JPH04322600A - Surround audio equipment - Google Patents

Abstract

PURPOSE:To use a DSP so as to implement properly surround processing, creation of a sound field and its correction picture. CONSTITUTION:The equipment is provided with 1st-3rd DSPs and a A/D converter. The 1st DSP implements generation of a test tone signal, adjustment of balancing levels of plural digital audio signals, and correction of a dynamic range. The 2nd DSP applies decode processing for surround stereo reproduction to an output signal of the 1st DSP to output a digital surround signal. The 3rd DSP applies sound field creation or correction processing to an output signal of the 2nd DSP. The D/A converter converts the output signal of the 3rd DSP into an analog audio signal, which is outputted.

Description

[Detailed description of the invention]

0001

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to surround audio equipment.

0002

2. Description of the Related Art As a surround audio device, there is a Dolby Pro Logic surround decoder.

[0003] This surround decoder is hardware that reproduces the sound field intended by the software creator for movies and other audio, and it converts audio signals transmitted in two channels while improving crosstalk characteristics. It will be made into 4 channels. Furthermore, as an additional function, it is possible to create or correct a sound field, and for example, it is possible to create a sound field as if you were listening to a performance in a large hall.

FIG. 5 shows an example of such a surround decoder, in which 1A to 1N are plural analog input terminals, 2A to 2M are plural digital input terminals, and 13 to 15 are plural analog input terminals.
are input selector switches that interlock with each other, and 16 is a D/A
It is a converter.

[0005]The terminals 1A to 1N are supplied with analog stereo audio signals L and R that have been encoded for surround playback, or general analog stereo audio signals L and R, respectively. L and R are selected by switches 13 and 14.

Similar digital audio signals are also supplied to the terminals 2A to 2M, respectively, and the switch 15
Selected by Note that the converter 16 is for converting the digital audio signals of the terminals 2A to 2M into analog signals, since all processing such as surround processing is performed by analog processing.

Signals L and R from the switch 14 are supplied to a Dolby Pro Logic decoder 23 through a balance adjustment circuit 21 and a switch 22.

Based on the levels and phases of the signals L and R supplied thereto, the decoder 23 converts the front left and right channel audio signals FL and FR from the signals L and R.
and a front center channel audio signal FC, and outputs rear left and right channel audio signals BL and BR in cooperation with a decoder 31, which will be described later.

In this case, the signals FL and FR are the same as the signals L and R.
This is obtained by enhancing the level of the in-phase component in accordance with the level of its in-phase component. Also, the signal FC is the signal L
, R are mainly in-phase components. Furthermore, the signals BL, B
R is obtained mainly from the difference component (L-R) of the signals L and R.

The front channel signals FL and FR from the decoder 23 are taken out to an output terminal 25 through a switch 24.

Furthermore, the center channel audio signal FC from the decoder 23 is taken out to an output terminal 26.

Furthermore, a difference component (L-R) between the signals L and R is taken out from the decoder 23, and this difference component (L-R)
are sequentially supplied to the decoder 31 and the delay circuit 32. In this case, the decoder 31 has level expansion characteristics and high frequency cut characteristics, and the delay circuit 32 is for reproducing reverberation components in a hall or the like.

In this way, the rear left and right channel signals BL and BR are taken out from the delay circuit 32, and this signal B
L and BR are output to the terminal 34 through the switch 33.

Reference numeral 29 denotes a test tone signal forming circuit, and when the switch 22 is connected to the forming circuit 29 side, from this forming circuit 29, the front left,
Test tone signals localized to the front right, front center (center), and rear are output.

Furthermore, when the switch 24 is connected to the switch 14 side, the audio signals L and R selected by the switch 14 are output directly to the terminal 25. (Dolby, Dolby Pro Logic, etc. are registered trademarks)

[
0016

SUMMARY OF THE INVENTION A Dolby Pro Logic surround decoder is constructed as described above.

