block is recorded. For recording scale factor information, for example, 6 bits are used per unit block. As a result, scale factor information for the number of unit blocks recorded in order from the 0th unit block is recorded.

After the scale factor information, the spectrum data in the unit block is recorded. The spectrum data is recorded in order from the 0th unit block in the number of unit blocks to be actually recorded. The number of pieces of spectrum data that exist in each unit block is determined in advance by the format, so that it is possible to correspond to the data by the above-described bit allocation information. Note that recording is not performed on a unit block having a bit allocation of 0.

After the spectrum information, the above-described double writing of the scale factor information and the double writing of the bit allocation information are performed. This double-write recording method is the same as the above-described recording of the scale factor information and the bit allocation information, except that the correspondence between the numbers corresponds to the double-write information shown in FIG. . In the last byte, that is, the 211th byte, and the position before the 1st byte, that is, the 210th byte, the information of the 0th byte and the information of the 1st byte are respectively double-written. The double writing for these two bytes is defined as a format, and variable setting of double writing recording such as double writing of scale factor information and double writing of bit allocation information cannot be performed.

Next, a decoding process for decoding highly efficient encoded data will be described. For example, FIG. 13 shows an example of a configuration of a decoding processing system provided in the audio compression encoder and the expansion decoder 14. The highly efficient encoded data is supplied to the arithmetic unit 710 via the input terminal 707. Also, the block length information used in the encoding process, that is, the output terminals 113, 115, and 117 in FIG.
Is supplied to the input terminal 708. Further, the normalization information change circuit 709 generates a value to be added or subtracted from the scale factor information of each unit block.

The arithmetic unit 710 is further supplied with numerical data from the normalization information change circuit 709. Arithmetic unit 710
Adds the numerical data supplied from the normalization information change circuit 709 to the scale factor information in the supplied high efficiency encoded data. However, when the numerical data supplied from the normalization information change circuit 709 is a negative number, the arithmetic unit 710 acts as a subtractor. The output of the arithmetic unit 710 is supplied to the adaptive bit allocation decoding circuit 706 and the output terminal 711.

The adaptive bit allocation decoding circuit 706 performs a process of releasing bit allocation with reference to the adaptive bit allocation information for each of the high band, the middle band, and the low band. Outputs of the adaptive bit allocation decoding circuit 706 for each of the high band, the middle band, and the low band are output from the inverse orthogonal transform circuits 703, 704, 7
05. Inverse orthogonal transform circuits 703, 704, 7
05 performs an inverse orthogonal transformation process on the supplied data. Thereby, the signal on the frequency axis is converted into a signal on the time axis. The signals on the time axis of the partial bands, which are the outputs of the inverse orthogonal transform circuits 703, 704, 705, are
1, 702 and decoded into a full band signal. As the band combining filters 701 and 702, for example, an IQMF (Inverse Quadrature Mirror filter) or the like can be used.

By operating the scale factor information by addition or subtraction by the arithmetic unit 710, it is possible to adjust the level of the reproduced data, for example, every 2 dB. For example, by outputting the same numerical value from the normalization information changing circuit 709 and uniformly adding or subtracting the numerical value to the scale factor information of all unit blocks,
It is possible to perform level adjustment in units of 2 dB for all unit blocks.

Also, for example, a normalization information change circuit 709
Output the independent numerical values for each unit block, and add or subtract the numerical values to or from the scale factor information of each unit block, so that the level can be adjusted for each unit block, and as a result, the filter function is realized. be able to. More specifically, the normalization information change circuit 709 outputs the unit block number and the unit block and the unit block by a method such as outputting a set of a value to be added to or subtracted from the scale factor information of the unit block. Is associated with the value to be added or subtracted from the scale factor information. Note that the scale factor information generated as a result of the addition or subtraction by the arithmetic unit 710 is limited so that the corresponding scale factor value falls within a range defined by the format of the high-efficiency encoded data.

In the above description, both the encoding circuit and the decoding circuit perform the scale factor information change processing. On the other hand, even when the scale factor information change processing is performed only in the decoding circuit, functions such as level adjustment and filter processing can be sufficiently obtained as a result of the change processing.

