JP3191257B2 - Acoustic signal encoding method, acoustic signal decoding method, acoustic signal encoding device, acoustic signal decoding device - Google Patents

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

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

[0002]

2. Description of the Related Art In general, an audio signal simply converted into a digital signal has some continuity or redundancy unless it is random data such as white noise. Less compression method) is known. Observing the acoustic signal makes it easy to understand that adjacent samples on the time axis have correlation (continuity). In addition, the amplitude distribution of an acoustic signal generally has a bias.
There is regularity in the occurrence probability of each bit being “0” or “1”. When this bias increases, the redundancy of the audio signal also increases.

[0003] As an example of the reversible compression method, a differential PCM method is well known as a compression method utilizing 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 in which a target sample value is predicted from preceding and succeeding sample values using the linearity, an adaptive prediction difference PCM method in which a difference width is adaptively determined, and the like are also known.

Further, as a method of 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 There is a vector coding method of performing coding by matching with the signal of the table described above. In addition, there is a method of performing frequency conversion of an audio signal in the time domain and adaptively allocating the information amount for each band based on the bias of the spectrum distribution, thereby reducing the code amount and performing encoding.

[0005] However, the mainstream of the lossless compression method is the differential PCM method, and a compression method combining this and the entropy coding method, for example, Huffman coding, is used. An encoding method using frequency conversion is mainly used for an irreversible high-efficiency encoding method that utilizes psychoacoustic characteristics and the like.

[0006]

In such a conventional audio encoding method, when encoding is performed only in the time domain, encoding using the characteristic caused by the deviation of the frequency distribution of the audio signal cannot be performed. Efficient compression could not be achieved. Particularly, in the differential PCM system, when a signal having a considerably large amplitude exists in a high frequency band, a 17-bit differential bit is required to have a dynamic range equivalent to the 16-bit linear PCM system in order to express a difference value from the previous sample. Needs width. Therefore, there is a problem in that encoding an audio signal whose amplitude in a high frequency band is largely changed greatly deteriorates encoding efficiency.

[0007] Also, even in the case of encoding in which adaptive prediction is introduced, encoding a music source with poor prediction accuracy (difficult to predict) increases the differential bit width and does not always increase the encoding efficiency. . In addition, the mathematical compression method does not fully exploit the continuity and the redundancy caused by the correlation of speech, and the encoding quality of mathematical compression is lower than that of the encoding method using the characteristics of acoustic signals. It was inferior (the code amount was large).

[0008] Furthermore, an encoding method using frequency transformation (particularly, orthogonal transformation) is theoretically reversible, but high quantization precision (calculation precision) is required to maintain reversibility. Therefore, the quantization width of the sample of the frequency domain information is increased, and as a result, when the reversibility is to be maintained, the configuration becomes large-scale and the compression ratio is not improved.

Accordingly, the present invention provides an audio system that retains efficient and complete reversibility by using time domain correction for a frequency transform coding system that has been conventionally considered unsuitable as a reversible coding system. An object of the present invention is to provide a signal encoding method and a signal decoding method.

[0010]

As means for achieving the above object, in the audio signal encoding apparatus of the present invention, in order to utilize the deviation of the frequency distribution possessed by the audio signal, the frequency domain information generation unit uses a digital signal. After dividing the sound signal into bands, based on the energy values obtained for the individual bands,
The sample value quantized by the bit allocation according to the information amount is encoded as frequency domain information, and the time domain information generation unit inversely quantizes the quantized sample value supplied from the frequency domain information generation unit, By encoding the residual signal between the signal after band synthesis and the original signal as time-domain information in order to re-convert this into a time-domain signal, and multiplexing both in a multiplexing unit, Lossless encoding is performed.

Further, in the audio signal decoding apparatus according to the present invention,
The supplied multiplexed signal is separated into frequency domain information and a residual signal, and the frequency domain information is band-synthesized and then corrected by the residual signal so that the same digital sound signal as the original signal is obtained. It was done.

