JPH10303666A - Record medium for processing digital audio signal, communication method for digital audio signal and digital audio signal record medium - Google Patents

Abstract

PROBLEM TO BE SOLVED: To extend the attack time by decreasing a delay time of an input signal so as to prevent echo. SOLUTION: A digital audio signal processing circuit is provided with a gain generating processing section 4 that generates a gain for compress/limit based on a level of a digital audio signal and with a smooth filter processing section 5 that provides an output of a time constant to the compress/limit gain outputted from the gain generating processing section 4. The smooth filter processing section 5 decreases the gain comparatively steeply a gain with the provided time constant more than that of a conventional circuit, the time constant of the gain provided by the smooth filter processing section 5 is extended by a time constant extension processing section 5 and the resulting time constant is set to the gain adjustment processing section 2. Furthermore, a compress/limit processing program and an audio signal processed by the program are transmitted via a network NW.

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a recording medium for processing a digital audio signal on which a program for compressing and limiting the level of a digital audio signal is recorded, and particularly when the signal level rises sharply. The present invention relates to a recording medium for digital audio signal processing in which a program for attenuating the sound sensation by gently attenuating it from the time of its rise is recorded. The present invention also provides
The present invention relates to a program for compressing and limiting the level of a digital audio signal, and a communication method for a digital audio signal for transmitting a digital audio signal processed by the program via a communication line. The present invention also relates to a digital audio recording medium on which a level of a digital audio signal is optimally compressed and compressed. The present invention also relates to a digital audio recording medium for recording a digital audio signal after optimally compressing (compressing) and limiting (restricting) the level thereof, and further reproducing the digital audio signal in accordance with a user's preference.

[0002]

2. Description of the Related Art Conventional compress / limit circuits are disclosed in, for example, JP-A-3-198413 and JP-A-6-198413.
For example, there is one disclosed in JP-A-164275. FIG.
FIG. 21A schematically shows such a conventional compress / limit circuit. FIG. 21A shows signals inputted to the delay circuit 1 and the level detection circuit (level detection processing) 3.
(B) is a signal delayed by the delay circuit 1, FIG.
(C) is a smooth filter (smooth filter processing) 5
21D shows an example of an output signal of the gain adjustment amplifier (gain adjustment process) 2 set in the gain adjustment amplifier (gain adjustment process) 2.

[0003] Digital audio signal in FIG. 20 is delayed by a minute time t ₁ by the delay circuit 1, then the gain is adjusted based on the gain control signal G by the gain adjustment amplifier (gain adjustment processing) 2. In this case, the gain G is normally set to “1”, and when the compress / limit function is set, when the signal level rises steeply, the gain G is set to “1” so as to attenuate gradually from the rise. Is set to a value that gradually decreases from

For example, when a steeply rising signal A, for example, a 1 kHz burst wave is input as shown in FIG.
Is detected by Then, the gain generation circuit 4 compares the detection level with a threshold Vth for compression / limit, and when the detection level is exceeded, the difference Δ (= a−Vt)
Generate a compressed gain g (Δ) according to h) and output it to the smoothing filter 5. Smooth filter 5
Is an IIR filter (low-pass filter) composed of a coefficient multiplier, a delay unit, and an adder, and has a time constant corresponding to a set coefficient and a gain that gradually decreases from “1” as shown by the curve in FIG. G is set to the gain adjustment amplifier (gain adjustment processing) 2.

Therefore, the signal b delayed by the delay circuit 1 is attenuated by the gain G that gradually decreases from “1” to, for example, 0.2 from the vicinity of the rising point,
The attenuation rate (time constant) of the gain G is a value corresponding to the set coefficient. Here, when the attack point to the attenuation starts of the gain G of the signal b, called the attack time period to be damped, the delay time t ₁ is about twice the attack time t _2, or more of the required ing.

FIG. 22 is a block diagram showing a general surround audio encoder. In this example, four systems of the Dolby surround system, that is, L, C, S,
An encoder that encodes a multi-channel signal composed of four R signals and converts it into two-channel signals Lt and Rt is shown. This encoder is an adder 9
1, 93, 97, 100, attenuators 92, 94, and BP
An F95, a noise reduction encoder 96, and phase shifters 98 and 99 are provided. The attenuators 92 and 94 are 3d
B, and the pass band of the BPF 95 is 1
00 Hz to 7 kHz. The phase shifter 98 is 90 °
Phase shift (+ 90 °), while the phase shifter 99
Is to retard by 90 ° (−90 °).

FIG. 23 shows two systems of signals Lt and Rt by the encoder and the like shown in FIG. 22, decodes the recorded or transmitted signal and decodes the original multi-channel signal (4 channels L = Lt, C in this example). = Lt + Rt, S
= Lt-Rt, R = Rt). This decoder
After the input two-channel signals Lt and Rt are decoded into four channels, each of these four channel signals (L
The level change of t, Rt, Lt + Rt, Lt-Rt) is monitored by the control circuit 78, and if any one of the signal levels becomes extremely large, the signal levels of the other three channels are attenuated or reduced. And the gains of the corresponding amplifiers 74 to 77 are controlled. The signals G _L , G _R , G _C ,
G _S is a gain control signal supplied from the control circuit 78 to each of the amplifiers 74 to 77. This gain control provides a surround sound listener with a sound field with a clearer sense of direction. For this reason, the control circuit 78 is configured to supply an output signal that attenuates the gain of each amplifier by 0 to 30 dB.

[0008]

However, in the conventional compress / limit circuit shown in FIG. 20, the attack time t _{2 is set} so that the signal level attenuates gently.
In order to increase the delay time, the delay time t ₁ of the input signal must be lengthened, so that the output signal becomes a signal like an echo having a long time delay, which is rather annoying. If the delay time t ₁ of the input signal is reduced to solve the problem of the echo, there is a problem that the attack time t ₂ cannot be increased due to the relationship of t ₁ ≧ 2t ₂ . Recently, the use of personal computers (hereinafter referred to as personal computers or PCs) for multimedia has been rapidly progressing, and moving image and audio signals have been transmitted to P
Processing by C is widespread. Recently, transmission of moving image and audio signals to a data transmission side and a data reception side via a communication line such as the Internet has become widespread. Further, there is a demand for a digital audio recording medium capable of reproducing a compressed / limited audio signal optimally for a user.