Next, a mixing process according to an embodiment of the present invention will be described. FIG. 14 shows an example of the configuration of the mixing circuit 900. A part of the high-efficiency encoded data relating to the music piece a reproduced from the disk 1 is supplied to the adaptive bit allocation decoding circuit 902a via the input terminal 901a.
Supplied to Adaptive bit allocation decoding circuit 902a
It has the same function as the adaptive bit allocation decoding circuit 706 in FIG. 13 and generates spectrum data (MDCT data) by performing decoding corresponding to adaptive bit allocation coding on supplied data. .

The output of the adaptive bit allocation decoding circuit 902a is supplied to an inverse orthogonal transform circuit 903a. The inverse orthogonal transform circuit 903a has the same function as the three inverse orthogonal transform circuits 703, 704, and 705 corresponding to the high band, the middle band, and the low band in FIG. Outputs the data corresponding to. Similarly, a part of the high-efficiency coded data relating to the music b reproduced from the disk 1 is supplied via the input terminal 902b. Adaptive bit allocation decoding circuit 902b,
The inverse orthogonal transform circuit 903b and the memory 904b include an adaptive bit assignment decoding circuit 902a, an inverse orthogonal transform circuit 903a,
Each memory has the same function as the memory 904a, and a part of the high-efficiency coded data related to the music piece b is processed in the same manner as a part of the high-efficiency coded data related to the music piece a.

As highly efficient coded data relating to music b,
High-efficiency encoded data reproduced from a recording medium other than the disk 1 may be used. Inverse orthogonal transform circuits 903a, 90
The output 3b is band-divided digital data, which is data that does not include, as a parameter, a block length related to orthogonal transform in the original high-efficiency coded data. Therefore, according to the above-described method, even when the high-efficiency coded data of the music piece a and the highly efficient coded data of the music piece b have different transform block lengths related to the orthogonal transformation at a desired editing position, mixing and the like are performed. Can be edited.

The memories 904a and 904b store the stored data at an appropriate timing with the multiplier 90
5a and 905b. On the other hand, the multiplication coefficient generation circuit 906 outputs a first multiplication coefficient value to be multiplied by the data of the music piece a and a second multiplication coefficient value to be multiplied by the data of the music piece b. The first multiplication coefficient value is a multiplier 905a
And the second multiplication coefficient value is supplied to a multiplier 905b. The multiplier 905a multiplies the data of the music piece a supplied from the memory 904a by a first multiplication coefficient value for each band, and supplies the multiplication result to the adder 907. Multiplier 90
5b multiplies the data of the music piece b supplied from the memory 904b by a second multiplication coefficient value for each band, and supplies the multiplication result to the adder 907.

The adder 907 adds the output of the multiplier 905a and the output of the multiplier 905b for each band. That is, the multiplied value related to the high frequency data of the music a and the multiplied value related to the high frequency data of the music b are added, and the multiplied value related to the middle frequency data of the music a and the multiplied value related to the middle frequency data of the music b Is added, and the multiplied value of the low frequency data of the music piece a and the multiplied value of the low frequency data of the music piece b are added. In this way,
Data on the time axis subjected to addition processing for each band is created. When an added value exceeding the upper or lower limit of the data size is obtained, the adder 907 may perform a limiter process for setting an upper or lower limit value instead of the added value. .

The output of the adder 907 is supplied to a block decision and orthogonal transformation circuit 908 and a band synthesis filter 911.
Supplied to Block decision and orthogonal transformation circuit 908
Are the three block decision circuits 109, 110,
111, and three MDCT circuits 103, 104, 1
It has a function equivalent to 05, that is, a function of determining a block length corresponding to high-, middle-, and low-frequency data and performing orthogonal transformation. Therefore, the output of the block determination and orthogonal transformation circuit 908 is spectrum data (MDCT data) on the frequency axis.

The output of the block decision and orthogonal transformation circuit 908 is supplied to an adaptive bit allocation encoding circuit 909. The adaptive bit allocation encoding circuit 909 has a function equivalent to the three adaptive bit allocation encoding circuits 106, 107, and 108 in FIG. 5, that is, a function of performing adaptive bit allocation corresponding to the high band, the middle band, and the low band. , And the output of the adaptive bit allocation encoding circuit 909 becomes highly efficient encoded data. The highly efficient encoded data is data corresponding to the result of the mixing process between the music a and the music b.
The continuity is maintained with the highly efficient coded data corresponding to any of b. For this reason, for both music a and b,
The high-efficiency encoded data output from the output terminal 908 is
It is possible to replace the corresponding part.

