JPH0946234A - Acoustic signal encoding and decoding method, acoustic signal encoding and decoding device - Google Patents

Abstract

PROBLEM TO BE SOLVED: To efficiently execute compression encoding by dividing a band, executing-encoding, making the digital acoustic signal of frequency area information, obtaining a residual signal between the digital acoustic signal and an original signal and multiplexing the residual signal with the digital acoustic signal. SOLUTION: Signals which are band-dividing to uniform 32 bands by a band dividing filter 1 are supplied to a maximum value selection part 2 and a quantizer 4. The selection part 2 outputs a maximum value S and the maximum value S decides the number of bits in a bit distribution part 3. The quantizer 4 reduces the number of the bits of respective pieces of sub-band data in the signal of the filter 1 based on bit distribution information from the distribution part 3. Sub-band data adjusted to lowest bit information by an inverse quantizer 5 are supplied to a band dividing filter 6 and are converted into the signals of time area information. A residual calculation part calculates an operation error occurred in band division/synthesis by the filters 1 and 6 and outputs it as the residual signal. A multiplexer 9 generates a bit stream for sub-data from the quantizer 4 and the residual signal from a calculation part 8.

Description

Detailed Description of the Invention

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to an encoding method and data decoding method for compressing a digital audio signal or the like, and more particularly to a digital audio signal (original signal) before encoding. The present invention relates to an audio signal encoding method and a decoding method thereof, which can reproduce a decoded signal obtained by decompressing the encoded signal in a state without information loss (Loss Less).

[0002]

2. Description of the Related Art Generally, a sound signal simply digitally converted has some continuity or redundancy unless it is random data such as white noise, and some lossless compression methods (Loss Less compression method) is known. The fact that adjacent samples on this time axis have correlation (continuity) can be easily understood by observing the acoustic signal. In addition, the amplitude distribution of the acoustic signal generally has a bias, and the digitized signal is
There is regularity in the occurrence probability that each bit is "0" or "1". When this bias increases, the redundancy of the acoustic signal also increases.

As an example of this reversible compression method, the difference PCM method is well known as a compression method that utilizes the continuity of music and that the difference value between adjacent samples is generally smaller than the sample value itself. . In addition, a prediction difference PCM method that predicts a target sample value from preceding and following sample values using the linearity, an adaptive prediction difference PCM method that adaptively determines a difference width, and the like are also known.

Further, as a method for reducing the redundancy by using a mathematical compression method, an entropy coding method using the bias of the appearance distribution of a digital signal, a pattern of a target signal and a past signal or preparation There is a vector coding method in which the signal is matched with the signal in the table and coded. Further, there is a method of performing coding by reducing the code amount by frequency-converting the acoustic signal in the time domain and adaptively distributing the information amount for each band based on the bias of the spectrum distribution.

However, the mainstream of the reversible compression method is the differential PCM method, and a compression method in which this is combined with an entropy coding method such as Huffman coding is used. The encoding method using the frequency conversion is mainly used for the irreversible high efficiency encoding method utilizing the psychoacoustic characteristics and the like.

[0006]

In such a conventional acoustic coding system, when the coding is performed only in the time domain, the coding utilizing the feature brought by the deviation of the frequency distribution of the acoustic signal cannot be carried out. Efficient compression could not be achieved. In particular, in the differential PCM method, when a signal having a large amplitude is present in the high frequency band, 17 differential bits are required to have a dynamic range equivalent to that of the 16-bit linear PCM method in order to express the differential value from the previous sample. Width needed. Therefore, there is a problem that encoding efficiency deteriorates when encoding an acoustic signal whose amplitude in the high frequency band is largely changed.

Further, even in the case of coding in which adaptive prediction is introduced, if a music source whose prediction accuracy is poor (difficult to predict) is coded, the differential bit width is expanded, and coding efficiency is not always improved. . In addition, the mathematical compression method does not fully utilize the redundancy caused by the continuity and correlation of speech, and when compared with the coding method that uses the characteristics of acoustic signals, the coding quality of mathematical compression is It was inferior (increased code amount).