In view of the above problems, the present invention provides a digital audio signal processing program capable of shortening the delay time of a digital audio signal, preventing echo and extending the attack time. It is a first object to provide a recorded recording medium. In the present invention, when performing compression / limit characteristic processing on a plurality of signals, it is possible to shorten the delay time of the input signal while balancing the signal levels of the plurality of signals, thereby preventing an echo and preventing an attack. It is a second object of the present invention to provide a recording medium in which a digital audio signal processing program capable of increasing the time is recorded. In the present invention, when performing compression / limit characteristic processing on signals of a plurality of systems for multi-channel reproduction, a common level is detected from signals of a small number of systems before decoding, so that a multi-channel signal level is reduced. It is a third object of the present invention to provide a recording medium on which a digital audio signal processing program capable of shortening the delay time of an input signal and preventing an echo while maintaining a balance and increasing the attack time is recorded. Aim. The present invention also provides a communication method for a digital audio signal for transmitting, via a communication line, a program capable of shortening the delay time of the digital audio signal, preventing echo, and increasing the attack time. 4th to offer
The purpose of. The present invention also provides a digital audio signal for transmitting, via a communication line, a digital audio signal processed by a program capable of shortening the delay time of the digital audio signal to prevent echo and increasing the attack time. A fifth object is to provide a communication method for audio signals. A sixth object of the present invention is to provide a digital audio recording medium capable of shortening the delay time of a digital audio signal, preventing echo and lengthening the attack time. A seventh object of the present invention is to provide a digital audio recording medium capable of shortening a delay time of a digital audio signal according to a user's preference to prevent an echo and extending an attack time. And

[0010]

According to the present invention, in order to achieve the first object, a time constant of a gain is extended by a digital audio signal processing program. That is, according to the present invention, a level detecting step for detecting the level of a digital input signal, and a gain generating step for generating a time-variable compress / limit gain based on the level detected by the level detecting step A time constant extending step of extending a time constant of the gain generated by the gain generating step; and an attenuation step of attenuating the digital input signal based on the gain whose time constant has been extended by the time constant extending step. , A recording medium for digital audio signal processing in which a program is recorded.

According to the present invention, there is provided a level detecting step for detecting a level of a digital input signal and a level of a signal obtained by delaying the digital input signal, and a temporal detection based on the level detected by the level detecting step. A gain generation step of generating a variable compression / limit gain, a time constant extension step of extending a time constant of the gain generated by the gain generation step, and a time constant extended by the time constant extension step. An attenuating step of attenuating a signal obtained by delaying the digital input signal based on the gain obtained.

According to the present invention, in order to achieve the second object, a digital audio signal processing program extends a gain time constant to attenuate a plurality of digital input signals. That is, according to the present invention, a level detecting step of detecting each level of the digital input signals of a plurality of systems, and detecting an average value or a maximum value of the levels of the digital input signals of the multiple systems detected by the level detecting step A detection step; a gain generation step of generating a temporally variable compression / limit gain based on an average value or a maximum value of each level detected by the detection step; A time constant extending step of extending a time constant of a gain, and an attenuation step of attenuating a signal in which each of the plurality of digital input signals is delayed based on the gain whose time constant has been extended by the time constant extending step. , A recording medium for digital audio signal processing in which a program is recorded.

Further, according to the present invention, a level detecting step for detecting each level of a plurality of digital input signals and a level of a signal obtained by delaying the plurality of digital input signals; A gain generation step of generating a time-variable compression / limit gain based on the average value or the maximum value of each level obtained, and a time constant for extending a time constant of the gain generated by the gain generation step A digital audio signal processing recording medium in which a program having an extension step and an attenuation step of attenuating each of the plurality of digital input signals based on the gain whose time constant has been extended by the time constant extension step is recorded. Is provided.

According to the present invention, in order to achieve the third object, a digital input signal is generated from a plurality of digital input signals, and a digital audio signal processing program is used to generate a gain time constant. Is extended to attenuate this multi-system digital input signal. That is, according to the present invention, a level detecting step of detecting each level of the digital input signals of a plurality of systems, and detecting an average value or a maximum value of the levels of the digital input signals of the multiple systems detected by the level detecting step A detection step; a gain generation step of generating a temporally variable compression / limit gain based on an average value or a maximum value of each level detected by the detection step; A time constant extension step for extending the gain time constant,
A decoding step of generating a digital signal of a further system based on the digital input signals of a plurality of systems, and an attenuation for attenuating each of the digital signals of the system based on a gain whose time constant has been extended by the time constant extending step. And a recording medium for digital audio signal processing on which is recorded a program having the following steps:

Further, according to the present invention, a level detecting step for detecting each level of the digital input signals of a plurality of systems, and an average value or a maximum value of each level of the digital input signals of the plurality of systems detected by the level detecting step A detection step of detecting a value, a gain generation step of generating a temporally variable compression / limit gain based on an average value or a maximum value of each level detected by the detection step, A program having a decoding step of generating a multi-system digital signal based on an input signal and an attenuating step of attenuating each of the multi-system digital signals based on a gain generated by the gain generating step is recorded. A recording medium for digital audio signal processing is provided.

Further, in order to achieve the fourth object, the present invention transmits a program for achieving the first to third objects via a communication line or a network. That is, according to the present invention, for the program data according to any one of claims 1 to 9,
Adding a header for transmission via a communication line or a network, converting the program data to which the header has been added into a digital signal sequence for a communication line or a network, and converting the digital signal sequence to a communication line or Transmitting to a receiving side via a network.

According to the present invention, claims 1 to 9 are provided.
Adding a header for transmission via a communication line or a network to the program data according to any one of the above, and converting the program data to which the header is added into a digital signal sequence for a communication line or a network A communication method for a digital audio signal for receiving the digital signal sequence when the digital signal sequence is transmitted via a communication line or a network, and receiving the digital signal sequence based on the header, Separating the program data from the digital signal sequence.

According to the present invention, in order to achieve the fifth object, a digital audio signal processed by a program for achieving the first to third objects is transmitted via a communication line or a network. It was done.
That is, according to the present invention, a step of processing an audio signal based on the program according to any one of claims 1 to 9, and a communication line or a network for the audio signal processed based on the program. Adding a header for transmission over
Converting the audio data to which the header has been added into a digital signal sequence for a communication line or a network, and transmitting the digital signal sequence to a receiving side via a communication line or a network. Is provided.

According to the present invention, an audio signal is processed based on a program according to any one of claims 1 to 9, and a communication line is provided for the audio signal processed based on the program. Or adding a header for transmission over a network, converting the audio data with the header added to a digital signal sequence for a communication line or network, and transmitting the digital signal sequence via a communication line or network. A communication method for a digital audio signal to receive the digital signal sequence in the case of receiving a digital signal sequence based on the header; and separating and reproducing an analog audio signal from the digital signal sequence. A communication method for digital audio signals having the following is provided.

According to the present invention, in order to achieve the sixth object, a digital audio signal processed by the above program is recorded on a digital audio recording medium. That is, according to the present invention, there is provided a digital audio recording medium in which an audio signal is processed and recorded based on the program according to any one of claims 1 to 6.

According to the present invention, in order to achieve the seventh object, both a digital audio signal processed by the above program and the above program are recorded on a digital audio recording medium. That is, according to the present invention, an audio signal is processed and recorded based on the program according to any one of claims 1 to 6, and the program according to any one of claims 1 to 9 is recorded. Digital audio recording medium is provided.