The output of the adaptive bit allocation encoding circuit 909 is recorded on the recording medium via the output terminal 910, so that the edited result is recorded. The output destination of the output terminal 908 may be the recording processing system of the disk 1 which is the input source of the input terminal 901a, or may be the recording processing system of another recording medium. Similarly, the output destination of the output terminal 908 may be the input source of the input terminal 901b or a recording processing system related to another recording medium. With the above configuration, a process of mixing the music a and the music b and recording the result of the mixing is realized.
As a recording medium at this time, in addition to a magneto-optical disk such as an MD, a disk-shaped recording medium such as a magnetic disk, a tape-shaped recording medium such as a magnetic tape or an optical tape, or an IC memory or a memory card may be used. it can.

The band synthesizing filter 911 is arranged as shown in FIG.
It has a band synthesizing function similar to the band synthesizing filters 702 and 703, and generates PCM samples by band synthesizing supplied data. The generated PCM sample is output via the output terminal 912. If the output terminal 912 is configured to output the output to the D / A converter 15 in FIG. 2, for example, it is possible to generate a sound corresponding to the edited result. This makes it possible to realize a configuration and a processing procedure in which the user can listen to and check the editing result and then perform an operation for performing desired mixing.

Separate decoding circuits may be provided as adaptive bit allocation decoding circuits 902a and 902b, respectively, or the decoding circuits 902a and 902b may be operated by time-sharing one adaptive bit allocation decoding circuit. And 9
The function as 02b may be realized. Also, separate input terminals may be provided as the input terminals 901a and 901b, or the songs a and b may be supplied in a time-division manner via one input terminal.

Also, the inverse orthogonal transform circuits 903a and 903a
As 3b, separate inverse orthogonal transform circuits may be provided, respectively, or one inverse orthogonal transform circuit may be operated as 903a and 903b by processing such as operating in a time-division manner. Also, the memories 904a and 904
As the memory 04b, separate memories may be provided, or memory management may be performed such that the same memory operates as 904a and 904b. The multiplication coefficient generation circuit 907 may be configured using, for example, a ROM or the like, and may store a value set in advance as a multiplication coefficient value. The value set in ROM is
For example, a value input by a user may be used.

Various shapes of crossfades can be realized according to the multiplication coefficient value output from the multiplication coefficient generation circuit 906. For example, when it is desired to obtain a mixing result such that the music a and the music b are linearly cross-fade, the multiplication coefficient value for the music a and the multiplication coefficient value for the music b are k and (1-k), respectively. It is good. Here, k is a value of 0 or more and 1 or less. Further, in addition to a linear crossfade, a crossfade having a shape such as a sine curve can be realized. 14 can be used as the components in FIG. 14 by controlling the operation in a time-division manner.

Next, an example of the cross-fade processing realized by one embodiment of the present invention will be described. One example of this is the N-th and N-times encoded with high efficiency recorded on a recording medium such as an MD as shown in FIG. 15A.
This is a process of cross-fading between the + 1st music (track) at a time interval T as shown in FIG. 15B. T is set at a time interval desired by the user, for example, about 2 seconds to 5 seconds. Set T
Is converted into the number of frames corresponding to 11.6 msec, and data corresponding to the converted number of frames is input to the mixing circuit 900 shown in FIG.

That is, as shown in FIG. 16, data of the number of frames corresponding to the time interval to be cross-fade from the end of the Nth music piece is selected as the part N. Here, the number of frames selected as the part N is four. Also, data of the number of frames corresponding to the time interval to be cross-fade from the head of the (N + 1) th music piece is selected as the part N + 1. The number of frames selected as part N + 1 is also set to four, and part N and part N + 1
Respectively correspond to the time intervals at which the crossfade is to be performed. Parts N and N + 1 are input terminals 901a and 901 of mixing circuit 900, respectively.
b.

In the mixing circuit 900, the processing described above with reference to FIG. 14 is performed. At this time, as an example of the multiplication coefficient generated by the multiplication coefficient generation circuit 906, a value that changes linearly in the case of a cross fade can be used as described later with reference to FIG. Also,
A value that changes to a shape such as a sine curve or a log curve can also be used as a multiplication coefficient. Various mixing results can be obtained depending on the multiplication factor used.
Further, a fade shape may be set in advance for the high-efficiency encoded information using the normalization information adjusting circuit 119 in FIG. 5 and the multiplication coefficient may be always set to 1 and added using this. .