Here, since the decoded signal completely matches the original signal, the band synthesizing unit in the encoding device and the band synthesizing unit in the decoding device have the same configuration, and the operation accuracy and the rounding process are the same. Taking a technique. As a result, a difference value between the band-combined signal and the original signal in the encoding device is obtained, and the difference value is encoded and transmitted as time-domain information. The signal added and decoded is exactly the same as the original signal.

A lossless compression encoding method for multiplexing and encoding information in both the frequency domain and the time domain of an audio signal and a decoding method for the audio signal are described in US Pat. Redundancy can be effectively reduced irrespective of the type of signal (audio sources of various genres), and the residual signal, which is a time-domain signal, is used for correction to ensure the original signal It is possible to restore. This method provides a complete lossless compression method without requiring a procedure for strictly calculating the quantization precision of the frequency domain signal required to maintain reversibility, compared to the case of encoding in the frequency domain alone. be able to.

[0014]

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS An embodiment of an audio signal encoding method, an audio signal decoding method, an audio signal encoding apparatus, and an audio signal decoding apparatus according to the present invention will be described with reference to the drawings. FIG. 1 is a configuration 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 audio signal encoding method will be described at the same time with reference to the drawings.

The audio signal encoding apparatus shown in FIG. 1 comprises a frequency domain information generator A, a time domain information generator B, and a multiplexer (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. The time domain information generation unit B includes an inverse quantizer 5, a band It comprises a synthesis filter 6, a delay unit 7, and a residual calculation unit 8. The input digital audio signal is formed of blocks each having a frame as a processing unit. In the present embodiment, the audio signal forming one frame is 512 samples per channel, and the band division width is 32 bands.

The precision of the least significant bit is fixed at an arbitrary bit width. Here, a fixed quantization precision of 16 bits is used, and information below this is rounded off to an upper bit (the 16th bit which is the least significant bit). reflect. It is assumed that the bit allocation information in the frequency domain represents codes 0 to 16 bits by 4 bits, and the bit allocation in the time domain represents codes 0 to 8 bits by 3 bits.

Here, the bit distribution width in the time domain depends on the characteristics of the filter, and depends on the number of multiplications during the product-sum operation for the band synthesis. If the worst value of error accumulation in this operation is within the time distribution bit width, the bit distribution width in the frequency domain guarantees the final quantization accuracy. In the present embodiment, encoding is performed with 3 bits, so 16-bit quantization accuracy is guaranteed. In addition, if the bit allocation width is expressed by 4 bits, lossless compression of an input audio signal having a quantization accuracy of 24 bits becomes possible. However, in this case, since the time domain information increases and the coding quality deteriorates, efficient information amount distribution with the frequency domain information is required.

Table 1 shows a 4-bit frequency domain bit allocation table, and Table 2 shows a 3-bit time domain bit allocation table. Note that code 0 indicates that there is no distribution, and if there is distribution, it is set between code 2 and the maximum code value including the sign bit.

[0019]

[Table 1]

[0020]

[Table 2]

Next, the operation of the audio signal encoding 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 unit 7 (described later). In this band division filter 1,
The input signal which is time domain information is developed into frequency domain information (step 101). Here, the sub-band filter is equally divided into 32 bands, and the bandwidth W is set as in the following equation. Note that the accuracy of the output subband data is 1 which is the least significant bit accuracy as described above.
It shall be 6 bits.

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

The band dividing filter 1 and a band synthesizing filter 6 to 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, and a signal for decomposing a signal into a base waveform. A wavelet transform to be expressed, and a Fourier transform, which is a typical method of frequency transform, and the like can be given. And
In the present invention, since the information component in the time domain is also used, it is not necessary for the signal after band division and synthesis to return to a complete original signal, and any frequency conversion method may be used. In this embodiment, a 512-tap sub-band filter (polyphase filter) will be used in order to make the description concrete. Note that a delay of 480 samples occurs through band division synthesis.

The audio signal of the frequency domain information divided into 32 equal bands by the band dividing filter 1 is supplied to the maximum value selector 2 and the quantizer 4. In the maximum value selection unit 2, 16 (5) for each of the 32 bands existing in one frame.
The absolute value of the subband data (amplitude value) or energy value is compared, and the maximum value S is selected and output (step 102).

