JPH08102677A - Sound signal decoder with mpeg standard - Google Patents

Abstract

PURPOSE: To make it possible to add a graphic equalizer function to MPEG sound decoder without increasing circuit size. CONSTITUTION: A scale factor(SF) changing circuit 102 for changing the value of an SF outputted from a decomposition extracting circuit 31a in accordance with the sound volume level of each required frequency band set up by an external CPU 103 based upon personal liking is prepared, and in a reverse quantization processing process, the value of the SF to be a multiplier for quantization data (sample) in each frequency band outputted from the circuit 31a is changed in each frequency band so that a sound signal component can be boosted or cut out to a required level in each frequency band. Consequently a graphic equalizer function can be added to a conventional MPEG sound decoder only by adding a small circuit for changing the value of the SF.

Description

Detailed Description of the Invention

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to an audio signal decoder for decoding a compressed and encoded digital audio signal, and is particularly suitable for decoding an MPEG standard audio signal.

[0002]

2. Description of the Related Art In recent years, a technique called MPEG has been developed as one of compression techniques for digital audio signals. This MPEG audio signal compression technology is based on IS
It is standardized by O / IEC (International Standards Organization / International Electrotechnical Commission), and is MPEG1 and MPEG2.
Including both. Hereinafter, these are collectively referred to as the MPEG standard.

In this MPEG standard, three systems of layer 1, layer 2 and layer 3 are prepared depending on the required sound quality and circuit scale. The main difference between Layer 1 and Layer 2 is the difference in frame size of how many pieces of coded and sampled data of digitized information are put together. In addition, with respect to layer 3, in addition to the above frame size, the encoding method is also different.

In addition, the MPEG standard defines various modes such as single channel, dual channel, stereo, and joint stereo as the channel mode of the audio signal. The difference between these modes is due to the difference in whether the sound source is one, two, or stereo and the way of encoding stereo sound in the high range.

Furthermore, the MPEG standard defines the transmission rate that can be used for each channel mode of each layer. For example, in the single channel of layer 1, values such as 32K, 64K, 96K, 128K, 160Kbit / sec are defined. This difference in transmission rate affects the amount of data after compression. The output rate (sampling rate) of the audio signal after playback is 32K, 44.1K, 48
Three types of KHz rates are specified.

The principle of compression and decoding of a voice signal adopted in the MPEG standard is as follows. One frame voice signal is compressed and decoded under layer 1, single channel, transmission rate 128 Kbit / sec and sampling rate 48 KHz. A case will be described as an example.

First, a digital audio signal for one frame consisting of 384 samples is divided into 32 sub-bands by a sub-band filter, and 12 sub-bands for each sub-band.
Get the sample value. Then, a scale factor (SF) is obtained from the maximum amplitude of each subband, and each data is normalized by the scale factor (SF) of that subband. Also, the bit amount per frame is calculated from the transmission rate and the sampling rate.

Also, the minimum and minimum hearing characteristics and masking characteristics based on the human psychoacoustic model (the former is not very sensitive in the low frequency and high frequency regions of human hearing, the latter is the frequency near the peak of a certain frequency spectrum). The quantization level (Allocation) is set for each sub-band so as to match the bit amount of one frame, based on the characteristic that the hearing sensitivity decreases.

Next, according to the quantization level (Allocation) set in this way, the normalized sample for each sub-band is quantized and coded to obtain quantized data (Sample). Then, this quantized data (Sample) is made into one data stream together with the scale factor (SF) and the quantization level (Allocation) described above, and a compressed audio signal is generated.

Here, the above-mentioned quantization level (Allocatio
n) is data used when performing an inverse quantization operation at the time of decoding, and represents how many bits are allocated to each sample included in one subband.

The scale factor (SF) is data used when performing an inverse scaling operation at the time of decoding, and gives an approximate output level. For example, it corresponds to the exponent part of a floating point number used in an electronic computer. The scale factor (SF) has a fixed value for each subband, and the value is 0 to
63 integers of 62 are used.