Further, by using a configuration and a processing procedure which enable the user to perform the above-described parameter input operation by listening to the sound generated based on the output of the output terminal 911, Is also good. A mixing process of mixing two songs (tracks) at an arbitrary ratio without performing a crossfade,
A mixing process or the like may be performed to form a new one frame in which the end and start portions of two music pieces are continuously played.

The portion corresponding to the result of the mixing process (the portion of four frames shown by hatching in FIG. 15) is output from the output terminal 910. This data portion can be replaced with a portion corresponding to the last time interval T of the N-th song (in the example of FIG. 15, four frames).
A portion corresponding to the (N + 1) -th first time interval T (FIG. 15)
(In one example, four frames).

Here, for example, in the case of a recording medium such as an MD which manages track information in an area such as U-TOC different from the area of general recording data, the Nth or (N + 1) th recording medium is used. With the above-described replacement of the frame corresponding to the time interval T of the music, a process of rewriting the track information is performed. With the above processing, the creation and recording of the high-efficiency encoded data that has been subjected to the mixing processing are executed.

For example, the personal computer 40 in FIG. 1 can be used as a configuration for receiving an input from a user relating to the above-described mixing processing.
That is, an input screen relating to mixing is displayed on a monitor attached to the personal computer 40 as a type of editing operation, and parameter input is received by a so-called drag-and-drop operation via a pointing device such as a mouse. Just do it.

The parameters input at this time are as follows:
The number of the music (track) related to the mixing process, the time interval T related to the mixing process, a multiplication coefficient, and the like are included. More specifically, for example, the Nth track and N
The + 1st track is displayed as two rod-like figures, one or both of which move to slide according to the operation of the user, and the time interval T
May be used as an input display screen in which is set.

Regarding the input of the multiplication coefficient, a diagram of the multiplication coefficient as shown in FIG. 17 may be displayed so that the user can perform more precise designation. FIG.
Is a diagram in which the horizontal axis indicates time and the vertical axis indicates multiplication coefficients for the music pieces a and b, showing an example of a temporal change in the multiplication coefficient. This example relates to a case where the music piece a is faded out along the straight line Ca and the music piece b is faded in along the straight line Cb during the period T.

Further, by replacing data as a result of processing such as cross-fade with original recording data, it is possible to delete a data portion that is less important to the user and increase the recordable capacity of the recording medium. It is. Thereby, the convenience at the time of recording can be improved. In addition, if the recordable capacity or the increase in the recordable capacity accompanying the editing process is displayed on an input screen of a monitor or the like attached to the personal computer 40, the convenience for the user is further improved. Becomes possible.

Further, an operation panel and a display monitor may be provided so as to be attached to the housing of the apparatus main body or the like, so that input operation of parameters relating to the mixing process may be performed by an input device such as a jog dial. .

In the above-described embodiment of the present invention, the mixing process is performed between two music pieces (tracks) that have been encoded with high efficiency. It is also possible to perform a mixing process between two or more music pieces (tracks).

Further, in one embodiment of the present invention, the editing process is performed between two music pieces (tracks) that have been encoded with high efficiency.
It is also possible to perform an editing process between the data and, for example, data in the form of a PCM sample. Specifically, for example, FIG.
In the configuration shown in FIG. 4, the above-described processing is performed by inputting the music data encoded at high efficiency via the output terminal 901a and inputting the music data in the form of the PCM sample divided into bands into the memory 904b. Such processing can be performed. However, the band division here needs to be equivalent to the band division at the time of encoding the high-efficiency encoded data input via the output terminal 901a.

That is, the present invention can be applied to a case where mixing is performed between a plurality of data including at least one piece of high efficiency encoded data. In addition, in general, existing digital effect processing or the like can be performed on the data stored in the memory when the phase relationship is maintained.

The above-described embodiment of the present invention employs ATRAC (Adapytive Transform Acoustic C) which is a compression encoding method used when MD is used as a recording medium.
oding) method. On the other hand, as a compression encoding method, MPEG (Moving Picture Expert
The present invention can also be applied to a case where SBC such as audio layer 1 and MPEG audio layer 2 is assumed.

The present invention is not limited to the embodiment of the present invention described above, and various modifications and changes can be made without departing from the gist of the present invention.