Further, a coding system utilizing frequency conversion (particularly orthogonal conversion) is theoretically reversible, but in order to maintain reversibility, high quantization precision (computation precision) is required. Therefore, when the quantization width of the frequency domain information sample is expanded and, as a result, it is attempted to maintain the reversibility, there is a problem that the configuration becomes large and the compression rate is not improved.

In view of the above, the present invention uses a time domain correction for a frequency transform coding method that has been conventionally considered unsuitable as a lossless coding method, and thus an acoustic sound that maintains efficient and complete reversibility. It is an object to provide a signal coding method and a decoding method thereof.

[0010]

As means for achieving the above object, in the acoustic signal coding apparatus of the present invention, in order to utilize the deviation of the frequency distribution of the acoustic signal, the frequency domain information generating section digitally After band division of the acoustic signal, based on the energy value obtained for each band,
The sample value quantized by bit allocation according to the amount of information is encoded as information in the frequency domain, in the time domain information generation unit, the quantized sample value supplied from the frequency domain information generation unit is dequantized, By encoding the residual signal of the signal and the original signal after band synthesis to reconvert this into a time domain signal as time domain information, and multiplexing both of them in a multiplexing unit, Encoding lossless compression.

Further, in the audio signal decoding device of the present invention,
The supplied multiplexed signal is separated into frequency domain information and residual signal, and the frequency domain information is band-combined and then corrected by the residual signal to obtain the same digital acoustic signal as the original signal. It was done.

Since the post-decoding signal completely matches the original signal, the band synthesizing unit in the encoding device and the band synthesizing unit in the decoding device have a common configuration, and the calculation accuracy and the rounding process are the same. Taking the technique. As a result, the difference value between the band-combined signal and the original signal in the encoding device is obtained, and this is encoded and transmitted as information in the time domain, so that the decoding device corrects the time domain in the signal after the band combination. The added signal and the decoded signal become exactly the same as the original signal.

The lossless compression coding method and the decoding method for multiplexing and coding the information of both the frequency domain and the time domain of the acoustic signal, and the decoding method therefor, handle the acoustic signal in the frequency domain, thereby deviating the spectrum distribution of the signal. It is possible to effectively reduce the redundancy caused by irrespective of the type of signal (audio sources of various genres). Furthermore, by using the residual signal which is a time domain signal for correction, it is possible to surely return to the original signal. It is possible to restore. This method provides a completely lossless compression method compared with the case of encoding in the frequency domain alone, without requiring a procedure for strictly calculating the quantization accuracy of the frequency domain signal necessary to maintain reversibility. be able to.

[0014]

BEST MODE FOR CARRYING OUT THE INVENTION An embodiment of an acoustic signal coding method, an acoustic signal decoding method, an acoustic signal coding apparatus, and an acoustic signal decoding apparatus of the present invention will be described with reference to the drawings. FIG. 1 is a block diagram showing an embodiment of an audio signal encoding device of the present invention,
FIG. 6 is a flowchart showing the operation. An embodiment of the acoustic signal coding method will be described at the same time with reference to the drawings.

The audio signal coding apparatus shown in FIG. 1 is composed of a frequency domain information generating section A, a time domain information generating section B, and a multiplexing section (multiplexer) 9. The frequency domain information generation unit A includes a band division filter 1, a maximum value selection unit 2, a bit allocation unit 3, and a quantizer 4, and the time domain information generation unit B includes an inverse quantizer 5 and a band. It is composed of a synthesis filter 6, a delay unit 7, and a residual calculation unit 8. Further, the input digital audio signal is formed of blocks in which a frame is a processing unit. In this embodiment, the acoustic signal forming one frame is 512 samples per channel, and the band division width is 32 bands.

The least significant bit precision is fixed to an arbitrary bit width, but here it is fixed quantization precision of 16 bits, and information below this is rounded off to the upper bit (the 16th bit which is the least significant bit). reflect. The bit allocation information in the frequency domain represents 4 bits to represent a code 0 to 16 bits, and the time domain bit allocation represents 3 bits to represent a code 0 to 8 bits.

Here, the bit allocation width in the time domain depends on the characteristics of the filter, and is left to the number of multiplications during the product-sum operation for band synthesis. If the worst value of the accumulated error in this operation is within the time allocation bit width, the bit allocation width in the frequency domain guarantees the final quantization accuracy. In this embodiment, since encoding is performed with 3 bits, 16-bit quantization accuracy is guaranteed. If the bit allocation width is represented by 4 bits, it is possible to perform reversible compression of the input acoustic signal having a quantization accuracy of 24 bits. However, in this case, since the time domain information increases and the coding quality deteriorates, it is necessary to efficiently distribute the information amount with the frequency domain information.