[0022]

Embodiments of the present invention will be described below with reference to the drawings. FIG. 1 is a block diagram showing a digital audio signal processing system to which an embodiment of a digital audio signal processing recording medium according to the present invention is applied;
2 is a flowchart for explaining a digital audio signal processing program of the personal computer in FIG. 1, FIG. 3 is a functional block diagram for explaining in detail a compress / limit process in the decoding process in FIG. 2, and FIG. 3 is a waveform diagram showing an attenuation characteristic by the compression / limit processing of FIG. 3, FIG. 5 is a functional block diagram for explaining an example of the smooth filter processing and the time constant extension processing of FIG. 3 in detail,
FIG. 6 is a functional block diagram for explaining in detail another example of the smooth filter processing and the time constant extension processing of FIG.

The personal computer 106 shown in FIG. 1 has a digital audio signal processing program as shown in FIG. 2 from the disk drive 104 or the network terminal 105 (compress / limit processing 13 shown in detail in FIG. 3 is included in the decoding processing). And music sources such as surround music.

The personal computer 106 is, for example, an Intel PP
CPU with an application-specific instruction set, such as the 55C Extended Instruction Set (MMX), added primarily to efficiently process digital signals such as images and audio
106a, a RAM 106b used as a buffer at the time of data processing, a data converter 106c for converting data supplied from the disk drive 104 or the network terminal 105, and the processed audio data via a D / A converter and an amplifier. Audio interface (I / F) 106d for supplying a plurality of speakers (103L and 103R in the figure, and surround speakers 103C and 103S), and a display processor 106 for controlling display of a display unit (not shown).
e, and an operation signal generation unit 106 that generates an operation signal based on an operation input signal from a mouse or keyboard (not shown).
f.

In such a configuration, as shown in FIG. 2, the CPU 106a sends a program load instruction (command) via a keyboard (not shown) in a state where the disk on which the digital audio signal processing program is recorded is set in the disk drive 104. ) Is input (step S1), the program data is read from the disk and loaded into the internal RAM (step S2). When the loading is completed, a program load flag is set (step S1).
3), end. At this time, since the CPU 106a is compatible with MMX, high-speed signal processing can be performed.

When a play command is input through a keyboard (not shown) in a state where a disk on which a music source such as surround music is recorded is set in the disk drive 104, the CPU 106a sets the first track on the disk. The sub-code indicating the type of the disc is accessed to check whether or not the sub-code is a "music source" (step S4). If YES, the decoding process (and the compression / The limit processing 13) is executed (step S5), and the processed audio data is passed to the audio I / F 106d (step S6). Next, returning to step S5, the decoding process (and the compression / limit process) is repeated. If it is not "music source" in step S4, "unable to play" is displayed on a display unit (not shown) (step S7), and the process ends.

The CPU 106a executes a compression / limit processing 13 as shown in FIG. Here, the operation unit (not shown) uses different selection keys to set the attack time t ₂ ,
The compression start threshold value Vth, the degree of compression of the gain G, and the like can be specified in advance. The signal of the designated value from the operation unit is transmitted to the CPU 106.
a is converted into predetermined parameters by the CPU 106a
The coefficients in the smoothing filter processing 5 and the time constant extension processing 6 shown in FIG. Note that the same reference numerals in the processing shown in FIG. 3 as those in the hardware processing of the conventional example (shown in FIG. 20) are assigned the same reference numerals.

[0028] In FIG. 3, the input digital audio signal is a signal which is delayed by a time t ₁ by the delay processing 1 is selected, then the gain G based on the gain control signal by the gain adjustment processing 2 is adjusted. The gain G is "1" at normal time when the compress / limit function is not set.
When the compress / limit function is set, when the signal level rises sharply, the signal level is set to a value that gradually decreases from “1” so that the signal level attenuates gradually from the rise.

For example, as shown in FIG.
When the rising signal A is input, as shown in FIG.
In addition, due to the delay processing 1, a relatively short time t₁Is
And passed to gain adjustment processing 2 where gain adjustment processing is performed.
It is temporarily stored in the buffer of the management 2. Where the gain adjustment
By the process 2, the attenuation of the gain G of the input signal A is started.
Attack point at time, attack time at decay time
Between, the delay time t 1 is the attack time t_TwoAbout twice
It is preferred.

In the level detection processing 3, the level a of the digital audio signal is detected, and in the subsequent gain generation processing 4, the detection level a is compared with the compression / limit threshold value Vth. (= A
A compressed gain g (Δ) corresponding to −Vth) is generated and passed to the smooth filter processing 5. The compression gain g (Δ) is, for example,

[0031]

G (Δ) = 1−Pi (KiΔ ² + Δ) / a (Equation 1)

This is a function of the difference Δ expressed as follows. Here, a is a level, and Pi and Ki are constants determined by the degree of compression of the gain G. In the smooth filter processing 5, as shown by the curve (rising point Pi → attack point Pt → decay point Pu) in FIG. 4C, the gain is reduced relatively steeply with the time constant according to the set coefficient as compared with the conventional example.

In the present embodiment, the time constant of the gain set by the smooth filter processing 5 is changed by the time constant extension processing 6 into the curve of FIG.
It is extended as shown by the attack point Pt → the attenuation point cc (> Pu)} and passed to the gain adjustment processing 2. Therefore,
If the gain is not extended by the time constant extension processing 6,
When the delay time t ₁ is shortened, the output signal d sharply falls as shown by the broken line in FIG. 4D and the attack time cannot be extended. However, in the present embodiment, the gain G is increased by the time constant extension processing 6. As shown in FIG.
As shown by the solid line in (d), the temperature gradually decreases, and the attack time can be prolonged. Further, since the delay time of the input signal is shortened, echo can be prevented.

Next, the smoothing filter processing 5 and the time constant extension processing 6 will be described in detail with reference to FIG. 5 is equivalent to a circuit in which primary IIR (infinite impulse response) filters are connected in cascade, and includes an addition process 51, delay processes 52 and 53, and a coefficient multiplication process 54.
~ 56 for three stages. In the coefficient multiplication processes 54 to 56,
A coefficient for setting the gain G to less than 1 is set by the CPU 106a shown in FIG.
As a result, the gain G at which the time constant sharply decreases is calculated and output as shown by the curve (rising point Pi → attack point Pt → decay point Pu) in FIG. 4C. The time constant extension processing 6 includes a switch processing SW, a delay processing 61, a coefficient multiplication processing 62, and a comparison processing 66. The coefficient multiplication processing 62 includes a coefficient K for extending the time constant of the gain G.
a (<1) is set by the CPU 106a.

The processing result of the smooth filter processing 5 is delayed through a switch processing switch 61, and a coefficient multiplication processing 62 is performed.
Is fed back to the switch processing SW via the switch processing SW, and the processing result of the smooth filter processing 5 and the processing result of the coefficient multiplication processing 62 are compared by the comparison processing 66 and the larger one is selected by the switch processing SW, whereby the curve Pi (at the rising edge ) → Pt (attack point), the processing result of the smooth filter processing 5 is selected, and in the section after Pt (attack point), the processing result of the coefficient multiplication processing 62, that is, the gain cc with a gradual decrease rate is selected.