However, in the above-mentioned surround decoder, the balance adjustment circuit 21, Dolby Pro Logic decoder 23, test tone signal forming circuit 29, decoder 31, and delay circuit 32 are constructed from analog discrete components or analog ICs. Therefore, the number of parts increases, the size of the device increases, and the cost increases.

[0018] Furthermore, since it is composed of analog discrete parts and analog ICs, adjustments are required during assembly. Moreover, even with such adjustment, changes occur over time.

Furthermore, since the switches 13 and 14 are required to switch the analog audio signals, the sound quality may deteriorate due to the contacts of the switches 13 and 14.

The present invention attempts to eliminate these problems.

[0021]

[Means for Solving the Problems] Therefore, in the present invention, if the reference numerals of each part correspond to the embodiments described later, a plurality of digital audio signals that have been encoded for surround playback are supplied. Input terminal 2A~2
M, a first DSP 51 that forms a test tone signal, adjusts the level balance between the plurality of digital audio signals, and corrects the dynamic range, and selectively supplies the plurality of digital audio signals to the first DSP 51. a switch circuit 43; a second DSP 52-56 that performs decoding processing for surround stereo reproduction on the output signal of the first DSP 51;
A third DSP 61 to 65 performs sound field creation or correction processing on the output signal of No. 6, and this third DSP 61
D/A converters 71 to 73 that convert the output signals of D/A converters 71 to 65 into analog audio signals, and output terminals 81 to 83 that output analog audio signals from the D/A converters 71 to 73 are provided.

[0022]

[Operation] The first DSP selectively forms a test tone signal, adjusts the level balance between a plurality of digital audio signals, and corrects the dynamic range.
Surround processing is performed by the second DSP, and further,
The third DSP creates or corrects the sound field, and overall achieves optimal audio playback suited to the listener's requirements and the audio environment.

[0023]

[Embodiment] In FIG. 1, 91 indicates a microcomputer for system control.
Various operation keys 92 are connected to the , and control signals, programs, or data corresponding to the key outputs are output. This control signal, program, or data is then supplied to switch circuits 41, 43, DSPs (digital signal processors) 51-65, and muting circuits 74-76, which will be described later.

Reference numeral 94 indicates a transmitter of the remote control, and this transmitter 94 outputs a remote control signal corresponding to the key operation, for example, infrared light if the remote control is an infrared type. This infrared light is then received by the light receiving element 93 and a remote control signal is supplied to the microcomputer 91.

In this way, when the key of the transmitter 94 is operated, the same processing as when the key 92 is operated is performed.

Furthermore, analog stereo audio signals L and R that have been encoded for surround playback or general analog stereo audio signals L and R are supplied to the analog input terminals 1A to 1N, respectively. . These signals L and R are supplied to the switch circuit 41, and the switch circuit 41 is controlled by a control signal from the microcomputer 91 to terminals 1A to 1N.
A pair of signals L and R is selected from among the audio signals, the selected signals L and R are supplied to the A/D converter 42 and converted into a digital audio signal, and this signal is supplied to the switch circuit 43 be done.

Furthermore, the digital input terminals 2A to 2M have
Similar digital audio signals are respectively supplied, and these signals are supplied to the switch circuit 43.

This switch circuit 43 is controlled by a control signal from the microcomputer 91 to connect the terminals 1A to 1N,
A pair of signals L and R from the 2A to 2M audio signals (digital audio signals) is selected, and the selected signals L and R are sequentially supplied to the DSPs 51 to 65.

The DSPs 51 to 65 are microcomputers dedicated to digital audio signal processing, and are each implemented as a one-chip IC. These DSPs 51 to 65 take in programs or data sent from the microcomputer 91, and perform predetermined signal processing on the digital audio signals passing through the DSPs according to the programs or data.

That is, for example, when a user operates a key 92, a processing program or data is loaded from the microcomputer 91 into one of the DSPs 51 to 65 that corresponds to the key operation. It processes digital audio signals according to programs or data.

[0031] At this time, these DSPs 51 to 6
5 works as circuits 21-24, 29, and 31-33 in FIG.