A 4-bit frequency domain bit allocation table is shown in Table 1, and a 3-bit time domain bit allocation table is shown in Table 2. It should be noted that the code 0 indicates that there is no distribution, and if there is distribution, it is set between the code 2 and the code maximum value including the sign bit.

[0019]

[Table 1]

[0020]

[Table 2]

Next, the operation of the audio signal coding apparatus shown in FIG. 1 will be described. The input digital audio signal (original signal) is first supplied to the band division filter 1 and the delay device 7 (described later). In this band division filter 1,
The input signal, which is time domain information, is expanded into frequency domain information (step 101). Here, the sub-band filter is equally divided into 32 bands, and the band width W thereof is set as in the following equation. The precision of the subband data to be output is the least significant bit precision 1 as described above.
6 bits.

W = (sampling frequency × 0.5) / 32 (Hz)

Further, as the band division filter 1 and the band synthesis filter 6 which will be described later, for example, a filter for performing an orthogonal transformation such as DCT, a sub-band filter using the principle of the filter, a signal which is decomposed into a base waveform is used. The wavelet transform to be expressed, and further Fourier transform, which is a typical method of frequency transform, can be used. And
In the present invention, since the information component in the time domain is also used, it is not necessary to return the signal after the band division synthesis to the complete original signal, so any frequency conversion method may be used. In the present embodiment, a 512-tap subband filter (polyphase filter) will be used for concrete description. Note that the delay is generated by 480 samples through the band division synthesis.

The acoustic signal of the frequency domain information, which is equally divided into 32 bands by the band division filter 1, is supplied to the maximum value selection unit 2 and the quantizer 4. In the maximum value selection unit 2, for each of 32 bands existing in one frame, 16 (5
The absolute value comparison of certain sub-band data (amplitude value) or energy value is performed (12/32), and the maximum value S is selected and output (step 102).

The maximum value S of the subband data output from the maximum value selection unit 2 is supplied to the bit allocation unit 3. The bit allocation unit 3 refers to the maximum value S of each band in the frequency domain bit allocation table shown in Table 1 and
The number of bits allocated to each of the two bands is determined (step 103). The bit allocation here is performed corresponding to the minimum number of bits required to represent the maximum value sample in 2's complement notation, as shown in Table 1.

In the quantizer 4, the band division filter 1 is based on the bit allocation information supplied from the bit allocation unit 3.
The number of bits of each subband data of the acoustic signal of the frequency domain information supplied by the above is reduced (step 104). The reduction here is (16-the number of allocated bits) reductions of the upper bits that are the same as the sign bits, excluding the sign bits. Table 3 shows an example of reducing the number of bits of the subband data.

[0027]

[Table 3]

As shown in Table 3, when the sub-band data represented by a decimal number is 6,31,84, ..., 12,54, the maximum absolute value thereof is -94, and the bit allocation information. (Number of allocated bits) is 8 bits from Table 1. Therefore, of the 16-bit subband data, the data (8 bits) obtained by combining the lower 7 bits and the most significant bit (sign bit) as a sign bit is used as the quantized data, and the upper 2nd to 9th 8 bits are used. Is a reduction bit. This is a 1-bit sign bit part
It is equivalent to reducing the bit leaving.

Therefore, in the inverse quantization, it is only necessary to fill the reduced number of high-order bits with the designated number of sign bits according to the bit allocation information. The subband data and the bit allocation information quantized in this way are output to a multiplexer (multiplexing unit) 9 and, at the same time, inversely quantized in order to retransform the subband data into the time domain. Output to the container 5.

The inverse quantization in the inverse quantizer 5 is performed by adding the same code as the sign bit to the upper bits (16-the number of allocated bits) (step 10).
5). Therefore, in such quantization and dequantization, no constraint relating to the operation, such as rounding off, is performed during the operation, so that the quantization error does not occur.