Further, the quantized data (Sample) is data used in the inverse quantization operation at the time of decoding and gives a fine value of the output data. For example, it corresponds to the mantissa part of a floating point number used in an electronic computer.

On the other hand, the decoding of the data compressed as described above is performed by an operation reverse to this compression. That is, first, the data stream of the target compressed audio signal is decomposed and the quantization level (Allocatio
n), scale factor (SF) and quantized data (Sam
ple) is extracted. Then, the extracted quantized data (Sample) is inversely quantized by the quantization level (Allocation), and the inverse scaling process is performed by the scale factor (SF) to obtain the subband information.

Next, synthesizer synthesis processing is applied to the subband information obtained by the inverse quantization operation or the like. Then, after a predetermined process is performed by the polyphase filter, the waveforms before and after 16 are synthesized and output as a decoded speech signal.

FIG. 5 is a schematic block diagram showing a part of a conventional audio signal decoder for decoding a compressed audio signal as described above. The audio signal decoder of Figure 5, 31 denotes an input unit, by decomposition extraction circuit 31a provided therein, the data stream D _S from the quantization levels of the compressed audio signal input (Al
location), scale factor (SF) and quantized data (Sample) are decomposed and extracted. Note that such processing is performed under the control of the input controller 32.

Next, 33 is an inverse quantization unit,
The following processing is performed under the control of the inverse quantization controller 37. That is, the dequantization unit 33 performs dequantization and descaling processes by the dequantization processing unit 34 and the first multiplier 35 provided therein.

Here, the inverse quantization processing unit 34 uses the extracted quantization level (Allocation) to perform an inverse quantization operation as shown in (Equation 1) on the quantized data (Sample). Thus, the calculated value Sample-Value is obtained.

[0018]

[Equation 1]

Further, the first multiplier 35 uses the above-mentioned scale factor (SF) to perform the inverse quantization processing unit 34.
Monomials 2 ^{1-SF / 3} to the arithmetic value Sample-Value determined by
Subband information S _j (j is 0
A value of ˜31 indicates 32 subband numbers).

At this time, the value SF-TBL (Scale-Factor) stored in advance in the first table ROM 36 as table information is used as the value of the above-mentioned monomial expression 21 ^{-SF / 3} . In addition, this table information SF-TBL (Scale-Factor),
Based on the given scale factor (SF), the monomial expression 2
^{1-SF / 3} differs from the actual calculated value only in the range of rounding error. Therefore, in the MPEG-Audio standard, values are given in a numerical table.

Next, reference numeral 38 is a buffer memory unit, which is a first buffer memory (RAM) inside the unit.
The subband information S _j obtained by the inverse quantization unit 33 is stored in 39. The subband information S _j stored in the first buffer memory 39 is used when synthesizer synthesis is performed in the next stage.

The subband information S _j stored in the first buffer memory 39 is supplied to the synthesizer synthesis unit 41 at the next stage under the control of the buffer memory controller 40. This synthesizer synthesis unit 41
Is provided with a second multiplier 42, a cumulative adder 43, and a second table ROM 44. With these, waveform information V _i is obtained from the frequency of each subband and the phase of the target time. It is designed to be done.

Specifically, first, the second multiplier 42 adds cos ((16 + i) × (2j + 1) π to the subband information S _j supplied from the first buffer memory 39. / 64) value is multiplied. The value of such cos ((16 + i) × (2j + 1) π / 64) stored in advance in the second table ROM 44 as table information is used. Note that i in the above equation is a parameter (sample number) in the time axis direction and takes a value of 0-63. Further, j is a parameter (subband number) in the frequency axis direction and takes a value of 0 to 31.

Next, the cumulative adder 43 adds all the multiplication values for each sub-band obtained by the second multiplier 42 to obtain the waveform information V _i . Note that a series of arithmetic processing in the synthesizer synthesis unit 41 is performed under the control of the synthesizer synthesis controller 45. Then, the waveform information V _i obtained by the synthesizer synthesis unit 41 in this way is stored in the second buffer memory 47 inside the output unit 46 of the next stage.