The maximum value S of the sub-band 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 frequency domain bit allocation table shown in Table 1 to find the maximum value S of each band (band),
The number of bits allocated to each of the two bands is determined (step 103). As shown in Table 1, the bit allocation is performed according to the minimum necessary number of bits in order to represent the maximum value sample in two's complement notation.

In the quantizer 4, based on the bit allocation information supplied from the bit allocation section 3, the band division filter 1
The number of bits of each sub-band data of the audio signal of the frequency domain information supplied from the terminal is reduced (step 104). The reduction here is a reduction of (16-the number of allocated bits) by removing the upper bits that are the same as the code bits, excluding the code bits. Table 3 shows an example of the reduction in the number of bits of the subband data.

[0027]

[Table 3]

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

Accordingly, in the inverse quantization, it is sufficient to fill the reduced upper bits by the designated number of sign bits (sign bits) from the bit allocation information. The sub-band data and bit allocation information quantized in this way are output to a multiplexer (multiplexing unit) 9 and inversely quantized in order to re-transform the sub-band data into the time domain. Output to the container 5.

The inverse quantization in the inverse quantizer 5 is performed by adding (16-number of allocated bits) the same code as the sign bit to the upper bits (step 10).
5). Therefore, in such quantization and dequantization, there is no restriction on the operation during the operation, for example, no rounding or the like, so that no quantization error occurs.

The sub-band data adjusted to the least significant bit precision (16 bits in this embodiment) by the inverse quantization by the inverse quantizer 5 is supplied to the band synthesis 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 to completely match the calculation accuracy, the filter coefficient accuracy, the operation process, and the rounding process for the data in the output stage with the band synthesizing filter 6 of the decoding device described later.

In general, digital audio signal processing uses D
SP (Digital Signal Processor) is often used.
Therefore, the band synthesis processing in the band synthesis filter 6 will be described using a 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 arithmetic precision and the data width inside the memory are 16 bits.

[0033] The band division and synthesis by a sub-band filter or the like is performed mainly 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 connected to a memory for various coefficients and a memory for intermediate data (operation memory) and an input / output bus required in the operation process. Connected.

In the figure, input data (sub-band data) and filter coefficient data corresponding to the input data are sequentially input to a multiplier 21 via a 16-bit input data bus. In this input data, data output from the inverse quantizer 5 is stored in a memory (not shown) with 16-bit precision, and necessary data is stored in the multiplier 21.
Is supplied from time to time. The filter coefficient data is
Similarly, it is stored in a memory for various coefficients (not shown) with 16-bit precision in advance. The multiplier 21 multiplies the input data by the filter coefficient data and outputs the result as 32-bit data.

The output of the multiplier 21 is 32 bits + α
Are supplied to the adder 23. This α indicates the upper extension bit. The product-sum operation uses a 32-bit bus, and the other uses a 16-bit bus. Then, the output from the adder 23 is temporarily stored in the register 22, and the output of the register 22 and the output from the next multiplier 21 are added by the adder 23, and are added to the register 22 for the next operation. The state of accumulation is repeated until there is no more input data.

When the first-order product-sum operation is completed, the output from the adder 23 is supplied to the register 24 as it is with 32-bit precision data, rounded to 16-bit precision data, and output to a calculation memory (not shown). I do. This arithmetic memory stores 16-bit precision data rounded from 32 bits and outputs the data to the multiplier 21 in order to perform a quadratic product-sum operation. And in the same way,
When the second-order product-sum operation is performed, 16-bit data (sound 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 of 32-bit precision for each data input of 16-bit precision, no arithmetic error occurs. When the data is stored in each memory, rounding processing with 16-bit precision is performed. However, in the adder 23, an integer area (upper-order extended bits) having a sufficient number of bits in preparation for overflow or underflow during the product-sum operation Is secured. Therefore, since the rounding process is performed only when the data is stored in the memory or when the final output value is obtained, an extra calculation error does not accumulate.