The sub-band data aligned to the least significant bit precision (16 bits in this embodiment) by the inverse quantizer 5 is supplied to the band synthesizing filter 6 and converted into a signal of time domain information. It is converted (step 106).
In the band synthesizing process in the band synthesizing filter 6, it is necessary that the calculation precision, the filter coefficient precision, the calculating process, and the rounding process for the data of the output stage are completely matched with the band synthesizing filter 6 of the decoding device described later.

In general, D is used for digital audio signal processing.
SP (Digital Signal Processor) is often used.
Therefore, the band synthesizing process in the band synthesizing filter 6 will be described by taking the DSP as an example. FIG. 3 shows the internal operation block configuration of the DSP used for the band synthesis filter 6. The DSP used here is, for example, a fixed point of 16 bits × 16 bits, and the internal calculation precision and the data width inside the memory are 16 bits.

In band division synthesis by a subband filter or the like, conversion is mainly performed by a product-sum operation. Therefore, the band synthesizing filter 6 is composed of a multiplier, an adder, and a register of an input / output stage, and is used as a memory for various coefficients and a memory for intermediate data (operation memory) and an input / output bus required in the arithmetic process. Connected.

In the figure, input data (subband data) and filter coefficient data corresponding to this input data are sequentially input to the multiplier 21 via a 16-bit input data bus. As the input data, the data output from the dequantizer 5 is stored in a memory (not shown) with 16-bit precision, and the necessary data is the multiplier 21.
It is supplied to us from time to time. Also, the filter coefficient data is
Similarly, it is stored in advance in 16-bit precision in various coefficient memories (not shown). Then, the multiplier 21 multiplies the input data by the filter coefficient data and outputs it as 32-bit data.

The output of the multiplier 21 is 32 bits + α
Is supplied to the adder 23. This α indicates a high-order extension bit. A 32-bit bus is used for the product-sum operation, and a 16-bit bus is used for other operations. Then, the output from the adder 23 is temporarily accumulated in the register 22, and the output of this register 22 and the output of the next multiplier 21 are added by the adder 23 and stored in the register 22 for the next calculation. The state of being accumulated is repeated until there is no input data.

When the first-order sum-of-products calculation is completed, the output from the adder 23 is supplied to the register 24 as 32-bit precision data, rounded to 16-bit precision data, and output to a calculation memory (not shown). To do. This operation memory stores 16-bit precision data that has been rounded from 32 bits and outputs it to the multiplier 21 for performing the quadratic product sum operation. And in the same way,
When the quadratic product sum operation is performed, 16-bit precision data (acoustic signal of time domain information) is output from the register 24 to the residual calculation unit 8 as data output.

Here, since the multiplier 21 outputs data with 32-bit precision for each data input with 16-bit precision, no arithmetic error occurs. Then, when the data is stored in each memory, rounding processing with 16-bit precision is performed. However, in the adder 23, an integer range (upper extension bits) of a sufficient number of bits is prepared for overflow or underflow during the product-sum operation. Has been secured. Therefore, the rounding process is performed only when storing it in the memory or when obtaining the final output value, and therefore, no extra calculation error is accumulated.

The band division synthesis operation as described above is performed, and the acoustic signal (post-decoding signal) of 16-bit precision time domain information output from the band synthesis filter 6 of FIG. 1 is input to the delay unit 7. The digital acoustic signal (original signal) that has been generated is supplied to the residual calculation unit 8 and the calculation error that has occurred in the band division synthesis in the band division filter 1 and the band synthesis filter 6 is calculated to obtain a residual signal. Output (Step 1
07). The residual signal is treated as a block of 24 samples here. Then, the minimum number of bits required to express the maximum value in this block is set as the block bit width, and this is encoded with 3 bits. Table 2 shows the relationship between the residual signal and the block bit width. In Table 2, the number shown as the bit allocation is the block bit width, and the numerical value is determined by the maximum value M of the absolute value of the residual signal in the block.

Since the time axis of the decoded signal that has passed through the band division filter 1 and the band synthesis filter 6 is delayed by the delay peculiar to the filter, the original signal is passed through the delay unit 7 to the residual calculation unit 8. By supplying (Step 1
08), the time axis of the decoded signal is aligned.