[0116]

According to the present invention, for example, arithmetic processing such as weighted addition is performed for each band of data obtained by performing processing such as inverse quantization on a part of two high-efficiency coded data for each band. The data generated as a result of the arithmetic processing is subjected to processing such as quantization, thereby performing editing processing such as mixing.

Therefore, it is possible to generate high-efficiency coded data that maintains continuity with the original high-efficiency coded data without performing a special phase correction process. In addition, there is no need to perform band division filter processing, band synthesis filter processing, and the like during editing processing.

Therefore, since the editing process is simplified, the processing time can be reduced, and the data quality such as sound quality is not degraded due to the editing process, or the degree thereof is minimized. Can be.

Further, since the arithmetic processing is performed on the data subjected to the inverse orthogonal transformation, the editing processing can be executed without considering the transformation block length related to the orthogonal transformation in the original high-efficiency coded data.

[Brief description of the drawings]

FIG. 1 is a block diagram illustrating an example of a configuration of an audio signal processing device including an MD recorder to which the present invention can be applied.

FIG. 2 is a block diagram illustrating an example of a configuration of an MD recorder in FIG.

FIG. 3 is a schematic diagram for explaining an MD medium format.

FIG. 4 is a schematic diagram for explaining an operation in the audio signal processing device shown in FIG. 1;

FIG. 5 is a block diagram illustrating an example of a configuration related to generation of highly efficient encoded data.

FIG. 6 is a schematic diagram for explaining an orthogonal transform block length for each band.

FIG. 7 is a block diagram showing a part of the configuration in FIG. 5 in detail.

FIG. 8 is a schematic diagram illustrating an example of a spectrum of a band divided in consideration of a critical band, block floating, and the like.

FIG. 9 is a schematic diagram illustrating an example of a masking spectrum.

FIG. 10 is a schematic diagram for explaining synthesis of a minimum audible curve and a masking spectrum.

FIG. 11 is a schematic diagram illustrating an example of an encoded data format according to an embodiment of the present invention.

FIG. 12 is a schematic diagram illustrating details of data of a first byte in FIG. 11;

FIG. 13 is a block diagram illustrating an example of a configuration related to digital signal decoding processing.

FIG. 14 is a block diagram illustrating an example of a configuration related to mixing.

FIG. 15 is a block diagram for describing an editing position.

FIG. 16 is a schematic diagram for describing an editing position in detail.

FIG. 17 is a schematic diagram illustrating an example of a multiplication coefficient.

[Explanation of symbols]

900 ... mixing circuit, 902a, 902b ...
.Adaptive bit assignment decoding circuits 903a and 903b.
· Inverse orthogonal transform circuit, 906 ··· Multiplication coefficient generation circuit, 9
05a, 905b ... multiplier, 907 ... adder,
908... Block determination and orthogonal transformation circuit 909
... Adaptive bit allocation coding circuit

Continuation of the front page F term (reference) 5D044 AB05 DE13 GK08 HL14 5J064 AA02 BA16 BB05 BC01 BC02 BC06 BC07 BC08 BC09 BC29 BD03 9A001 BB03 EE04 GG17 HH15 JJ07 KK54

Claims (23)