That is, the DSP 51 mainly functions as a balance adjustment circuit 21, a test tone signal forming circuit 29, and a switch 22. (1) Forming a test tone signal (2) Automatically adjusting the level balance between sources (3) )
〃 〃
Manual adjustment (4) Performs dynamic range correction (signal expansion/compression), etc. And this (
The test tone signal formation, balance adjustment, and dynamic range correction in sections 1) to (4) are performed selectively, and as described above, when the key 92 is operated, D
It is switched by changing the program or data of SP51.

Further, the DSPs 52 to 56 function as the Dolby Pro Logic decoder 23 to perform surround processing, and the DSP 55 outputs the front signals FL and FR, and the DSP 56 outputs the front center signal FC and the difference component (L- R) is output. In addition, this DSP52~5
The surround processing of No. 6 can also be turned on and off by operating a key 92. Also, the DSP51 (
2) When performing automatic adjustment of the level balance between the term sources, a predetermined signal is fed back from the DSP 52 to the DSP 51 and used for the automatic adjustment.

The DSPs 61 to 65 are mainly used to create or correct a sound field. That is,
The DSPs 61 and 62 perform parametric equalizer processing (
The DSPs 63 to 65 perform equalizer processing to change the center level and center frequency using bandpass characteristics or band elimination characteristics, and also add reverberation to reproduce acoustic characteristics in a hall or the like.

The front channel signals FL and FR are taken out from the DSP 63, and the center channel signal FC is taken out from the DSP 64. Also, DSP
65 is a decoder 31, a delay circuit 32, and a switch 33
It also functions as a rear channel signal BL and BR by performing level expansion, high frequency cut, addition of reverberation, etc. on the difference component (LR), and outputs rear channel signals BL and BR.

Note that the processing amount in the DSPs 61 to 65 can be changed in, for example, 30 ways by operating the key 92.

Then, the signals FL and FR from the DSP 63
are supplied to the D/A converter 71 and converted into analog signals FL and FR, and these signals FL and FR are taken out to the terminal 81 through the muting circuit 74, and the signal FC from the DSP 64 is input to the D/A converter 71. It is supplied to the converter 72 and converted into an analog signal FC, and the signal FC is sent to the terminal 82 through the muting circuit 75.
It is taken out. In addition, the signals BL and B from the DSP65
R is supplied to the D/A converter 73 and converted into analog signals BL and BR, and these signals BL and BR are taken out to the terminal 83 through the muting circuit 76.

According to this configuration, according to the operation of the key 92, the microcomputer 91 sends information to the DSPs 51 to 65.
By loading a predetermined program or data and executing the program in the DSPs 51 to 65, the following operation modes can be realized.

■ Surround effect and sound field creation or correction mode (Part 1) In this case, as described above, Dolby Pro Logic processing is performed in the DSPs 52 to 56, and the DSPs 61 to 56
At 65, sound field creation or correction processing is performed, and signals FL, FR, FC, BL, and BR are output.

Therefore, as shown in FIG. 2A, the speakers SPFL and SP are placed on the left and right sides in front of the listener LSNR.
FR, speaker SPFC placed in the front center, speakers SPBL placed on the rear left and right, and signal FL from SPBR.
, FR, FC, BL, and BR sounds are output and surround playback is performed, and the sound field is
This corresponds to a processing amount of 65.

For example, when playing music, the DSP
The surround effects of 52 to 56 provide a sense of presence as if you were in a performance venue, and the sound field processing of DSPs 61 to 65 allows you to set the performance venue as if it were a large hall.

■ Surround effect and sound field creation or correction mode (Part 2) In this case as well, the same processing as in ■ is performed, but the signal FC is not output, and in the DSP 63, the signal FC is changed to the signal FL, The signals are mixed at a predetermined ratio with respect to FR, and the signal line of the signal FC is turned off (muting state) in the DSP 64.

Therefore, in this case, the speakers SPFL, SPFR, and SPBL arranged as shown in FIG. 2B
, SPBR performs surround reproduction and sound field creation or correction similar to (2).