FIG. 6 shows a time constant extension process 6a of another example. In FIG. 6, the smoothing filter processing 5 is the same, and the time constant extension processing 6a includes a switch processing SW, a delay processing 61, a coefficient multiplication processing 62, a subtraction processing 63, and an addition processing 64. Similarly, the time constant is extended. The coefficient Ka is set in the coefficient multiplication process 62 by the CPU 106a. Also, the output of the delay processing 61 is subtracted from the delay processing 5
2 is subtracted and passed to a coefficient multiplication process 62,
The coefficient Ka is multiplied. In the subsequent addition processing 64, the processing result of the coefficient multiplication processing 62 and the processing result of the delay processing 52 in the smooth filter processing 5 are added and fed back to the switch processing SW.

In this case as well, the processing result of the smoothing filter processing 5 is compared with the processing result of the addition processing 64 by the comparison processing 66, so that the output signal of the addition processing 64 in the section after Pt (attack point), That is, the gain cc is selected by the switch processing SW. As described above, since the value obtained by subtracting the processing result of the smooth filter processing 5 from the processing result of the time constant extension processing 6a is multiplied by the coefficient Ka, the gain indicated by 0.2 in FIG. The attenuation curve cc is formed so as to converge. In such time constant extension processes 6 and 6a, the attenuation is made gentle so that the attack time t ₂ before extension is increased from 4 ms to 100 ms, for example.

In the first embodiment, the signal level is gradually decreased to increase the attack time. However, if the attenuation is too gentle, the attack, which is called "clear sound" or "crispness", disappears. There is. FIG. 7 and FIG. 8 show a second embodiment in which the attenuation of the signal level is made gentle to extend the attack time and enhance the attack feeling. Here, as shown in FIG. 8A, the delay time due to the delay processing 1 is ΔΔ from the delay time t ₁ of the first embodiment.
short by t. FIG. 7 shows the smooth filter processing 5a and the time constant extension processing 6b of the second embodiment, and the smooth filter processing 5a is equivalent to a 5-stage IIR filter. And
In the time constant extension processing 6b of this embodiment, a multiplication processing 65 for squaring the gain (<1) calculated by the addition processing 64 shown in FIG. 6 is added.

The second embodiment will be described with reference to FIG. 8. For example, when a signal A which rises sharply as shown in FIG. 8A is input, the delay processing is performed as shown in FIG. 1, a relatively short time (t ₁ −Δt) compared to the conventional example
And is passed to the gain adjustment processing 2. Further, the level a of the digital audio signal is detected by the level detection processing 3 and then compared with the compression / limit threshold value Vth by the gain generation processing 4, and when the threshold value Vth is exceeded, the difference Δ (= a−Vth) is obtained. Is generated and passed to the smooth filter processing 5a. The compress gain g (Δ) is, for example,
Are used.

Then, in the second embodiment, the time constant of the gain set by the smooth filter processing 5a is changed by the time constant extension processing 6b to the curve {rising edge Pi → attack point Pt → cc ′ (c) in FIG. Pu <decay point c
c ′ <cc)}. In this case, comparing the processing result of the smoothing filter processing 5a and the processing result of the multiplication processing 65 by the comparison processing 66 and selecting the larger one by the switch processing SW, as shown in FIG. In the section between Pt and the attack point Pt ′ in the case of the first embodiment, a steep gain by the smooth filter processing 5a is selected.
As shown in FIG. 8D, the output signal sharply falls in the Pt-Pt 'section, so that the attack feeling can be emphasized. Further, since the gain of the gentle curve cc 'is selected after the attack point Pt' in the case of the first embodiment, the attack time can be extended by making the signal level decay gently.

Next, a third embodiment will be described with reference to FIGS. In the third embodiment, the level detection processing 3 is performed in place of the compression / limit processing 13 shown in FIG.
a and the maximum value selection processing 3b are added, and the level detection processing 3
In a, the level of the signal a ′ delayed by the delay processing 1 is detected. Then, the level of the signal a 'and the level of the signal a detected by the level detection processing 3 are passed to the maximum value selection processing 3b and compared, and the higher level is passed to the gain generation processing 4. The gain adjustment processing 2, the gain generation processing 4, the smooth filter processing 5, and the time constant extension processing 6 are the same as the processing shown in FIG.

[0042] According to such processing, even when the signal level becomes zero at time t _a later attack started as shown in FIG. 10, the gain G is between time t _a of the delay time t ₁ is The signal level processed by the gain adjustment processing 2 can maintain the curve shown in FIG. 4 because the signal level processed by the gain adjustment processing 2 does not become zero but continues to be 0.2, thus achieving more accurate compression / limit characteristics. Can be.

Next, a description will be given of a fourth embodiment of the present invention capable of performing the compression / limit characteristic processing while balancing the signal levels of a plurality of systems. FIG. 11 shows the fourth
It is a functional block diagram for explaining compression / limit processing 13A of an example. The fourth embodiment differs from the first embodiment of FIG. 1 in the following points. That is, while the first embodiment is configured for one system (one channel) of signals,
The fourth embodiment of FIG. 11 is configured for three systems (three channels) of signals L, R, and S.

[0044] Specifically, L, R, each signal of each channel of the S, delay processing 1L, 1R, 1S by, is a relatively short amount of time t ₁ delayed compared with the prior art the gain adjustment processing 2L, 2R, 2S And the gain adjustment processing 2L, 2L
It is temporarily stored in the buffer for R and 2S. In the level detection processes 3L, 3R, and 3S, the levels of the digital audio signals of the L, R, and S channels are detected, respectively, and passed to the subsequent common level detection process 7. In the common level detection processing 7, a signal of a large level (maximum level signal) is selected from the input signals of the L, R, and S channels, and then, as in the first embodiment, a gain generation processing 4, a smooth filter processing 5. It is passed to the time constant extension processing 6.
Then, from the time constant extension processing 6 to the gain adjustment processing 2L, 2
A gain G for compression / limit processing is passed to R and 2S.

FIG. 12 is a functional block diagram for explaining specific processing of the common level detection processing 7 and the gain generation processing 4 of FIG. The common level detection processing 7 has a comparator (COMP) processing 25 and a switch processing 26. The comparator processing 25 compares the input signals of the L, R, and S channels. The input signal of the largest channel is selected based on the input signal. It should be noted that although there are three systems in this example, the signal of the maximum level is similarly selected when applied to a configuration in which the input signals are two systems (for example, L and R).

As shown in FIG. 12, the gain generation processing 4 includes a comparator processing (COMP) 29 and a threshold register (R
EF) 30, addition (subtraction) processing 31, 36, 39, multiplication processing 34, 35, 38, 41, 42, division processing 37, predetermined value registers 33, 40, and switch processing 32.
The threshold value Vth is previously set in the threshold value register 30 by the CPU 106a, the data corresponding to the logic "0" is set in the predetermined value register 33, and the data corresponding to the logic "1" is set in the predetermined value register 40. Is done.

