JPH09200055A - Audio data decoder - Google Patents

Abstract

PROBLEM TO BE SOLVED: To improve reproduction of audio data through the use of a D/A converter with a simple configuration by providing an interpolation data generating means to interpolate basic time base information through the similar arithmetic operation to that of a frequency/time conversion means in the case of reproducing audio data. SOLUTION: A multiplexer means 3 conducts multiplex processing between basic time base information generated by a frequency/time conversion means 1 and interpolation data generated by an interpolation data generating means 2. Digital decoded audio data are generated by applying prescribed processing to the output data from the multiplexer means 3. Furthermore, interpolation data are generated simultaneously in addition to basic time information according to the coding standard by applying finer setting to a processing rate of the frequency/time conversion than the processing rate according to the standards. Thus, many interpolation data are obtained by using a finner rate without the use of a complicated 1-bit D/A converter system to obtain the interpolation data.

Description

Detailed Description of the Invention

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to an audio decoding device, and more particularly to an oversampling method for decoding audio data encoded in the frequency domain using a time / frequency conversion technique.

[0002]

2. Description of the Related Art A variety of audio signal encoding methods have been known. As an example,
There is a method of converting an audio signal from a time domain signal to a frequency domain signal and performing coding in the frequency domain. As a method for performing time / frequency conversion, for example, a subband filter or MDCT (Modified Discrete Cosine Transf
orm) is used, and MPEG (Moving Picture Image Coding) is used as a coding method using such a method.
Experts Group) Audio can be mentioned.

In the above-mentioned MPEG audio layer I,
In order to efficiently use the psychoacoustic model such as the critical band (the frequency width that the masking effect that the hearing sensitivity decreases at the frequency near the peak of a certain frequency spectrum), the entire band is set to 32 equally spaced frequency bands Will be divided. Then, the divided signals in each band are sub-sampled at 1/32 of the original sampling frequency and encoded.

Decoding of audio data encoded in this way at a predetermined sampling rate is basically performed by an operation reverse to the above encoding. FIG.
FIG. 4 is a block diagram showing a configuration of a conventional MPEG audio decoding device so that a processing flow can be easily understood. In this example, the sampling frequency is 44.1KHz.
, And the audio data is encoded with a bit width of 16 bits.

In FIG. 6, the encoded audio data is first input to the unpack circuit 51.
Generally, audio data encoded by MPEG audio mainly includes allocation, scale factor, and sample.
It is composed of The unpack circuit 51 separates and extracts each data of the allocation, scale factor, and sample from the bit stream of the input encoded audio data.

Each data separated by the unpacking circuit 51 is then input to the frequency / time conversion circuit 52. In the frequency / time conversion circuit 52, the subband information S _k that is a signal in the frequency domain is obtained based on each data input from the unpack circuit 51, and further the subband information S _k is calculated according to the following (Equation 1). _{From k} , a V vector V [i] which is a signal in the time domain is obtained.

[0007]

[Equation 1]

The V vector obtained by the frequency / time conversion circuit 52 is temporarily stored in the V buffer 53 and then supplied to the filter circuit 54, where it is subjected to a filtering process using a predetermined filter coefficient. As a result, digital PCM data (44.1 KHz) is generated. Then, the PCM data thus obtained is converted into an analog signal by the 16-bit DAC (digital-analog converter) 55 and output.

[0009]

However, in the conventional audio decoding device configured as described above, aliasing noise may occur in the vicinity of the frequency of 1/2 of the sampling frequency, and the reproduced analog signal of the reproduced analog signal may be generated. The waveform was sometimes distorted. Therefore, when the encoded audio data is decoded and the audio signal before the encoding is reproduced, there is a problem that the reproducibility of the voice deteriorates.

For example, at a sampling rate of 44.1 KHz
When the 20 KHz cosine wave is decoded, the analog audio signal output from the 16-bit DAC 55 in FIG. 6 has a waveform as shown in FIG. As compared with FIG. 10 showing the waveform before encoding, it can be seen that the reproducibility of voice is significantly deteriorated.