① Mode in which only surround effects are performed In this case, no signal processing is performed in the DSPs 61 to 65. That is, the front channel signals FL and FR from the DSP 55 pass through the DSPs 61 → 62 → 63, but are supplied to the D/A converter 71 without any sound field processing, and the signals FL and FR are sent to the terminal 81. taken out.

[0045] Also, the difference (L-R) from DSP56
passes through the DSP 61 → 62 → 65, but is supplied to the D/A converter 73 without any sound field processing, and the difference (LR) is output to the terminal 83 as signals BL and BR of the rear channels. It will be done. Note that the signal line of the DSP 64 is turned off.

Therefore, in this case, the speakers SPFL, SPFR, and SPBL arranged as shown in FIG. 2B
, SPBR performs surround reproduction.

■ Mode in which only sound field creation or correction is performed In this case, no signal processing is performed in the DSPs 52 to 56. That is, the signals L and R from the DSP 51
is supplied to the DSP 61 without undergoing surround processing in the DSPs 52 and 53.

Then, from now on, for the signals L and R, DS
P61 to P65 perform the process of creating or correcting the sound field as described above, and the output signals of converters 71 and 72 become
Front channel signals FL, FR, F are connected to terminals 81 and 82.
The output signal of the converter 73 is taken out as the rear channel signals BL and BR at the terminal 83.

In this mode, by changing the amount of processing in the DSPs 61 to 65 by operating the keys 92 as shown in (1) and (2), the signal FL is processed without performing any processing on the signals L and R. , FR, FC or signal BL
, BR, or can be output after performing the maximum amount of processing.

Therefore, in this mode, various sound fields can be realized, from an uncorrected general two-channel stereo sound field to a newly created sound field.

■ Mode in which no surround effect or sound field processing is performed This is a case in which no signal processing is performed;
Signals L and R from the SP 51 pass through the DSPs 52 to 63, but are output to the terminal 81 as they are without being processed at all. That is, this is a case of general two-channel stereo playback.

FIGS. 3 and 4 show that (
1) to (4) A processing routine of the microcomputer 91 for selectively executing the processing in sections 1) to 4) is shown.

That is, the processing of the microcomputer 91 starts from step 101, and then in step 102,
It is checked whether there is any input from the key 92, and if there is no key input, the process repeats step 102.

When step 102 is repeated, if there is an input from the key 92, this is determined in step 102, and the process proceeds from step 102 to step 111, where the key input in step 102 is (1 ) A check is made to see if it was a key input instructing the formation of a test tone signal, and if not, the process proceeds from step 111 to step 121.

In step 121, it is checked whether the key input in step 102 is a key input instructing automatic adjustment of the level balance between the sources in item (2), and if not, the process returns to step 121. The process advances from step 121 to step 131, and in step 131, the key input in step 102 is changed to (3
) A check is made to see if there was a key input instructing manual adjustment of the level balance between the sources of the term; if not, the process continues from step 131 to step 14.
Go to 1.

Furthermore, in this step 141, it is checked whether the key input in step 102 is a key input instructing the correction of the dynamic range in item (4), and if not, the process returns to step 14.
1 to step 150, and in step 150, the process corresponding to the key input in step 102 ((1
) to (4) (other than items 4) are executed, and then the process of the microcomputer 91 returns to step 102 and waits for a key input again.

On the other hand, in step 111, if the key input in step 102 is the key instructing the formation of the test tone signal in item (1), the process proceeds to step 1.
11, the process proceeds to step 112, and in step 112, the muting circuits 74 to 76 are turned on. Subsequently, in step 113, the program and data for forming a test tone signal are transferred from the microcomputer 91 to the DSP 51, and then in step 114, the muting circuits 74-76 are turned off.

[0058] Also, at this time, in the DSP 51,
In step 113, when the program and data for forming a test tone signal are transferred from the microcomputer 91, the program is executed to form a test tone signal to be output as the front left channel signal LF, for example.