Next, the input unit 31 and the dequantization unit 3 in the audio signal decoder configured as described above.
The operation of No. 3 will be described with reference to the flowchart of FIG.
In FIG. 6, first, in step P1, the decomposition extraction circuit 31
a, the quantization level (Allocation), scale factor (SF) from the data stream D _s of the compressed audio signal.
And quantized data (Sample) are extracted.

The quantization level (Alloca
section) and the quantized data (Sample) are supplied to the inverse quantization processing unit 34. The scale factor (SF) is also supplied to the first table ROM 36.

Next, in step P2, the inverse quantization processing unit 34 inversely quantizes the quantized data (Sample) in accordance with the above quantization level (Allocation), and the calculated value Samp
le-Value is required. Calculated value Samp thus obtained
The le-Value is given to the first multiplier 35.

In step S3, SF-TBL (Scale-Factor) is read from the first table ROM 36 according to the value of the scale factor (SF) given to the first table ROM 36 from the decomposition extraction circuit 31a as described above. Is read out and is supplied to the first multiplier 35.

Then, in step S4, the value of SF-TBL (Scale-Factor) read from the first table ROM 36 by the first multiplier 35 and the value from the inverse quantization processing unit 34 are given. Subband information S _j is obtained by multiplying the calculated value Sample-Value. The subband information S _j thus obtained is given to the first buffer memory 39 and stored therein for use in the subsequent processing.

[0030]

By the way, when a compressed audio signal is decoded by the audio signal decoder of the MPEG standard as described above and audio is output, by boosting or cutting the signal level of a specific audio band. , There are times when you want to create a sound field that suits your personal taste. However, in the conventional sound reproducing device, a processing device such as a graphic equalizer has to be added in order to create a desired sound field.

Therefore, there has been a problem that the circuit scale becomes very large and the hardware configuration of the apparatus becomes complicated. The present invention has been made to solve such a problem, and an object of the present invention is to allow an MPEG standard audio signal decoder to have a function of a graphic equalizer without increasing the circuit scale. And

[0032]

The MPEG standard audio signal decoder according to the present invention is designed to quantize level data, scale factor data, and quantized data divided into a plurality of frequency bands from a data stream of an MPEG standard compressed audio signal. An audio signal obtained by extracting each data of (1) and performing a dequantization process of multiplying the quantized data for each frequency band by a multiplier generated based on the scale factor to decode the compressed audio signal. In the decoder, an input unit for inputting setting information on a desired frequency band selected from the plurality of frequency bands and a desired boost amount or cut amount in the frequency band, and the setting input by the input unit. Depending on the information
Multiplier changing means for changing the value of the multiplier used in the inverse quantization process is provided.

Another feature of the present invention is that the value of the scale factor corresponding to the desired frequency band input by the input means is adjusted according to the desired boost amount or cut amount input by the input means. The multiplier changing means is configured so as to change the value.

Still another feature of the present invention is that the multiplier value generated based on the scale factor corresponding to the desired frequency band input by the input means is:
The multiplier changing means is configured to change by a shift operation according to a desired boost amount or cut amount input by the input means.

[0035]

The present invention utilizes the fact that the quantized data included in the data stream of the compressed audio signal is already divided into a plurality of frequency bands according to the MPEG standard, and thus the dequantization process at the time of decoding the compressed audio signal is performed. In the above, the multiplier for the quantized data sequentially supplied for each frequency band is changed according to the desired boost amount or cut amount, and only by changing the multiplier in this way, the audio signal component corresponding to the change. Is boosted or cut to a desired level for each frequency band.

[0036]

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS First, the principle of dequantization processing adopted by the MPEG standard audio signal decoder of the present invention will be described below by taking the case of layer 1 as an example.