Conventionally, in order to solve such a problem,
A technique has been considered in which a 1-bit DAC system 56 as shown in FIG. 7 is used instead of the 16-bit DAC 55 in FIG. 1-bit DAC system 56
Is provided with an interpolator 59 including a FIFO memory 57 and a multiplier / adder 58, and a DAC 60.

The 1-bit DAC system 56 is an MP
The PCM data output from the EG audio decoder 50 is accumulated in the FIFO memory 57 to some extent, and the accumulated PCM data is digitally filtered by the multiplier / adder 58. As a result, interpolation data in which data values between discrete real data are estimated is obtained, and D / A conversion is also performed by the DAC 60 including the interpolation data to output an analog audio signal.

FIG. 8 is a diagram in which the configuration of the functional blocks shown in FIG. 7 is rewritten according to the hardware image. In FIG. 8, the same blocks as the blocks shown in FIG. 7 are designated by the same reference numerals.

The multiplier / adder 61 in the MPEG audio decoder 50 shown in FIG. 8 performs frequency / time conversion processing in the frequency / time conversion circuit 52 in FIG. 7 and predetermined filter processing in the filter circuit 54. Is. As the various coefficients necessary for performing those processes, those stored in the coefficient ROM / RAM 62 are used.

The memory 63 shown in FIG. 8 includes a work memory used when performing the frequency / time conversion process and a predetermined filter process, and the V buffer 53 shown in FIG. The PCM data output unit 64
The PCM data generated by the above-described predetermined filter processing and stored in the memory 63 is transferred to the MPEG audio decoder 5
0 is output to the outside.

On the other hand, the coefficient ROM / RAM 65 in the 1-bit DAC system 56 shown in FIG.
8 stores filter coefficients and the like used when performing digital filter processing. It should be noted that a plurality of types of filter coefficients are stored, and the sound quality of the reproduced voice is determined to some extent depending on which is used.

By using the 1-bit DAC system 56 as shown in FIG. 7 or FIG. 8, it becomes possible to reproduce a waveform relatively close to the original waveform by using the interpolation data.
It is possible to reduce deterioration of sound quality.

However, when the 1-bit DAC system 56 is used, in addition to the DAC 60, a multiplier / adder 58 having a considerable computing capacity, a FIFO memory 57,
Since various configurations such as the coefficient ROM / RAM 65 are required, there is a problem that the circuit scale becomes large and the cost becomes high.

The present invention has been made in order to solve such a problem, and the reproducibility of an analog audio signal output through a decoding process and a D / A conversion process has a simple structure and is inexpensive. The purpose is to be able to improve by using.

[0020]

The audio decoding device of the present invention is an audio decoding device for decoding audio data coded in the frequency domain using time / frequency conversion, wherein the audio coded in the frequency domain is used. Frequency / time conversion means for performing frequency / time conversion processing on data to generate basic time axis information according to the standard, and when the basic time axis information is generated by the frequency / time conversion means Interpolation data generating means for generating time axis information for interpolating the basic time axis information by the same calculation as the frequency / time conversion processing is provided.

Another feature of the present invention is that time /
In an audio decoding device for decoding audio data encoded in the frequency domain using frequency conversion, frequency / time conversion processing for converting the audio data encoded in the frequency domain from information in the frequency domain into information in the time domain The frequency / time conversion processing rate is set more finely than the processing rate in the case of generating basic time axis information according to the standard.
Equipped with time conversion means.

Another feature of the present invention is decoding means for converting audio data encoded in the frequency domain using time / frequency conversion from information in the frequency domain into information in the time domain for decoding. And D / for generating interpolation data for interpolating the decoded audio data generated by the decoding means and converting digital audio data including the decoded audio data and the interpolation data into an analog audio signal. A
In an audio decoding device including a conversion means, a multiplier / adder and a memory used in the decoding means, and the D /
It is characterized in that the multiplier / adder and the memory used in the A conversion means are combined into one and shared.

Another feature of the present invention is that the frequency / time conversion processing performed by the decoding means is set to a processing rate that is finer than the processing rate in the case of generating basic time axis information according to the standard. It is characterized by doing so.