Therefore, the digital audio signals L and R from the switch circuit 43 are muted in the DSP 51, and a test tone signal is formed in the DSP 51, and the test tone signal is outputted to the terminals 81-83. Therefore, for example, a test tone is output from the front left speaker SPLF.

Next, in step 115, the counter is cleared to 0, and then in step 116, it is checked whether there is any input from the key 92. If there is no key input, the process moves from step 116 to step 117. Then, in step 117, it is checked whether the counter in step 115 has elapsed for one second.

[0061] Then, when one second has not passed,
The process returns from step 117 to step 116, and thereafter steps 116 and 117 are repeated until the key 92 is input or until one second has elapsed.

However, when one second has elapsed, the process proceeds from step 117 to step 118, and this step 11
8, a command instructing to switch the output channel of the test tone signal to the next channel is sent to the DS.
P51, and then the process returns to step 115.

Therefore, when there is no input from the key 92, steps 115 to 118 are repeated, and the channels to which the test tone signal is output are changed every second to the front left channel, center channel, front right channel, and rear channel. It can be switched, so the test tone is 1
The output is sequentially output from the speakers of each channel every second.

[0064] In this way, steps 115 to 11
8 is being repeated, if there is an input from the key 92, this is determined in step 116, and the process proceeds from step 116 to step 119. In step 119, the DSP 51 is instructed to finish forming the test tone signal. A command to do so is provided, after which processing returns to step 102.

Therefore, the DSP 51 finishes forming the test tone signal and returns to the state of supplying the digital signal from the switch circuit 43 to the DSP 52.

Therefore, when steps 115 to 118 are repeated and test tones are being output from each channel in sequence, when key 92 is operated, the test tones are no longer output and the normal operating state is returned.

Further, in step 121, if the key input in step 102 is a key input instructing automatic adjustment of the level balance between sources in item (2), the process proceeds from step 121 to step 122. , in this step 122, the muting circuits 74-
76 is turned on. Subsequently, in step 123, the program and data for automatically adjusting the level balance between the sources are transferred from the microcomputer 91 to the DSP 51, and then in step 124, the muting circuits 74 to 76 are turned off, and then the processing is step 10
Return to 2.

At this time, when the program and data for automatically adjusting the level balance between the sources are transferred to the DSP 51 in step 123, the program is executed. Therefore, the level balance between the sources is automatically adjusted, the result is set in the DSP 51, and the level will not change even if the source is switched thereafter.

Further, in step 131, if the key input in step 102 is a key input instructing manual adjustment of the level balance between the sources in item (3), the process proceeds from step 131 to step 132. ,
In this step 132, the muting circuit 74
~76 are turned on. Subsequently, in step 133, a program and data for manually adjusting the level balance between the sources are transferred from the microcomputer 91 to the DSP 51, and then in step 134, the muting circuits 74 to 76 are turned off, and then processing is step 1
Return to 02.

[0070] Therefore, by operating the key 92, the input level for each source is set in the DSP 51, and the result is set in the DSP 51. Thereafter, when the source is switched, the set level will be used.

Further, in step 141, if the key input in step 102 is a key input instructing correction of the dynamic range in item (4), the process proceeds from step 141 to step 142, and this step 1
At 42, muting circuits 74-76 are turned on. Subsequently, in step 143, the microcomputer 9
1 transfers the program and data for dynamic range correction to the DSP 51, and then in step 144 the muting circuits 74 to 76 are turned off,
Thereafter, the process returns to step 102.

Therefore, by operating the key 92, the input/output characteristics of the DSP 51 are set to those for correcting the dynamic range, and the result is set in the DSP 51. From then on, the DSP 51 uses the set input/output characteristics. Becomes a characteristic.

[0073]

As described above, according to the present invention, all surround processing is realized by digital processing, so there is no need for adjustment during assembly, and the best characteristics can be obtained without adjustment. Moreover, no change occurs over time.