The delay peculiar to this filter will be briefly described with reference to FIG. When a digital acoustic signal (original signal) of time domain information as shown in FIG. 3A is supplied to the band-splitting filter 1, the original signal is accumulated in the filter bank while shifting by 32 samples, and 32 subbands at any time. A sample is generated ((B) in the figure). Then, in the band synthesis filter 6, the sub-band samples are accumulated in the filter bank while shifting by 32 samples,
32 output signals (decoded signals converted to the time axis) are generated at any time ((C) in the same figure). At this time, the delay caused by a series of band division synthesis is 480 samples, and the decoded signal outputs the same data with a delay of 480 samples with respect to the original signal. Therefore, in the delay device 7, 48
The sample is delayed by 0 sample to absorb the delay peculiar to the filter ((D) in the figure), and the residual signal can be calculated in the residual calculation section 8.

Then, in the multiplexer 9, the sub-band data (digital acoustic signal of frequency domain information) with the reduced number of bits supplied from the quantizer 4 and the residual signal supplied from the residual calculator 8 are supplied. And for the frame sync word,
Various modes and auxiliary information, auxiliary information of frequency domain signals (frequency domain side information), and additional information of time domain signals (time domain side information) are added, for example, as shown in FIG.
A bit stream is generated by arranging as shown in (step 109). Since the residual signal is included in the same frame after being multiplexed in this way, the amount of delay required for decoding by correcting the signal that has been band-synthesized and converted into the time domain at the time of decoding is reduced. be able to.

Next, FIG. 2 shows a block diagram of an embodiment of the acoustic signal decoding apparatus of the present invention, and FIG. 7 shows a flow chart of its operation, showing an embodiment of the acoustic signal decoding method of the present invention. Also explained. In addition, the inverse quantizer 4 and the band synthesizing filter 6 shown in the acoustic signal encoding apparatus of FIG.
The dequantizer 4 and the band synthesizing filter 6 shown in FIG. 2 have completely the same configuration.

The bit stream coded by the audio signal coding apparatus is supplied to the demultiplexer 10 to decode the sync word, the mode, the auxiliary information, etc., and further, the frequency domain signal and the time domain signal. Separated (step 20)
1). The signals in the frequency domain after separation are supplied to an inverse quantizer 5 which is a pre-processing for band synthesis. The signal in the time domain is supplied to the residual correction unit 11 to correct the signal that has passed through the band synthesis filter 6 later.

As described above, the inverse quantizer 5 is exactly the same as the inverse quantizer 5 and the band synthesizing filter 6 used in the acoustic signal coding apparatus together with the band synthesizing filter 6, and its The operation is also the same. Also here, the same code as the sign bit is placed in the higher order (16-number of allocated bits).
Individually added and output (step 202). The signal output from the inverse quantizer 5 is supplied to the band synthesis filter 6, and the frequency domain signal is converted into the time domain signal (step 203). Since the process from the inverse quantizer 5 to the band synthesizing filter 6 is exactly the same as in the acoustic signal encoding device, the error with the original signal generated during the encoding process is the acoustic signal as a signal in the time domain. If corrected by the residual component supplied from the encoding device, the signal after band synthesis returns to the same signal as the original signal.

Therefore, in the residual correction unit 11, the time domain correction signal (residual signal) supplied from the demultiplexer 10 is added to the time domain band synthesized signal supplied from the band synthesis filter 6. Thus, the original signal is restored (step 204). Since the delay involved in the band synthesizing process in the band synthesizing filter 6 is adjusted on the acoustic signal encoding device side, the residual signal in the time domain is the residual error correcting unit in the state where the time axis coincides with the post-band synthesizing signal. 12 are supplied. That is, the residual signal is generated from the signal that has passed through the band division filter 1 and the band synthesis filter 6 and the original signal delayed in accordance with this signal on the acoustic signal encoding device side. Although delayed by 480 samples with respect to the signal, since the signal in the frequency domain does not pass through the band synthesizing filter 6, the signal is multiplexed with a small delay amount. Therefore, on the acoustic signal decoding device side, the delay amount after the signal in the frequency domain has passed through the band synthesis filter 6 is the same as the residual signal, and the time axes match.