Since the present invention comprises the above technical means, in the frequency / time conversion process which is one of the series of processes for decoding the encoded audio data, the basic operation according to the encoding standard is performed. In addition to the time axis information, the interpolation data for interpolating the basic time axis information is generated by the same operation as the operation of the frequency / time conversion processing described above. 1 of the composition
It becomes unnecessary to use a bit DAC system.

According to another feature of the present invention, the frequency / time conversion processing is performed by setting the frequency / time conversion processing rate more finely than the processing rate according to the standard. In addition to the basic time-axis information according to the above, interpolation data for interpolating the basic time-axis information is generated at the same time, and in order to obtain the interpolation data, the 1-bit DAC having a complicated configuration is used. Not only does the system not need to be used, but the finer the setting of the processing rate, the more interpolation data can be obtained.

Further, according to another feature of the present invention, the number of multipliers / adders separately provided for the decoding means and the D / A conversion means is sufficient, and the decoding means and the D / A conversion means are provided.
The A-conversion means has a separate memory, and the hardware amount is shared in the audio decoding device configured to obtain interpolation data during D / A conversion. It will be reduced by that amount.

[0027]

BEST MODE FOR CARRYING OUT THE INVENTION An embodiment of an audio decoding device according to the present invention will be described below in detail with reference to the drawings.

FIG. 1 is a block diagram showing elemental features of the audio decoding apparatus according to the present embodiment. It should be noted that this audio decoding device is for decoding audio data encoded in the frequency domain using time / frequency conversion, and FIG. 1 shows one of the series of decoding processes. Only the part that performs a certain frequency / time conversion process is shown.

In FIG. 1, is a frequency / time conversion means, which performs frequency / time conversion processing on the audio data coded in the frequency domain, and outputs basic time axis information in accordance with the coding standard. To generate. For example, when the encoding method is MPEG audio, this frequency / time conversion means generates the V vector V [i] which is the basic time axis information according to the arithmetic expression based on the standard shown in (Equation 1). To do.

Further, is an interpolation data generating means, and by the same operation as the operation of the frequency / time conversion processing performed when the basic time axis information is generated by the frequency / time converting means, the basic time Interpolation data for interpolating axis information is generated. For example, the encoding method is MPE
In the case of G audio, this interpolation data generating means generates the interpolation data V [i] ′ according to (Equation 2) similar to the above (Equation 1).

[0031]

[Equation 2]

As shown in FIG. 1, the original data used when the interpolation data generating means generates the interpolation data is used when the frequency / time converting means generates the basic time axis information. It is the same as the original data.

Numeral 3 is a multiplexing means, which performs a process of combining the basic time axis information generated by the frequency / time conversion means with the interpolation data generated by the interpolation data generation means. Then, the data output from the multiplexing means is subjected to a predetermined process to generate digital decoded audio data. Then, it is converted into an analog audio signal by a D / A conversion means (not shown) and output.

As described above, according to the embodiment of FIG. 1, in the frequency / time conversion processing in the series of decoding processing,
In addition to the basic time-axis information according to the encoding standard, interpolation data for interpolating the basic time-axis information is generated at the same time by the same operation as the above-mentioned frequency / time conversion processing operation. Therefore, 1-bit D with complicated configuration
Interpolation data can be obtained without using an AC system, and the reproducibility of voice can be improved by using the interpolation data.

FIG. 2 is a block diagram showing an example of the configuration of a concrete audio decoding device which realizes the features of the present invention shown in FIG. FIG. 2 shows an audio decoding device in the case of adopting MPEG audio as an example of an encoding method using time / frequency conversion, in which audio data is encoded at a sampling frequency of 44.1 KHz. Be present.

As shown in FIG. 2, the MPEG of the present embodiment.
The audio decoder 1 includes an unpacking circuit 2 and a frequency /
It is composed of a time conversion circuit 3, a V buffer 4 and a filter circuit 5. The unpacking circuit 2 calculates an allocation and a scale factor (ScaleFac) from a bit stream of input encoded audio data.
tor) and sample data are separated.