The above-described operation of band division synthesis is performed, and an acoustic signal (decoded signal) of 16-bit precision time domain information output from the band synthesis filter 6 of FIG. The digital audio signal (original signal) is supplied to the residual calculation unit 8 to calculate an arithmetic error generated in the band division synthesis by the band division filter 1 and the band synthesis filter 6 and to generate a residual signal. Output (Step 1
07). Here, the residual signal is treated as a block of 24 samples. Then, the minimum number of bits required to represent the maximum value in this block is defined as a block bit width, and this is coded with 3 bits. Table 2 shows the relationship between the residual signal and the block bit width. In Table 2, the number indicated 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 due to a delay inherent in the filter, the original signal is sent to the residual calculation unit 8 via the delay unit 7. By supplying (step 1
08), the time axis with the decoded signal is aligned.

The delay inherent in the filter will be briefly described with reference to FIG. When a digital audio signal (original signal) of time domain information as shown in FIG. 2A is supplied to the band division filter 1, the original signal is accumulated in the filter bank while shifting by 32 samples, and 32 subbands are occasionally obtained. A sample is generated (FIG. 2B). Then, in the band synthesis filter 6, the sub-band samples are accumulated in the filter bank while shifting by 32 samples.
At any time, 32 output signals (decoded signals converted into a time axis) are generated (FIG. 3C). At this time, a delay caused by a series of band division synthesis is 480 samples, and the decoded data outputs the same data with a delay of 480 samples from the original signal. Therefore, 48
The delay inherent in the filter is absorbed by delaying by 0 samples (FIG. 3D), and the residual signal calculation unit 8 can calculate the residual signal.

Then, in the multiplexer 9, the sub-band data (digital audio signal of frequency domain information) supplied from the quantizer 4 and having the reduced number of bits, and the residual signal supplied from the residual calculator 8. And the frame sync word,
Various modes, auxiliary information, auxiliary information (frequency-domain side information) for frequency-domain signals, and auxiliary information (time-domain side information) for time-domain signals are added.
(Step 109). Since the residual signal is included in the same frame after being multiplexed in this way, at the time of decoding, the amount of delay required when decoding by correcting the signal that has been subjected to the band synthesis processing and converted into the time domain is reduced. be able to.

Next, FIG. 2 is a block diagram showing one embodiment of an audio signal decoding apparatus according to the present invention, and FIG. 7 is a flowchart showing the operation thereof, showing one embodiment of an audio signal decoding method according to the present invention. Also described. Note that the inverse quantizer 4 and the band synthesis filter 6 shown in the audio signal encoding device of FIG.
And the inverse quantizer 4 and the band synthesis filter 6 shown in FIG.

The bit stream encoded by the audio signal encoding device is supplied to a demultiplexer 10, which decodes a synchronization word, a mode, auxiliary information, and the like, and further converts the signal into a signal in the frequency domain and a signal in the time domain. Separated (step 20
1). The frequency domain signal after the separation is supplied to an inverse quantizer 5 which is a pre-process for band synthesis. Further, the signal in the time domain is supplied to the residual correction unit 11 in order to correct the signal having passed through the band synthesis filter 6 later.

As described above, the inverse quantizer 5 is completely the same as the inverse quantizer 5 and the band synthesis filter 6 used in the audio signal encoding device together with the band synthesis filter 6. The operation is the same. And here, the same code as the sign bit is placed in the higher order (16-number of allocated bits).
Are added and output (step 202). The signal output from the inverse quantizer 5 is supplied to the band synthesis filter 6, and the signal in the frequency domain is converted into a signal in the time domain (step 203). Since the processing steps from the inverse quantizer 5 to the band synthesis filter 6 are exactly the same as those of the audio signal encoding apparatus, an error between the original signal generated during the encoding processing and the audio signal If the signal is corrected by the residual component supplied from the encoding device, the signal after the band synthesis returns to the same signal as the original signal.

Therefore, the residual correction unit 11 adds the time domain correction signal (residual signal) supplied from the demultiplexer 10 to the time domain band-combined signal supplied from the band combining filter 6. Thus, the original signal is restored (step 204). Since the delay in the band synthesis processing in the band synthesis filter 6 is adjusted on the acoustic signal encoding device side, the residual signal in the time domain is processed by the residual correction unit in a state where the time axis coincides with the post-band synthesis signal. 12 are provided. In other words, the residual signal is generated on the acoustic signal encoding device side from the signal that has passed through the band division filter 1 and the band synthesis filter 6 and the original signal delayed according to this signal. Although the signal is delayed by 480 samples, since the signal in the frequency domain is not passed through the band synthesis filter 6, the signal is multiplexed with a small delay amount. Therefore, on the acoustic signal decoding device side, the delay amount after the frequency domain signal has passed through the band synthesis filter 6 becomes the same as the residual signal, and the time axis matches.