In the subtraction process 31, a difference Δ between the signal of the channel selected by the common level detection process 7 and the threshold value Vth from the threshold value register 30 is calculated. Further, the comparator processing 29 compares the level a of the maximum level signal selected by the common level detection processing 7 with the threshold value Vth.
If> Vth, the difference Δ is calculated by the switch processing 32 and given to the addition processing 36 and the multiplication processing 34 and 42.
Ki, Pi, and Qi are multiplication processes 35, 41, and 3, respectively.
8 is the coefficient set. Therefore, when the level of the maximum level signal selected by the common level detection processing 7 is not larger than the threshold value Vth, the processing result of the addition processing 36 becomes 0, and the processing result g (Δ) of the gain generation processing 4
Becomes 1.

In the example shown in FIG. 12, the signal of Lch is R
In this case, the case where the coefficients Qi, Ki, and Pi determine the curve characteristics with respect to the threshold value Vth for the Lch signal is shown. That is, the processing result g (Δ) of the gain generation processing 4 is

[0049]

Ka = 1−Pi (QiΔ ³ + KiΔ ² + Δ) / Lin (Equation 2)

## EQU1 ## This value becomes the compression / limit gain. Here, Pi, Qi and Ki are constants determined by the degree of compression of the gain G, and Lin is the maximum level.

Also, compared to equation (1), equation (2) has a coefficient Q
Since it has the third order term of i, the degree of freedom in designing the curve characteristics is improved. In the gain generation processing 4, the division by the input level (Lin in the above example) is performed to normalize to Lin = 1. Processing result g of gain generation processing 4
(Δ) is processed by the smooth filter processing 5 and the time constant extension processing 6 in the same manner as in the first embodiment.
L, 2R, and 2S are given as gains G, respectively.
Therefore, a curve characteristic is formed by the action of the time constant during the attack. That is, when looking at the Lch, using the time constant dIN in the delay processing 1L and the time constant τf of the smooth filter processing 5, the attack time τa is:

[0052]

Τa = τf-dIN (Equation 3)

Is given as The same applies to Rch and Sch, and an output signal in which the compression / limit gain is adjusted in a well-balanced manner by the gain adjustment processing 2L, 2R, and 2S is obtained.

In the above embodiments, when the attack time is switched to the longer one, the coefficient Ka of the time constant extension processing 6 is continuously changed from 0 to 1 to make the continuous time change. When the attack time is switched to the shorter one, the continuous time change is given by continuously changing only the time constant of the smooth filter processing 5 with the coefficient Ka set to 0. Can be. Therefore, the operation can be satisfactorily performed without interruption from a short attack time to a sufficiently long attack time.

The gain generation processing 4 in each of the above-described embodiments is based on the difference Δ
Although an example of generating (Δ) has been described, the present invention is not limited to this, and any method may be used as long as it generates a compression / limit gain g according to the level. In the common level detection processing 7 of the fourth embodiment, the maximum level signal among a plurality of input signals is selected, but instead of selecting the maximum level signal, the average of the levels of the plurality of input signals is calculated. There may be.

FIG. 13 shows a modification 13B of the compress / limit processing in the fourth embodiment, which corresponds to the third embodiment (see FIG. 9). That is, in FIG.
Level detection processes 3La, 3Ra, and 3Sa are added to the signals of the L, R, and S channels, and the level detection processes 3La, 3Ra, and 3Sa respectively include delay processes 1L,
The level of the signal a 'of each channel of L, R, S delayed by R, 1S is detected. And this L, R, S
Of the signal a 'of each channel and the level detection processing 3
The levels of the signal a detected by L, 3R, and 3S are passed to a common level detection process 7 for comparison, and the higher level is passed to a gain generation process 4. The gain adjustment processing 2L, 2R, 2S, the gain generation processing 4, the smooth filter processing 5, and the time constant extension processing 6 are the same as the processing shown in FIG.

Next, a description will be given of a fifth embodiment of the present invention capable of performing the compression / limit characteristic processing while balancing the signal levels of a plurality of systems for surround audio. First, the surround audio encoders L, C, S, and R as described with reference to FIG.
The multi-channel signal composed of the four signals is encoded and converted into a two-channel signal of Lt and Rt.
This signal is recorded on the disk shown in FIG. 1 together with a surround digital audio signal processing program as shown in FIG. 14, and the recorded data is read by the personal computer 106 and subjected to digital audio signal processing including compression / limit processing. Is done.

In the surround digital audio signal processing program shown in FIG. 14, the input two-channel signals Lt and Rt are decoded into four channels by delay processing 1L and 1R, addition processing 72 and subtraction processing 73, and then these four channels are processed. (L = Lt, C = Lt + R)
(t, S = Lt-Rt, R = Rt) are monitored by the control processing 78, and if any one signal level becomes extremely large, the signal levels of the other three channels are attenuated / reduced. So that the gains G _L , G _R , and G corresponding to the gain adjustment processing 74 to 77 of each channel are performed.
_C and G _S are controlled. The gains G _L , G _R , G _C , and G _S are values given from the control processing 78 in order to provide a surround sound listener with a sound field having a clearer sense of direction. Values that attenuate the gains of the gain adjustment processes 74 to 77 by 0 to 30 dB are passed to the multiplication processes 79 to 82, respectively.

The compress / limit characteristic processing 83 shown in FIG. 14 passes a common coefficient G obtained from the average or maximum value of the signal levels of the two input channels Lt and Rt to the multiplication processing 79 to 82, where In processing 79 to 82, the common coefficient G is set to the control processing 78
Are multiplied by the gain control values G _L , G _R , G _C , and G _S set as described above, and the multiplication results G _L ^* , G _R ^* , G _C ^* , and G _S ^* are passed to gain adjustment processes 74 to 77, respectively.

In FIG. 14, the delay processing 1L, 1R is 2
The compression / limit characteristic processing 83 for delaying the input signals Lt and Rt of the system and obtaining a common coefficient G from the average or the maximum value of the signal levels of the two channels Lt and Rt is shown in FIG. This is the same as the two channels shown. With such a program, the compression / limit characteristic processing of four channels can be realized using the processing of two systems.

Next, a case will be described in which, for example, a surround signal processing program having the above-described compress / limit characteristic processing and a surround music source are recorded on a recording medium. First, as a first method, a disk is used in which a plurality of tracks are formed on a recording surface, the track direction is divided into a plurality of sectors, and a unique address is assigned to each track and sector. Then, a first data area in which a surround signal processing program is recorded on a plurality of tracks and sectors, a second data area on which a surround music source is recorded on a plurality of tracks and sectors, TOC (Table Of Contents) in which data indicating that a surround signal processing program and a surround music source are recorded in the second data area, respectively, are recorded.
By dividing into areas, computer software compatible with Dolby Surround can be realized ("Dolby" is a registered trademark of Dolby Corporation in the United States).

As a second method, for example, a CD extra (system identifier is “CD EXTRA”) or an enhanced music CD having a diameter of 120 mm
Optical disk D is used. FIG. 15 shows the recording area, in which the first lead-in area 7 extends from the inner circumference to the outer circumference.
a, first data area 7b, first lead-out area 7
c, second lead-in area 7d, second data area 7
e, the second lead-out area 7f.