Further, the frequency / time conversion circuit 3 obtains a V vector by performing synthesizer synthesis processing based on each data separated by the unpacking circuit 2. That is, in this synthesizer synthesizing process, the subband information S _k which is a signal in the frequency domain is obtained from each data separated by the unpacking circuit 2,
Further, according to the following (Equation 3), a V vector V [i] which is a time domain signal is _obtained from the subband information S _k .

[0038]

(Equation 3)

In this (formula 3), the step size of the sample i is set finer than in the case of the conventional (formula 1). That is, in (Equation 1), the step width of the sample i is 1, such as i = 0,1,2, ..., On the other hand, in (Equation 3), i = 0,0.
The step width of sample i is set to 0.5 as 5,1,1.5,2, .... Thus, in addition to the basic data corresponding to i = 0, 1, 2, ..., The interpolation data corresponding to i = 0.5, 1.5, ... Is calculated at the same time.

As described above, in this embodiment, 1-bit D
Rather than generating interpolated data in the D / A conversion process after decoding using the AC system, the step size of the sample is reduced in the frequency / time conversion process performed in the series of decoding processes. Oversampling is executed by calculation, and interpolation data is generated at the same time.

Further, the V buffer 4 temporarily stores the V vector obtained by the frequency / time conversion circuit 3. The filter circuit 5 performs a filtering process on the V vector stored in the V buffer 4 by using a predetermined filter coefficient, so that the digital PC
This is to generate M data. As shown in (Equation 3), in the frequency / time conversion circuit 3, the rate is
The frequency of the generated PCM data is 88.2KHz because the processing is performed by setting it to 2 in detail.

The DAC 6 connected to the subsequent stage of the MPEG audio decoder 1 thus constructed converts the digital PCM data output from the MPEG audio decoder 1 into an analog signal and outputs it. In the present embodiment, since it is not necessary to generate interpolation data at the time of D / A conversion, the configuration is complicated.
It is not necessary to use the bit DAC system, and it is possible to use as the DAC 6 a D / A converter that is simple in structure and inexpensive.

Here, when the audio data obtained by encoding the original cosine waveform shown in FIG. 10 is decoded by the MPEG audio decoder 1 of this embodiment, D
The waveform of the analog audio signal output from AC6 is shown in FIG.

As is clear from a comparison between the waveforms of FIG. 4 and FIG. 9, according to the present embodiment, a waveform closer to the waveform before encoding shown in FIG. 10 is obtained as compared with the conventional one. It is possible to improve the reproducibility of voice. Moreover, in the present embodiment, the reproducibility of voice can be improved without using a complicated DAC such as a 1-bit DAC system or providing any other additional configuration.

In the above embodiment, the double oversampling is realized by finely setting the step size of the sample i to ½ of the normal size as in (Equation 3). If the step size is set to 1 / M of the normal size, M times oversampling can be realized.

FIG. 5 shows D when the step size of the sample is set to 0.125 and 8 times oversampling is executed.
It is a figure which shows the waveform of the analog audio signal output from AC6. As is clear from FIG. 5, 2
Compared to the case of performing double oversampling, a waveform closer to the original sound can be reproduced, and the reproducibility of voice can be further improved.

As described above, in this embodiment, 1-bit D
Compared to the case where interpolation data is generated using the AC system, by arbitrarily setting the step size of the sample,
It is possible to generate more interpolation data, and there is an advantage that the reproducibility of voice can be significantly improved. Further, when the frequency / time conversion processing is performed, the interpolation data can be simultaneously generated according to the same arithmetic expression as that of the processing, so that it is not necessary to change the normal decoding processing process.

FIG. 3 shows another embodiment of the present invention, and is a diagram showing an example of a hardware configuration of an audio decoding device according to another embodiment. As shown in FIG. 8, when the 1-bit DAC system is used to generate the interpolated data, the MPEG audio decoder 50 and the 1-bit DAC system 56 are conventionally provided separately.

On the other hand, in the embodiment shown in FIG.
MPEG audio decoder 50 and 1-bit DAC
The hardware configuration is simplified by combining the configurations that have been duplicated in the system 56 into one.