Furthermore, each of the DSPs 51 to 65 can be constructed from one chip of IC, and these
Since general-purpose DSP can be used as SP51-65,
The cost is low and the device does not become large. Alternatively, the DSPs 52 to 56 may be constructed from a dedicated one-chip IC.

Furthermore, since all processes other than signal selection in the switch circuit 41 are digitally processed, there is no signal deterioration, such as a decrease in S/N or an increase in distortion, and excellent sound quality can be maintained.

Furthermore, by selectively supplying various programs and data to the DSP 51, (1) to (
4) Formation of the test tone signal, automatic adjustment of the level balance between sources, manual adjustment, and correction of the dynamic range are selectively performed, so there is no waste and the number of parts can be reduced from this point of view as well. It can be reduced and the cost can be reduced.

[0077] In addition, the creation or correction of the sound field is performed using the DSP6.
1 to 65, predetermined signal processing can be performed by simply changing the program or data, and detailed and diverse sound field processing can be performed.

Furthermore, when creating or correcting a sound field, the signals FL and
Since R, FC, and (LR) are supplied in series, the hardware is simple, and the equalizer processing and reverberation processing can be performed independently and appropriately in the DSPs 61 to 65.

Furthermore, since it is possible to turn on/off the surround processing in the DSPs 52 to 56 and to adjust the sound field processing in the DSPs 61 to 65 independently, it is possible to respond to a wider variety of sources or demands. Can be done.

[Brief explanation of drawings]

FIG. 1 is a system diagram showing an example of the present invention.

FIG. 2 is a plan view for explaining a sound field.

FIG. 3 is a flowchart showing part of a processing routine.

FIG. 4 is a flowchart showing part of a processing routine.

FIG. 5 is a system diagram showing a conventional example.

[Explanation of symbols]

1A~1N Analog input terminal 2A~2M Digital input terminal 41 Switch circuit 42 A/D converter 43 Switch circuit 51-65 DSP 71-73 D/A converter 74-76 Muting circuit 81-83 Output terminal 91 Microcomputer 92 Operation keys 94 Remote control transmitter

Claims (3)

[Claims] 1. An input terminal to which a plurality of encoded digital audio signals for surround playback are supplied, forming a test tone signal, adjusting the level balance between the plurality of digital audio signals, and a dynamic range. a first DSP that performs correction, a switch circuit that selectively supplies the plurality of digital audio signals to the first DSP, and a decode circuit for decoding the output signal of the first DSP for surround stereo reproduction. A second DSP that performs processing, and a third DSP that performs sound field creation or correction processing on the output signal of this second DSP.
A surround audio device comprising: a D/A converter that converts the output signal of the third DSP into an analog audio signal; and an output terminal that outputs the analog audio signal from the D/A converter. 2. An input terminal to which a plurality of encoded digital audio signals for surround playback are supplied, a switch circuit that selectively outputs the plurality of digital audio signals, and an output signal of the switch circuit. , a first DSP performs decoding processing for surround stereo reproduction, and a second DSP performs equalizer processing.
SP, a third DSP that performs reverberation processing, a D/A converter, and an output terminal that outputs an analog signal from the D/A converter, and the output signal of the first DSP is A surround audio device in which a digital signal is supplied in series to a second and third DSP to undergo the equalizer processing and the reverberation processing, and the digital signal is supplied to the D/A converter. 3. An input terminal to which a plurality of encoded digital audio signals for surround playback are supplied, forming a test tone signal, adjusting the level balance between the plurality of digital audio signals, and a dynamic range. a first DSP that performs correction, a switch circuit that selectively supplies the plurality of digital audio signals to the first DSP, and a decode circuit for decoding the output signal of the first DSP for surround stereo reproduction. A second DSP that performs processing and a third DS that performs equalizer processing
P, a fourth DSP that performs reverberation processing, a D/A converter, and an output terminal that outputs an analog signal from the D/A converter, and the output signal of the second DSP is A surround audio device, wherein the digital signals are supplied in series to the third and fourth DSPs to undergo the equalizer processing and the reverberation processing, and the digital signals are supplied to the D/A converter.