The first lead-in area 7a is the first TO where the address of the data in the first data area 7b is recorded.
The second lead-in area 7d constitutes a second TOC area in which an address of data of the second data area 7e is recorded. Then, a surround signal processing program and a surround music source are recorded in the first data area 7b and the second data area 7e, respectively. As another method, the information may be recorded on a recording medium such as a DVD-ROM, an enhanced DVD, an MD-ROM, an enhanced MD, a flexible disk, and an MO disk.

FIG. 16 shows the processing when such a disk is used. First, if the program has not been loaded in step S11, the process proceeds to step S12, where the first data (program) is read from the disk. Next, it is checked whether or not the program is recorded on the disk (step S13), and if there is no program, the process is terminated. On the other hand, if the program is recorded on the disc,
The program is loaded into the program RAM (see FIG. 1) in the U106a (step S14), and this is performed until the program ends (step S15).

When the loading of the program is completed, the test data is loaded into the buffer RAM in the CPU 106a (step S15 → S16), and the test data is decoded (step S17). Then, it is checked whether or not the decoding result is OK (step S18). If OK, the program loaded flag is set (step S19), and the process returns to step S11. On the other hand, step S1
If the decoding result is not OK in step 8, "unable to play" is displayed (step S20), and this process ends.

If the program has been loaded in step S11, the second data (music source) is read from the disk and loaded into the buffer RAM in the CPU 106a (step S21). Then the music source ends the audio frame (E
OF) (step S22), and in the case of NO, the decoding process including the compression / limit characteristic process is executed as described above (step S23), and the decoding result is output to the audio I / F 106d. (Step S24), returning to step S21. Then, this process is repeated until EOF (step S22).

In this embodiment, a plurality of programs which are different for each processing characteristic of the compress / limit, and a program which can be arbitrarily selected by the user through a GUI (graphical user interface) Is prepared. That is, when the user instructs the GUI during the decoding shown in step S23 shown in FIG. 16, the interrupt routine shown in FIG. 17 starts. In this routine, first, as shown in FIG. 18, illustration images (in this example, nine types) indicating individual compression / limit processing characteristics are displayed (step S31). And
When the user selects and designates one of the programs, the program accesses the designated program (step S32 → S33), and then mutes the audio data (step S34).
Next, the instruction program is reloaded by overwriting the previous program (step S35), and then the mute is released and compression / compression is performed based on the specified program.
A limit process is performed (step S36), and the process returns.

FIG. 19A shows a modification of FIG. 17, in which step S31a is added between steps S31 and S32. In this modification, first, when an illustration image showing individual compression / limit processing characteristics is displayed as shown in FIG. 18 (step S31), as shown in FIG. Input / output characteristics)
And the characteristics (input / output characteristics) of the currently selected compress / limit processing program are displayed together (step S31a). Then, when the user selects and instructs another program (step S32), the same processing as that shown in FIG. 17 is executed. Also, by displaying the characteristics in an illustration, selection and designation become easy, and by displaying the characteristics of the input signals together, level management can be performed accurately while comparing.

Next, a description will be given of a seventh embodiment in which the above-mentioned program and digital audio signals processed by this program are transmitted via a communication line. FIG.
4 is a block diagram showing the network terminal 105 in the personal computer 106 shown in FIG. 1 in detail, and FIGS. 25 and 26.
Is a flowchart showing processing of the data conversion unit in FIG. 24;
FIG. 27 is an explanatory diagram showing a communication network, and FIG.
FIG. 7 is an explanatory diagram showing packet processing on the network No. 7; The terminal 105 is provided on both the data transmitting side and the data receiving side personal computer 106.
Buffer T1, transmission buffer T2, data converter T4, adapter T3 as a communication interface, communication terminal T6 and controller T
5

The data conversion unit T4 on the data transmission side is shown in FIG.
As shown in (4), the transmission data stored in the transmission buffer T2 is divided into packets of a predetermined length and packetized (step S4).
1) Then, a header including the destination address is added to the head of the packet (step S42), and then this is output onto the network NW via the adapter T3 and the communication terminal T6 (step S43). As shown in FIG. 26, the data conversion unit T4 on the data receiving side removes the header from the packet received from the network NW via the communication terminal T6 and the adapter T3 (step S51), and then restores the received data (step S52). ) And then transfer it to the program RAM 106 shown in FIG.
b (step S53).

Communication terminals T on the data transmission side and the data reception side
6 are connected via a network NW as shown in FIG. 27, for example. In this network NW, for example, CAT
TCP / IP (Tr
Data is transmitted in units of packets using the protocol of an Ansmission Control Protocol / Internet Protocol. In this case, as shown in FIGS. 27 and 28, the packets R output from the data transmission side are separated for each packet by selecting an optimum route by the router R on the network NW, and then separated by the router R. Each packet is sent to a packet switch Pn (n = 1 to
The packets are sequentially transmitted to the personal computer 106 on the data receiving side via k).

Therefore, the personal computer 10 on the data receiving side
At 6, the music source in the disc set in the disc drive device 104 is transferred to the program RA.
Compress and limit can be performed based on the program on M106b.

Next, an eighth embodiment in which the music source is compressed and limited on the data transmitting side and transmitted to the data receiving side will be described. FIG. 29 shows an encoder for compressing and limiting a music source and recording it on, for example, a DVD audio disk. A / D
The converter 111 has a stereo 2ch or Dolby PROLO
A GIC surround audio signal is input, and each channel is converted into a PCM digital signal.
The digital signal is compressed / limited by the signal processing circuit 112 and the memory 113.

FIG. 30 shows the signal processing circuit 112 and the memory 1
13 is a flowchart showing one channel of the compression / limit processing 13 shown in FIGS. 3 and 9. First, when a digital signal is input from the A / D converter 111 (step S61), the sampling data of each channel is delayed by one sample (step S6).
2), the next sampling data is passed to the delay processing (step S63). Further, the level of the digital signal from the A / D converter 111 is detected (step S64), and then the gain is generated (step S65). Next, in the case of the compress / limit processing 13 shown in FIG. 3, the process proceeds to step S67, while in the case of the process shown in FIG. 9, the maximum value is selected (step S66), and the process proceeds to step S67.
In step S67, smooth filter processing is performed, then time constant extension processing is performed (step S68), output from the delay unit (step S69), and gain adjustment (step S7).
0), and perform output processing (step S71).

FIG. 31 shows the signal processing circuit 112 and the memory 1
13, one channel of the surround signal of each of the L, R, and S channels shown in FIG.
This is shown in a flowchart of the limit process. First,
When a digital signal is input from the A / D converter 111 (step S61), the sampling data of each channel is delayed by one sample (step S62), and the next sampling data is passed to the delay processing (step S6).
3). Further, the level of the digital signal from the A / D converter 111 is detected (step S64), the common level is detected (step S64a), and the gain is generated (step S65). Next, a smooth filter process (step S67), a time constant extension process (step S68), and output from the delay unit (step S69),
The gain is adjusted (step S70), and output processing is performed (step S71).