That is, the multiplier / adder 11 of FIG. 3 is equivalent to the multiplier / adder 61 in the MPEG audio decoder 50 of FIG.
It is also used as the multiplier / adder 58 in the 1-bit DAC system 56. That is, the multiplier / adder 11 of FIG.
The synthesizer synthesizing process in the frequency / time conversion circuit 52 (the arithmetic process according to the above (Equation 1)), the predetermined filter process in the filter circuit 54, and the digital filter process in the multiplier / adder 58 are performed.

The coefficient ROM / RAM 12 shown in FIG.
Coefficient ROM in the MPEG audio decoder 50 of FIG.
The / RAM 62 also serves as the coefficient ROM / RAM 65 in the 1-bit DAC system 56. That is, the frequency / frequency in the frequency / time conversion circuit 52 of FIG.
Various coefficients necessary for performing time conversion processing, predetermined filter processing in the filter circuit 54, and digital filter processing in the multiplier / adder 58 are stored.

Further, the memory 13 of FIG. 3 has the MPE of FIG.
The memory 63 in the G audio decoder 50 also serves as the FIFO memory 57 in the 1-bit DAC system 56. That is, each processing in the above-mentioned multiplication / adder 11 of FIG. 3 is performed while using the memory 13 as a work memory.

The PCM data output unit 14 shown in FIG. 3 is a memory 13 generated by each processing in the multiplier / adder 11.
The PCM data stored in is output to the outside. The DAC 15 converts the digital PCM data output from the PCM data output unit 14 into an analog signal and outputs the analog signal, and corresponds to the DAC 60 shown in FIG. 7 or 8.

As described above, in the embodiment shown in FIG. 3, the conventional MPEG audio decoders 50 and 1 shown in FIG.
Since the configuration that is redundantly provided with the bit DAC system 56 is combined and shared as one, the amount of hardware can be reduced. Although the total size of the memory amounts of the coefficient ROM / RAM does not change between the case of FIG. 3 and the case of FIG. 8, the configuration can be simplified by combining them into one memory in FIG.

Further, in the embodiment shown in FIG. 3, since it has the function of the 1-bit DAC system, it is possible to generate the interpolation data using the decoded audio data, and the reproducibility of the voice deteriorates. Of course, it can be prevented.

As still another embodiment, a combination of the embodiment shown in FIG. 2 and the embodiment shown in FIG. 3 can be considered. That is, in the present embodiment, in the configuration as shown in FIG. 3, the synthesizer synthesizing process performed by the multiplier / adder 11 is performed according to (Equation 3) instead of (Equation 1).

In this way, both the interpolated data obtained by finely setting the step size of the sample when performing the frequency / time conversion processing and the interpolated data obtained based on the function of the 1-bit DAC system are used. Can be used to reproduce an analog audio signal, and it is expected that the reproducibility of voice can be further improved with a simple configuration.

In the above-mentioned embodiment, MPEG audio is taken as an example of one of the encoding methods.
If the encoding method adopts the time / frequency conversion method,
Since oversampling as described above can be performed during frequency / time conversion during decoding,
The encoding method does not matter.

For example, MDCT coding method, subband coding method, AC-3 coding method, or ATRAC.
The present invention can be applied to a transform coding method such as (Adaptive TRansform Acoustic cording).

[0060]

As described above, the present invention provides a frequency / time conversion means for performing frequency / time conversion processing on audio data encoded in the frequency domain to generate basic time base information in accordance with the standard. In addition, an interpolation data generating means for generating time axis information for interpolating the basic time axis information by the same calculation as the frequency / time conversion process performed when generating the basic time axis information is provided. Since it is provided, in the frequency / time conversion processing in the series of decoding processing,
In addition to the basic time-axis information according to the coding standard, interpolation data for interpolating the basic time-axis information can be obtained, without using a 1-bit DAC system having a complicated configuration. Interpolation data can be obtained.
Therefore, by utilizing the interpolation data obtained by the interpolation data generating means, it is possible to improve the reproducibility of the sound by using the inexpensive DAC with a simple configuration.