[Claims] 1. A method for generating a plurality of signal components by dividing an input digital signal into a plurality of frequency components, and generating a plurality of bits for each of a plurality of blocks specified in relation to time and frequency based on the generated signal components. Digital signal processing for determining the allocation amount and referring to the information compression parameter including the determined bit allocation amount to perform an editing process on the highly efficient encoded data generated by quantizing the signal component. In the device, a part of the high-efficiency coded data is dequantized, an arithmetic process is performed based on the result of the dequantization, and the data generated as a result of the arithmetic process is quantized to convert the highly-efficient coded data into A digital signal processing device for generating. 2. An input digital signal is divided into a plurality of frequency band components, a signal component on a frequency axis is generated by orthogonally transforming the band-divided signal, and time and frequency are calculated based on the generated signal component. Along with determining the bit allocation amount for each of a plurality of blocks specified in association with each other,
A digital signal processing device that performs editing processing on high-efficiency coded data generated by quantizing the signal component with reference to the information compression parameter including the determined bit allocation amount. A part of the data is inversely quantized, the result of the inverse quantization is inversely orthogonally transformed, the arithmetic processing is performed on the result of the inverse orthogonal transformation, and the data generated as a result of the arithmetic processing is orthogonally transformed. A digital signal processing device for generating highly efficient encoded data by quantizing a result. 3. The digital signal processing device according to claim 1, wherein a part of the high efficiency encoded data corresponds to a period designated by a user. 4. The method according to claim 1, wherein the calculation process is a process of multiplying data of each band in the result of the inverse quantization by a multiplication coefficient, and adding the multiplication result for each band. Digital signal processor. 5. The digital signal processing device according to claim 4, wherein the time change of the multiplication coefficient is set by a user. 6. The first and second multiplication methods according to claim 1, wherein the first and second high-efficiency coded data are dequantized in part and the result of the dequantization is multiplied by a multiplication coefficient. A digital signal processing device that generates a result and performs a process of adding the first multiplication result and the second multiplication result. 7. The method according to claim 2, wherein a part of the first and second high-efficiency coded data is inversely quantized, an inverse orthogonal transform is performed on the result of the inverse quantization, and each band in the inverse orthogonal transform result is obtained. A first multiplication result is generated by multiplying the first multiplication data by a multiplication coefficient, and a process of adding the first multiplication result and the second multiplication result is performed. Signal processing device. 8. The digital signal processing device according to claim 6, wherein the first and second high-efficiency encoded data are recorded on the same recording medium. 9. The method according to claim 1, wherein the first and second dequantization means are provided with two dequantization means for performing dequantization, or one dequantization means is operated in a time-division manner. 2. A digital signal processing device, wherein a part of each of the high efficiency encoded data is dequantized. 10. The method according to claim 2, wherein two inverse orthogonal transform means for performing an inverse orthogonal transform are provided, or one inverse orthogonal transform means is operated in a time-division manner. A digital signal processing device for performing an inverse orthogonal transform on a result of inverse quantization of each part of highly efficient encoded data. 11. The digital signal according to claim 1, wherein a first multiplication result is generated by dequantizing a part of the high-efficiency encoded data and multiplying the result of the inverse quantization by a multiplication coefficient. A second multiplication result is generated by multiplying data obtained by performing the same frequency band division as when encoding the highly efficient encoded data by a multiplication coefficient, and the first multiplication result and the second multiplication result are calculated. A digital signal processing device for performing a process of adding a result of multiplication of the digital signal. 12. The method according to claim 2, wherein a part of the high-efficiency coded data is inversely quantized, an inverse orthogonal transform is performed on a result of the inverse quantization, and data of each band in the result of the inverse orthogonal transform is multiplied. A first multiplication result is generated by multiplying a coefficient, and data obtained by subjecting a digital signal to the same frequency band division as when encoding the high-efficiency coded data is multiplied by a multiplication coefficient. A digital signal processing device for generating a second multiplication result and performing a process of adding the first multiplication result and the second multiplication result. 13. A plurality of signal components are generated by dividing an input digital signal into a plurality of frequency components, and bits for each of a plurality of blocks specified in relation to time and frequency based on the generated signal components. Digital signal processing for determining the allocation amount and referring to the information compression parameter including the determined bit allocation amount to perform an editing process on the highly efficient encoded data generated by quantizing the signal component. In the method, a part of the highly efficient encoded data is dequantized, an arithmetic processing is performed based on the result of the inverse quantization, and the data generated as a result of the arithmetic processing is quantized to convert the highly efficient encoded data. A digital signal processing method characterized by generating. 14. An input digital signal is divided into a plurality of frequency band components, a signal component on a frequency axis is generated by orthogonally transforming the band-divided signal, and time and frequency are converted based on the generated signal component. A high efficiency generated by quantizing the signal component with reference to an information compression parameter including the determined bit allocation amount while determining the bit allocation amount for each of the plurality of blocks specified in association with each other. A digital signal processing method that edits encoded data. In the digital signal processing method, a part of high-efficiency encoded data is inversely quantized, the result of inverse quantization is subjected to inverse orthogonal transform, and arithmetic processing is performed on the result of inverse orthogonal transform. Performing orthogonal transformation of data generated as a result of the arithmetic processing, and quantizing the result of the orthogonal transformation to generate highly efficient encoded data. Digital signal processing method. 