The audio data subjected to the compression / limit processing as described above is such that after the copyright data is embedded from the copyright data supply unit 140, the DVD encoding circuit 1
14 is encoded into a DVD format as shown in FIG. FIG. 32 shows a DVD format audio pack. One pack has a 4-byte pack start.
14-byte packed header consisting of 6-byte SCR, 3-byte Mux rate and 1-byte stuffing, and 2034-byte user data (audio data)
(Total 2048 bytes). The data encoded in this manner is output to the outside via the output terminal OUT1, or is recorded on a DVD via the output terminal OUT2 after being subjected to EFM modulation by the modulation circuit 115. Also, the embedded copyright data is output from the output terminal OUT.
3 to the outside.

FIG. 33 shows a decoder for processing a stream signal reproduced from a DVD. The stream signal reproduced from the DVD is EFM-demodulated by the demodulation circuit 211 and then de-formatted by the DVD decoding circuit 212 to separate PCM audio data. Next, the audio data is processed by the signal processing circuit 213.
And L + R two channels or L + R by the memory 214
+ S + C are separated into multi-channel signals, and the signals of each channel are converted into analog signals by D / A converters 215a and 215b.
3 and 224 to the speakers SP1 and SP2.

When the PCM data deformatted by the DVD decryption circuit 212 is input via the input terminal 225 to permit the PCM data to be output as it is, the decryption unit 226 switches the switch 221.
Is turned on, the output signal of the DVD decoding circuit 112 is output to the PCM data output terminal 2 via the switch 221.
22. Then, the PCM data is transmitted to the data receiving side via the network NW as shown in FIG. 27, and the audio I / F 106d shown in FIG.
And reproduced by the speaker 103.

The present invention is not limited to the case in which the data transmitting side transmits unidirectionally to the receiving side. For example, the user of the data receiving side transmits the data based on the homepage of the data transmitting side.
The present invention can also be applied to a case where a program transmission request or an audio signal transmission request is transmitted, and the data transmission side transmits a program or an audio signal based on the request. In the case of the Internet, a TCP / IP protocol group is used as the interface T3.

Further, according to the present invention, a digital audio disk (for example, DVD) in which only a digital audio signal processed by the program is recorded without recording a program for compressing and limiting the level of the digital audio signal is realized. can do. According to such a disc, by recording a digital audio signal whose attack time is optimally adjusted,
During playback, audio signals with a wide dynamic range can be compressed and limited for playback.
Weak and strong sounds can be reproduced very easily.

According to the present invention, a program for compressing and limiting the level of a digital audio signal and a digital audio signal recorded in, for example, a first data area 7b and a second data area 7e shown in FIG. A disk can be realized. In such a disc, for example, the 3 × shown in FIG.
In the multi-window screen of No. 3, nine types of programs from the program with the least compress / limit effect shown in the upper right to the program with the greatest compress / limit effect shown in the lower left are recorded, and the digital audio signal is previously recorded in the upper right. The program may be processed and recorded by a program having the least compress / limit effect as shown.

According to such a disk, a user can select and reproduce a program according to his / her preference. For example, a disc maker may record the dynamic range of an audio signal in advance in accordance with the disc, and select a playback state to be compressed so that the user can easily listen in a non-ideal environment such as a noisy vehicle. it can.

[0083]

As described above, according to the present invention, the time constant of the gain is extended by the digital audio signal processing program, so that the delay time of the digital audio signal is reduced to prevent echo. And the attack time can be extended. Further, according to the present invention, the time constant of the gain is extended by the program for digital audio signal processing to attenuate the digital input signals of a plurality of systems. In performing the processing, it is possible to prevent the echo by shortening the delay time of the input signal and to lengthen the attack time while balancing the signal levels of a plurality of systems. According to the present invention, furthermore, a digital input signal of a plurality of systems is generated from a digital input signal of a plurality of systems, and a time constant of a gain is extended by a program for processing a digital audio signal, thereby obtaining a digital input signal of the multi-system. When performing compression / limit characteristic processing on signals of a plurality of systems for multi-channel reproduction, a common level is detected from signals of a small number of systems before decoding, so that a multi-channel signal is attenuated. The echo signal can be prevented by shortening the delay time of the input signal while the signal level is balanced, and the attack time can be lengthened. Also, by displaying the characteristics in an illustration, selection and designation become easy, and by displaying the characteristics of the input signals together, level management can be performed accurately while comparing. Further, according to the present invention, since the above-mentioned program is transmitted via a communication line or a network, it is possible to shorten the delay time of the digital audio signal, prevent echo, and increase the attack time. Can be transmitted via a communication line. Further, according to the present invention, the digital audio signal processed by the above program is transmitted via a communication line or a network, so that the delay time of the digital audio signal can be shortened to prevent echo. In addition, a digital audio signal that can lengthen the attack time can be transmitted through a communication line. Further, according to the present invention, the digital audio signal processed by the above program is recorded on the digital audio recording medium, so that the delay time of the digital audio signal can be shortened to prevent echo and to prevent an attack. A digital audio recording medium whose time can be extended can be realized. Further, according to the present invention, both the digital audio signal processed by the above program and the above program are recorded on the digital audio recording medium, so that the delay time of the digital audio signal can be reduced according to the user's preference. It is possible to realize a digital audio recording medium that can be shortened to prevent echo and extend the attack time.

[Brief description of the drawings]

FIG. 1 is a block diagram showing a digital audio signal processing system to which an embodiment of a digital audio signal processing recording medium according to the present invention is applied.

FIG. 2 is a flowchart illustrating a digital audio signal processing program of the personal computer of FIG. 1;

FIG. 3 is a functional block diagram for explaining in detail a compress / limit process in the decoding process of FIG. 2;

FIG. 4 is a waveform chart showing an attenuation characteristic by the compression / limit processing of FIG. 3;

5 is a functional block diagram for explaining an example of a smooth filter process and a time constant extension process of FIG. 3 in detail.

FIG. 6 is a functional block diagram for explaining another example of the smooth filter processing and the time constant extension processing of FIG. 3 in detail.

FIG. 7 is a functional block diagram for explaining in detail a smooth filter process and a time constant extension process of the second embodiment.

8 is a waveform chart showing an attenuation characteristic by the compress / limit processing of FIG. 7;

FIG. 9 is a functional block diagram for explaining in detail a compress / limit process according to a third embodiment;

FIG. 10 is a waveform chart showing an attenuation characteristic by the compress / limit processing of FIG. 9;

FIG. 11 is a functional block diagram for explaining in detail a compress / limit process of a fourth embodiment.

12 is a functional block diagram for explaining in detail a common level detection process and a gain generation process of FIG.

FIG. 13 is a functional block diagram for explaining in detail a modification of the compression / limit processing of the fourth embodiment.

FIG. 14 is a functional block diagram for explaining in detail a process of a surround decoder according to a fifth embodiment;

FIG. 15 is an explanatory diagram illustrating an example of a recording medium.

16 is a flowchart for explaining a digital audio signal processing program recorded on the recording medium of FIG.

FIG. 17 is a flowchart illustrating a program for displaying an illustration of a compress / limit characteristic according to a sixth embodiment.

FIG. 18 is an explanatory diagram showing a display example by the illustration display program of FIG. 17;

FIG. 19 is a flowchart for explaining a modification of the illustration display program for the compress / limit characteristics of the sixth embodiment.