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

[0047]

The audio signal encoding method and the audio signal encoding apparatus of the present invention can perform efficient compression encoding utilizing the characteristics of an audio signal by using frequency domain information. Further, the audio signal decoding method and the audio signal decoding apparatus of the present invention perform appropriate and minimum necessary correction of the frequency domain information by the residual signal which is the time domain information. There is an effect that decoding can be performed.

[Brief description of the 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 illustrating an embodiment of an audio signal decoding device according to 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 bit stream.

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

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

[Explanation of symbols]

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

Claims (4)

(57) [Claims] A digital audio signal of frequency domain information obtained by dividing a band of a digital audio signal of time domain information supplied as an original signal.
Digital audio signal of the frequency domain information
The amplitude value or
An energy value is obtained, and the obtained amplitude value or energy is obtained.
Calculates the quantization bit distribution based on the
Digital sound of the frequency domain information according to bit allocation
A first step of encoding a signal, and a digital audio signal of the frequency domain information encoded in the first step is inversely quantized and then subjected to band synthesis to perform time domain synthesis.
Reconverted to information signal and time obtained by this reconversion
And the signal of the region information, the second step and the time domain generated in the second step of generating a residual signal as a time domain information and the digital audio signal in the time region information supplied as the original signal A third step of multiplexing the information residual signal and the digital audio signal of the frequency domain information encoded in the first step. 2. An audio signal decoding method for decoding an audio signal encoded by the audio signal encoding method according to claim 1, wherein the supplied encoded audio signal is processed in a time domain. information
Residual signal and the first step of the digital audio signal and the binary <br/> away the frequency domain information, digital frequency domain information separated by the first step
After dequantizing the audio signal, the band is synthesized and the time domain
A second step and, a digital transformed into the time domain information in the second step of converting the digital audio signal information
The acoustic signal is converted to the time domain information separated in the first step.
And a third step of correcting with a residual signal of the report . 3. A digital audio signal of time domain information supplied as an original signal is band-divided into digital signals of frequency domain information.
A band dividing filter to obtain an acoustic signal, and the maximum value selecting section in individual bands that are divided in the band dividing filter, for detecting a maximum value of the amplitude value or the energy value of the digital acoustic signal of the frequency region information, the A bit allocation unit for calculating a quantization bit allocation based on the maximum value of the amplitude value or the energy value supplied from the maximum value selection unit; and a quantization bit allocation supplied from the bit allocation unit. A quantizer for encoding a digital audio signal of the frequency domain information to be obtained, an inverse quantizer for inversely quantizing a signal supplied from the quantizer, and a signal supplied from the inverse quantizer A band synthesizing filter for synthesizing a band and reconverting it into a signal of time domain information; a signal of time domain information supplied from the band synthesizing filter and the original signal; A residual calculator for generating a residual signal between the supplied time domain information and the digital audio signal; multiplexing the residual signal supplied from the residual calculator and the signal supplied from the quantizer And a multiplexing unit for converting. 4. An audio signal encoding apparatus according to claim 3, wherein
Audio signal decoding for decoding the encoded and multiplexed audio signal
An encoding device, wherein the supplied encoded and multiplexed audio signal is
Digital acoustic signal of wave number domain information and residual of time domain information
A separator for separating the signal into signals, and digital sound in the frequency domain supplied from the separator.
An inverse quantizer that inversely quantizes the signal, and a band synthesis of the signal supplied from the inverse quantizer to time
A band combining filter that converts the digital audio signal of the region information, data of the time region information supplied from the band combining filter
The time the digital acoustic signal supplied from the demultiplexing unit
An audio signal decoding device, comprising: a residual correction unit configured to correct the residual information using a residual signal of region information .