15. A plurality of signal components are generated by dividing an input digital signal into a plurality of frequency components, and bits for each of a plurality of blocks specified in relation to time and frequency based on the generated signal components. Determines the allocation amount, refers to the information compression parameter including the determined bit allocation amount, performs an editing process on the highly efficient encoded data generated by quantizing the signal component, and performs an editing process. In a digital signal recording device that records results, a part of high-efficiency encoded data is inversely quantized, arithmetic processing is performed based on the result of the inverse quantization, and data generated as a result of the arithmetic processing is quantized. A digital signal recording apparatus for generating high-efficiency coded data and recording the generated high-efficiency coded data. 16. An input digital signal is divided into a plurality of frequency band components, a signal component on a frequency axis is generated by orthogonally transforming the band-divided signal, and time and frequency are converted based on the generated signal component. A high efficiency generated by quantizing the signal component with reference to an information compression parameter including the determined bit allocation amount while determining a bit allocation amount for each of a plurality of blocks specified in association with each other. Edit the encoded data,
In a digital signal recording device that records the result of editing processing, a part of the high-efficiency encoded data is dequantized, the result of inverse quantization is subjected to inverse orthogonal transform, and the result of the inverse orthogonal transform is subjected to arithmetic processing. Digital signal recording characterized by orthogonally transforming data generated as a result of processing, generating high-efficiency encoded data by quantizing the result of the orthogonal transformation, and recording the generated high-efficiency encoded data apparatus. 17. The digital signal recording device according to claim 15, wherein the recording medium is a disk-shaped recording medium. 18. The digital signal recording apparatus according to claim 15, wherein the recording medium is a tape-shaped recording medium. 19. The digital signal recording device according to claim 15, wherein the recording medium is a memory card. 20. A method for generating a plurality of signal components by dividing an input digital signal into a plurality of frequency components, and generating a plurality of bits for each of a plurality of blocks specified in relation to time and frequency based on the generated signal components. Determines the allocation amount, refers to the information compression parameter including the determined bit allocation amount, performs an editing process on the highly efficient encoded data generated by quantizing the signal component, and performs an editing process. In a digital signal recording method for recording a result, a part of high-efficiency coded data is dequantized, an arithmetic processing is performed based on a result of the inverse quantization, and data generated as a result of the arithmetic processing is quantized. A digital signal recording method comprising: generating high-efficiency encoded data according to the above; and recording the generated high-efficiency encoded data. 21. An input digital signal is divided into a plurality of frequency band components, a signal component on a frequency axis is generated by orthogonally transforming the band-divided signal, and time and frequency are calculated based on the generated signal component. A high efficiency generated by quantizing the signal component with reference to an information compression parameter including the determined bit allocation amount while determining the bit allocation amount for each of the plurality of blocks specified in association with each other. Edit the encoded data,
In a digital signal recording method for recording the result of editing processing, a part of high-efficiency encoded data is inversely quantized, the result of inverse quantization is subjected to inverse orthogonal transform, and the result of the inverse orthogonal transform is subjected to arithmetic processing. Digital signal recording characterized by orthogonally transforming data generated as a result of processing, generating high-efficiency encoded data by quantizing the result of the orthogonal transformation, and recording the generated high-efficiency encoded data Method. 22. A plurality of signal components are generated by dividing an input digital signal into a plurality of frequency components, and bits for each of a plurality of blocks specified in relation to time and frequency based on the generated signal components. While determining the allocation amount, referring to the information compression parameter including the determined bit allocation amount, an editing process result obtained by editing the high-efficiency coded data generated by quantizing the signal component is obtained. In the recorded recording medium, a part of the high-efficiency coded data is dequantized, the arithmetic processing is performed based on the result of the dequantization, and the data generated as a result of the arithmetic processing is quantized. Recording medium on which highly efficient encoded data is recorded. 23. An input digital signal is divided into a plurality of frequency band components, a signal component on a frequency axis is generated by orthogonally transforming the band-divided signal, and time and frequency are calculated based on the generated signal component. A high efficiency generated by quantizing the signal component with reference to an information compression parameter including the determined bit allocation amount while determining the bit allocation amount for each of the plurality of blocks specified in association with each other. On a recording medium on which the edited processing result obtained by editing the encoded data is recorded, a part of the high-efficiency encoded data is inversely quantized, the inverse quantization result is subjected to inverse orthogonal transform, and the inverse orthogonal transform result is obtained. High-efficiency coded data generated by performing arithmetic processing on the data, orthogonally transforming the data generated as a result of the arithmetic processing, and quantizing the result of the orthogonal transformation is recorded. Recording medium, characterized in that it is.