FIG. 20 is a block diagram showing a conventional compress / limit circuit.

FIG. 21 is a waveform chart showing main signals of the compress / limit circuit of FIG. 20;

FIG. 22 is a block diagram illustrating a surround audio encoder.

FIG. 23 is a block diagram showing a conventional surround audio decoder.

FIG. 24 is a block diagram showing in detail a network terminal in the personal computer shown in FIG. 1 in the seventh embodiment.

FIG. 25 is a flowchart showing the processing of the data conversion unit in FIG. 24;

FIG. 26 is a flowchart showing the processing of the data conversion unit in FIG. 24;

FIG. 27 is an explanatory diagram showing a communication network.

FIG. 28 is an explanatory diagram showing packet processing on the network in FIG. 27; It is.

FIG. 29 is a block diagram illustrating an encoder according to an eighth embodiment.

30 is a flowchart illustrating an example of processing of the signal processing circuit of FIG. 29.

FIG. 31 is a flowchart illustrating another example of the processing of the signal processing circuit in FIG. 29;

FIG. 32 is an explanatory diagram showing a DVD format. It is.

FIG. 33 is a block diagram showing a decoder according to the eighth embodiment.

[Explanation of symbols]

1, 1L, 1R, 1S Delay processing 2, 2L, 2R, 2S, 2C, 74 to 77 Gain adjustment processing (attenuation step) 3, 3a, 3L, 3R, 3S, 3La, 3Ra, 3Sa
Level detection processing (level detection step) 3b Maximum value selection processing 4 Gain generation processing (composing a gain generation step together with smooth filter processing 5) 5, 5a Smooth filter processing 6, 6a, 6b Time constant extension processing (time constant extension step) 7) Common level detection processing (detection step) 72 Addition processing (delay processing 1L, 1R and subtraction processing 73)
73 constitutes a decoding step) 73 Subtraction processing 106a CPU NW network

Claims (15)

[Claims] A level detecting step of detecting a level of a digital input signal; a gain generating step of generating a time-variable compress / limit gain based on the level detected by the level detecting step; A time constant extending step of extending a time constant of the gain generated by the gain generating step; and an attenuation step of attenuating the digital input signal based on the gain whose time constant has been extended by the time constant extending step. A recording medium for digital audio signal processing on which a program is recorded. 2. A level detecting step for detecting a level of a digital input signal and a level of a signal obtained by delaying the digital input signal, and a time varying compressor / compressor based on the level detected by the level detecting step. A gain generating step of generating a gain for the limit, a time constant extending step of extending a time constant of the gain generated in the gain generating step, and a time constant extended based on the gain in which the time constant is extended by the time constant extending step. An attenuation step for attenuating a signal obtained by delaying a digital input signal. 3. A level detecting step for detecting each level of a plurality of digital input signals, and a detecting step for detecting an average value or a maximum value of each level of the plurality of digital input signals detected by the level detecting step. A gain generation step of generating a temporally variable compression / limit gain based on an average value or a maximum value of each level detected by the detection step; A time constant extending step of extending a time constant; and an attenuating step of attenuating each of the digital input signals of the plurality of systems based on the gain whose time constant has been extended by the time constant extending step. Recording media for digital audio signal processing. 4. A level detection step for detecting each level of a plurality of digital input signals and a level of a signal obtained by delaying the plurality of digital input signals, and averaging each level detected by the level detection step. Compress / variable in time based on value or maximum value
A gain generating step for generating a gain for the limit, a time constant extending step for extending a time constant of the gain generated in the gain generating step, and a gain in which a time constant is extended in the time constant extending step. An attenuating step of attenuating a signal obtained by delaying each of the digital input signals of the plurality of systems. 5. A level detection step for detecting each level of a plurality of digital input signals, and a detection step for detecting an average or a maximum value of each level of the plurality of digital input signals detected by the level detection step. A gain generation step of generating a temporally variable compression / limit gain based on an average value or a maximum value of each level detected by the detection step; A time constant extending step of extending a time constant, a decoding step of generating a multi-system digital signal based on a plurality of digital input signals, and a time constant extended by the time constant extended by the time constant extending step. Attenuating step for attenuating each of the digital signals of the multiple systems. Recording media for digital audio signal processing. 6. A level detecting step for detecting each level of a plurality of digital input signals, and a detecting step for detecting an average value or a maximum value of each level of the plurality of digital input signals detected by the level detecting step. A gain generation step of generating a temporally variable compression / limit gain based on an average value or a maximum value of each level detected by the detection step; and further based on a plurality of digital input signals. Digital audio signal processing recorded with a program, comprising: a decoding step of generating a multi-system digital signal; and an attenuating step of attenuating each of the multi-system digital signals based on the gain generated in the gain generating step. Recording media. 7. The gain generation step includes a plurality of gain generation steps for generating a plurality of different compression / limit gains, and the program further selects one of the plurality of gain generation steps. 7. The recording medium for digital audio signal processing according to claim 1, further comprising a step. 8. The recording for digital audio signal processing according to claim 7, wherein said program further has a program for displaying a plurality of different compression / limit characteristics in said plurality of gain generation steps as an illustration. Medium. 9. The digital audio device according to claim 1, wherein the program further has a program for displaying the characteristics of the digital input signal and the characteristics for compression / limit by illustration. Recording medium for signal processing. 10. The program data according to claim 1, wherein a header for transmitting the program data via a communication line or a network is added, and the program data to which the header is added. Converting the digital signal sequence into a digital signal sequence for a communication line or a network, and transmitting the digital signal sequence to a receiving side via a communication line or a network. 11. The program data according to claim 1, further comprising a header for transmitting the program data via a communication line or a network, and transmitting the program data to which the header is added. A digital audio signal communication method for receiving a digital signal sequence when converting the digital signal sequence into a digital signal sequence for a line or a network and transmitting the digital signal sequence via a communication line or a network, based on the header Receiving a digital signal sequence through the digital signal sequence; and separating program data from the digital signal sequence. 12. A step of processing an audio signal based on the program according to claim 1; and processing the audio signal processed based on the program via a communication line or a network. Adding a header for transmitting the data, converting the audio data to which the header has been added into a digital signal sequence for a communication line or a network, and receiving the digital signal sequence via a communication line or a network. Transmitting the digital audio signal to a digital audio signal. 13. An audio signal is processed based on the program according to claim 1, and the audio signal processed based on the program is transmitted via a communication line or a network. A digital signal sequence for a communication line or a network, and the digital signal sequence when the digital signal sequence is transmitted via a communication line or a network. A digital audio signal communication method for a digital audio signal, comprising: receiving a digital signal sequence based on the header; and separating and reproducing an analog audio signal from the digital signal sequence. Communication method for 14. A digital audio recording medium on which an audio signal is processed and recorded based on the program according to claim 1. Description: 15. An audio signal processed and recorded based on the program according to any one of claims 1 to 6, and a digital signal recorded with the program according to any one of claims 1 to 9. Audio recording medium.