As shown in FIG. 4, in the data stream generated in the compression process of the audio signal, the quantized data band-divided into 32 subbands 0 to 31 (indicated by the solid line in FIG. 4) is included. include. Then, the quantized data for each sub-band is sequentially input to the audio signal decoder from sub-band 0 to sub-band 31 and subjected to inverse quantization processing to be inverse quantized data for each sub-band. It

That is, assuming that the value (two's complement) obtained by inverting the most significant bit of the quantized data is S ''', the inverse quantized data S' (shown by the solid line + broken line in FIG. 4) is Is asked for. The value of the scale factor is constant for each subband, and its value is 0 to 0 according to the MPEG standard.
It is limited to 63 kinds of 62.

S ″ = (2 ^nb / 2 ^nb-1 ) × (S ′ ″ +2 ^{−nb + 1} ) nb is the quantization level data for each subband S ′ = factor × S ″ factor is a scale factor

Next, when trying to create a sound field that suits individual tastes, the inverse quantized data S'of a desired subband is multiplied by a coefficient f'corresponding to a desired boost amount or cut amount. Then, new dequantized data S such as S = f ′ × factor × S ″ is obtained.

In this way, by obtaining the dequantized data S corresponding to the desired boost amount or cut amount for each sub-band, the voice signal of an arbitrary sub-band (frequency band) is boosted to a desired level or I am able to cut it.

That is, by utilizing the fact that the quantized data included in the MPEG standard compressed audio signal is already band-divided into a plurality of sub-bands, the values determined for each of these sub-bands are used. By making the scale factor variable, the same effect can be obtained without adding a large-scale processing device such as a graphic equalizer.

The MPEG standard audio signal decoder according to each of the following embodiments is constructed using the above-described principle. FIG. 1 shows an MPEG according to the first embodiment.
3 is a diagram showing a schematic configuration of an audio decoder 100. FIG. Note that, in FIG. 1, components denoted by the same reference symbols as those shown in FIG. 5 have the same functions, and thus detailed description thereof will be omitted.

In FIG. 1, 103 is a general-purpose CPU (external CPU) externally attached to the MPEG audio decoder 100.
The desired frequency band and volume level (boost amount or cut amount) are arbitrarily set. Then, the setting information s1 relating to these frequency bands and volume levels is set to C
Transfer to the PU interface circuit 101.

The CPU interface circuit 101 described above
Is provided inside the MPEG audio decoder 100, receives setting information s1 for each frequency band from the external CPU 103, outputs predetermined control information s2 based on the setting information s1, and outputs the scale factor changing circuit 102.
Control.

Further, the scale factor changing circuit 102
Increases or decreases the value of the scale factor (SF) supplied from the decomposition extraction circuit 31a based on the control information s2 supplied from the CPU interface circuit 101.

Next, the operation at the time of inverse quantization in the MPEG standard audio signal decoder according to the present embodiment will be described with reference to FIG.
It will be described together with the flowchart of. First, step P
1. With the external CP, change the frequency band and volume level you want to change.
When set in U103, the setting information s1 of the sub-band corresponding to the frequency band and the boost amount or cut amount corresponding to the volume level is set from the external CPU 103
It is transferred to the PU interface circuit 101.

On the other hand, when decoding the PCM audio signal inside the MPEG audio decoder 100, first in step P2, the quantization level (Allocation), scale factor (SF) and 32 are calculated from the data stream D _s of the compressed audio signal. Quantized data (Sampl
e) is extracted by the decomposition extraction circuit 31a.

The quantization level (Alloca
tion) and the quantized data (Sample) are input to the inverse quantization processing unit 34. At this time, the extracted quantized data (Sample) for each sub-band is the inverse quantization processing unit 3 in order from sub-band 0 to sub-band 31.
4 is input. Further, the scale factor (SF) unique to each subband is calculated by the scale factor changing circuit 102.
Is input to

In the inverse quantization processing unit 34, the quantized data (Sample) for each sub-band input from the decomposition / extraction circuit 31a is similarly input to the quantized level (Al).
dequantized according to location) and the calculated value Sample-Value
Is required. This is the process in step P3.