According to another feature of the present invention, frequency / time conversion processing for converting audio data encoded in the frequency domain from information in the frequency domain into information in the time domain is performed. Since the frequency / time conversion means for setting the processing rate of the frequency / time conversion more finely than the processing rate for generating the basic time base information according to the standard is provided, the 1-bit DAC having a complicated configuration is provided. Interpolation data can be obtained without using a system, and the more the processing rate is set, the more interpolation data can be obtained. Therefore, the DAC with a simple structure and at a low cost can be obtained. Can be used to further improve the sound reproducibility.

According to another feature of the present invention, a multiplier / adder and a memory used in the decoding means for decoding the coded data, and interpolation data of the decoded audio data generated by the decoding means. And a multiplier / adder and a memory used in the D / A conversion means for converting the digital audio data including the decoded audio data and the interpolated data into an analog signal. Since they are shared, the multiplier and the adder, which the decoding means and the D / A conversion means have separately, and the memory, which the decoding means and the D / A conversion means have separately, are provided one each. This is all that is needed, and a configuration for obtaining interpolated data at the time of D / A conversion and improving the sound reproducibility is realized with a small amount of hardware. It is possible.

[Brief description of drawings]

FIG. 1 is a block diagram showing the elementary features of the present invention.

FIG. 2 is a block diagram showing a configuration example of a specific audio decoding device that realizes the features of the present invention shown in FIG.

FIG. 3 is a block diagram showing a configuration example of an audio decoding device according to another embodiment of the present invention.

FIG. 4 is a diagram showing an example of a waveform of an analog audio signal obtained when double oversampling is executed in the embodiment of FIG.

5 is a diagram showing an example of a waveform of an analog audio signal obtained when 8 times oversampling is executed in the embodiment of FIG.

FIG. 6 is a block diagram showing a configuration of a conventional audio decoding device.

FIG. 7 is a block diagram showing a configuration when a 1-bit DAC system is used to solve a conventional problem.

8 is a block diagram showing a hardware image of the audio decoding device shown in FIG. 7.

9 is a diagram showing an example of a waveform of an analog audio signal obtained when a decoding process is performed by the audio decoding device in FIG.

FIG. 10 is a diagram showing an example of a waveform of an original audio signal before encoding.

[Description of Codes] Frequency / time conversion means Interpolation data generation means Multiplex means 1 MPEG audio decoder 2 Unpack circuit 3 Frequency / time conversion circuit 4 V buffer 5 Filter circuit 6 DAC 11 Multiplier adder 12 Coefficient ROM / RAM 13 Memory 14 PCM data output section 15 DAC

Claims (4)

[Claims] 1. An audio decoding device for decoding audio data encoded in a frequency domain by using time / frequency conversion, wherein the audio data encoded in the frequency domain is subjected to frequency / time conversion processing to conform to a standard. According to the frequency / time conversion means for generating the basic time axis information, and the same calculation as the frequency / time conversion processing performed when the basic time axis information is generated by the frequency / time conversion means, An audio decoding device, comprising: interpolation data generating means for generating time axis information for interpolating basic time axis information. 2. An audio decoding device for decoding audio data encoded in the frequency domain using time / frequency conversion, wherein the audio data encoded in the frequency domain is converted from frequency domain information into time domain information. A frequency / time conversion process for conversion is performed, and the frequency / time conversion process is performed by setting the frequency / time conversion processing rate more finely than the processing rate for generating basic time axis information according to the standard. An audio decoding device comprising time conversion means. 3. Decoding means for converting audio data encoded in the frequency domain using time / frequency conversion to information in the time domain from information in the frequency domain for decoding, and generated by the decoding means. Audio decoding including interpolation data for interpolating the decoded audio data, and D / A conversion means for converting digital audio data including the decoded audio data and the interpolation data into an analog audio signal In the apparatus, an audio decoding device characterized in that the multiplying-adder and the memory used in the decoding means and the multiplying-adder and the memory used in the D / A conversion means are collectively combined into one. apparatus. 4. The frequency / time conversion process performed by the decoding means is performed by setting a processing rate finer than a processing rate in the case of generating basic time axis information according to the standard. The audio decoding device according to claim 3.