As a result, the digital audio signal (original signal) input to the audio signal encoding apparatus of the present invention is transmitted or stored as a data-compressed signal as an encoded signal, and the audio signal decoding apparatus of the present invention is used. Thus, it is possible to output as a decoded signal that completely matches the original signal.

[0047]

The acoustic signal coding method and the acoustic signal coding device of the present invention can perform efficient compression coding by utilizing the characteristics of the acoustic signal by the frequency domain information. Further, since the acoustic signal decoding method and the acoustic signal decoding device of the present invention appropriately and minimally correct the frequency domain information by the residual signal which is the time domain information, the original signal is surely obtained. It has the effect that it can be decrypted.

[Brief description of drawings]

FIG. 1 is a configuration diagram showing an embodiment of an audio signal encoding device of the present invention.

FIG. 2 is a configuration diagram showing an embodiment of an audio signal decoding device of the present invention.

FIG. 3 is a configuration diagram showing an internal operation block configuration of a DSP used for a band synthesis filter.

FIG. 4 is a diagram for explaining a delay amount specific to a filter.

FIG. 5 is a configuration diagram illustrating a generation example of a bitstream.

FIG. 6 is a flowchart showing an operation example of the acoustic signal encoding device of the present invention.

FIG. 7 is a flowchart showing an operation example of the acoustic signal decoding device of the present invention.

[Explanation of symbols]

 1 band division filter 2 maximum value selection unit 3 bit allocation unit 4 quantizer 5 inverse quantizer 6 band synthesis filter 7 delay device 8 residual calculation unit 9 multiplexing unit (multiplexer) 10 demultiplexer 11 residual error Correction unit A Frequency domain information generation unit B Time domain information generation unit

Claims (5)

[Claims] 1. A first step of band-dividing and encoding a digital acoustic signal of time domain information supplied as an original signal to obtain a digital acoustic signal of frequency domain information, and encoding in the first step. A second step of generating, as time domain information, a residual signal between the digital sound signal obtained by band-synthesizing the digital sound signal of the frequency domain information and the digital sound signal of the time domain information supplied as the original signal; And a third step of multiplexing the residual signal of the time domain information encoded in the second step and the digital acoustic signal of the frequency domain information encoded in the first step. Characteristic acoustic signal encoding method. 2. An acoustic signal decoding method for decoding an acoustic signal encoded by the acoustic signal encoding method according to claim 1, wherein the encoded acoustic signal supplied is time-domain information. A first step of separating the residual signal and an acoustic signal of frequency domain information, and band-combining the acoustic signal of the frequency domain information separated in the first step to convert into an acoustic signal of time domain information And a third step of correcting the acoustic signal converted into the time domain information in the second step by the residual signal separated in the first step. A characteristic audio signal decoding method. 3. A band division filter for band-dividing a digital acoustic signal of time domain information supplied as an original signal, and an amplitude of the digital acoustic signal of frequency domain information in each band divided by this band division filter. A maximum value selection unit that detects the maximum value or energy value, a bit distribution unit that calculates the quantized bit distribution based on the maximum amplitude value or energy value supplied from this maximum value selection unit, and this bit A quantizer for encoding the digital audio signal of the frequency domain information supplied from the band division filter by the quantized bit allocation supplied from the allocation unit, and dequantizing the signal supplied from the quantizer. An inverse quantizer, a band synthesizing filter for band-combining the signals supplied from the inverse quantizer and re-converting into a signal of time domain information, and this band A residual difference calculation unit for generating a residual signal between the time domain information signal supplied from the synthesis filter and the time domain information digital acoustic signal supplied as the original signal; An acoustic signal coding apparatus comprising: a multiplexing unit that multiplexes a residual signal and a signal supplied from the quantizer. 4. A demultiplexer for demultiplexing a supplied multiplexed signal into frequency domain information and a residual signal, and time domain information by band combining the frequency domain information supplied from the demultiplexer. A sound comprising a band synthesizing filter for converting into a band, and a residual correction unit for correcting the information in the time domain supplied from the band synthesizing filter by the residual signal supplied from the demultiplexer. Signal decoding device. 5. The dequantizer and the band synthesizing filter which are the same as the dequantizer and the band synthesizing filter which compose the acoustic signal coding apparatus according to claim 3, and are constituted by the same dequantizer and band synthesizing filter. The acoustic signal decoding device described.