Further, in the scale factor changing circuit 102, it is determined whether or not the subband of the quantized data (Sample) which is inversely quantized by the inverse quantization processing unit 34 matches the subband set in the CPU interface circuit 101. Depending on whether or not the quantized data (Samp
The value of the scale factor (SF) corresponding to the sub-band of (le) is changed.

That is, in step P4, the quantized data (Samp
It is determined whether the sub-band of (le) matches the sub-band set in the CPU interface circuit 101.
Then, when the subbands of both do not match, the process jumps to Step P6. In this case, the decomposition extraction circuit 31
The scale factor (SF) output from a is directly sent to the first table ROM 36 without any processing by the scale factor changing circuit 102.

On the other hand, when the subband of the dequantized quantized data (Sample) matches the subband set in the CPU interface circuit 101,
Go to step P5. In this case, the scale factor changing circuit 102 generates a new scale factor (SF ') from the scale factor (SF) output from the decomposition extraction circuit 31a.

The value of the scale factor (SF ') newly generated is determined according to the boost amount or cut amount transferred from the external CPU 103 to the CPU interface circuit 101. That is, the new scale factor (SF ') is obtained by adding or subtracting the value according to the boost amount or the cut amount to the original scale factor (SF) supplied from the decomposition extraction circuit 31a.

A new scale factor (SF ') may be obtained by multiplying the original scale factor (SF) by a coefficient corresponding to the boost amount or cut amount. In any case, the new scale factor (S
The value of F ′) is limited to take a value of 0 to 62 defined by the MPEG standard. For example, when boosting the volume level, it changes between 0 <SF '<SF, and when cutting the volume level, it changes between SF <SF'<62.

In this way, in the present embodiment, the scale factor changing circuit 102 for changing the value of the scale factor (SF) sent from the decomposition extraction circuit 31a according to the external input is provided, and the first table ROM 36 is provided. The scale factor (SF) for specifying the table information of the monomial expression 2 ^{1-SF / 3} to be read from is changed to a value different from the fixed value defined by the MPEG standard.

As described above, the scale factor (S
When F) or (SF ') is given to the first table ROM 36, in the next step P6 the first table ROM 36
The table information of the monomial expression 21 ^{-SF / 3} corresponding to the value of the scale factor (SF) or (SF ') is read from.

Then, in step P7, the first multiplier 3
5, the calculated value Sample-Value obtained by the dequantization processing unit 34 is multiplied by the value of the monomial expression 2 ^{1-SF / 3} read from the first table ROM 36 to obtain the subband information S _j. (J is a value of 0 to 31 and indicates 32 subband numbers).

By performing such processing for all subbands 0 to 31, each desired subband is
The audio signal component can be boosted or cut to a desired level. As described above, in this embodiment, a simple circuit is simply added, and a desired frequency band is boosted or cut with respect to a conventional MPEG standard audio signal decoder without significantly increasing the circuit scale. The function of the graphic equalizer can be added.

Next, a second embodiment of the present invention will be described with reference to FIG. MPEG according to the second embodiment
The audio decoder 110 is configured by using a control circuit 111 and a shifter 112 instead of using the scale factor changing circuit 102 shown in FIG.

The control circuit 111 controls the shifter 112 based on the control information s2 output from the CPU interface circuit 101. The control information s2 is
It is output based on the frequency band and volume level setting information s1 transferred from the external CPU 103. Further, the shifter 112, under the control of the control circuit 111, performs a shift operation on the value of the monomial expression 2 ¹ -SF ^{/ 3} read from the first table ROM 36 according to the value of the scale factor (SF).

In the present embodiment, the subband of the quantized data (Sample) which is inversely quantized by the inverse quantization processing unit 34 and the subband which is set in the CPU interface circuit 101 and whose volume should be changed coincide. In this case, the value of the monomial expression 21 ^{-SF / 3} read from the first table ROM 36 is shifted by the shifter 112. That is, only the value of the monomial expression 21 ^{-SF / 3} of the subband corresponding to the frequency band set in the CPU interface circuit 101 is shifted by the shifter 112.

The shift amount in this shift calculation is determined according to the boost amount or cut amount set in the CPU interface circuit 101, as in the first embodiment. That is, the first table R
The value supplied to the first multiplier 35 is set to a desired value by shifting the value of the monomial expression 2 ^{1-SF / 3} read from the OM 36 by an amount corresponding to the boost amount or the cut amount. Adjusted.

Thus, in this embodiment, the shifter 112 is
, The value of the monomial expression 2 ^{1-SF / 3} read from the first table ROM 36 can be arbitrarily changed for each subband according to the individual preference, so that the audio signal of the desired subband can be obtained. Can be boosted or cut to the desired level. Therefore, also in the present embodiment, as in the first embodiment, a graphic equalizer for boosting or cutting a desired frequency band is used for a conventional MPEG standard audio signal decoder without significantly increasing the circuit scale. Functions can be added.

Further, in this embodiment, since the multiplier to be supplied to the first multiplier 35 is changed by the shift operation, the operation load at the time of inverse quantization can be reduced and the processing can be performed at high speed. You will be able to do it.

[0066]

As described above, according to the present invention, for the quantized data for each frequency band extracted from the data stream of the MPEG standard compressed audio signal, the multiplier generated based on the similarly extracted scale factor is used. In the inverse quantization processing for multiplication, a multiplier changing means for changing the value of the multiplier used in the inverse quantization processing according to a desired boost amount or cut amount is provided, and the level of the inversely quantized data is changed as described above. Since it was changed arbitrarily for each frequency band by using the multiplier, the voice signal component can be boosted to the desired level for each frequency band simply by adding a small circuit for changing the multiplier. Or it can be cut, and the function of the graphic equalizer can be added to the audio signal decoder of the conventional MPEG standard without significantly increasing the circuit scale. It can be pressurized.

[Brief description of drawings]

FIG. 1 is a diagram showing a configuration of an audio signal decoder that is a first embodiment of the present invention.

FIG. 2 is a flowchart showing an operation of the audio signal decoder according to the first embodiment at the time of inverse quantization.

FIG. 3 is a diagram showing a configuration of an audio signal decoder which is a second embodiment of the present invention.

FIG. 4 is a diagram showing an example of quantized data divided into 32 subbands.

FIG. 5 is a diagram showing a configuration of a conventional audio signal decoder.

FIG. 6 is a flowchart showing an operation of the conventional audio signal decoder at the time of inverse quantization.

[Explanation of symbols]

 31a Decomposition / Extraction Circuit 34 Inverse Quantization Processing Unit 35 First Multiplier 36 First Table ROM 101 CPU Interface Circuit 102 Scale Factor Change Circuit 103 External CPU 111 Control Circuit 112 Shifter

Claims (3)

[Claims] 1. Quantization level data, scale factor data, and quantized data divided into a plurality of frequency bands are extracted from a data stream of an MPEG standard compressed audio signal, and the quantized data for each frequency band is extracted. In the audio signal decoder that decodes the compressed audio signal by performing the inverse quantization process that multiplies the multiplier generated based on the scale factor with respect to the desired frequency selected from the plurality of frequency bands. Of the multiplier used in the inverse quantization process according to the setting information input by the input unit, and the input unit for inputting the setting information regarding the desired frequency band and the desired boost amount or cut amount in the frequency band. An MPEG standard audio signal decoder, characterized in that a multiplier changing means for changing a value is provided. 2. The multiplier changing means so as to change the value of the scale factor corresponding to the desired frequency band input by the input means in accordance with the desired boost amount or cut amount input by the input means. The MPEG standard audio signal decoder according to claim 1, wherein 3. A multiplier value generated based on a scale factor corresponding to a desired frequency band input by the input means is shifted according to a desired boost amount or cut amount input by the input means. The MPEG standard audio signal decoder according to claim 1, wherein the multiplier changing means is configured to be changed by calculation.