JPS6319919A - Voice coding system - Google Patents

Abstract

PURPOSE:To regenerate voices of high quality by adding the total value of drop-out bits to the first differential value of the next block and at the same time changing the value of the scale data to repeat again the coding processing if the code data has a carry. CONSTITUTION:In a DC-PCM coding system, the absolute value DD of the differential value DDS is limited 4 to a prescribed level. At the same time, the total CC of drop-out bits obtained when the immediately preceding block is through is added to the first differential value DC of the next block. Furthermore, the value of the scale data DK is increased by one scale in case the code data has a carry owing to a fact that '1' is added to the code data by the carry CC of the total value of drop-out bits. Under such conditions, the coding processing is repeated again with the corresponding block.

Description

[Detailed description of the invention] [Technical field] The present invention relates to a speech encoding method.

[Prior art] For example, when transmitting an audio signal using a high-speed digital line or digitally processing the audio signal such as volumeizing and synthesizing the audio signal for a voice response device, this audio signal is converted into a digital signal by some method. 6 Basically, audio signals have a frequency band of 0.3-3.4KH.
This is an analog signal of
n) encoding).

Then, to convert this digital signal back to the original audio signal,
It is sufficient to convert the signal into an analog signal using a digital/analog converter with a sampling frequency of 8 kHz and a resolution of 8 bits, and then pass it through a low-pass filter to shape the waveform. At this time, the resolution of the analog/digital converter and digital/analog converter (i.e. bit I of the PCM code
The larger I) is, the higher the quality of the reproduced audio is.

By the way, such a PCM encoded audio signal has 1
The bit rate per second (data rate; hereinafter referred to as bit rate) is 64 Kbps, and transmitting audio signals with such a high bit rate requires a very high-speed transmission path, and it is necessary to store such audio signals. This requires a huge amount of memory. Therefore, various proposals have been made to reduce the bit rate of audio signals.

One of them is a differential PCM encoding method that forms a difference between chronologically adjacent PCM codes. This differential PCM encoding method utilizes redundancy based on the correlation of audio waveforms, and changes in values between adjacent samples are often included in a limited dynamic range. , the number of bits per sample can be reduced. The adaptive differential PCM method (ADPCMf) recommended by the ccITT (International Telegraph and Telephone Consultative Committee), which is one of the adaptive differential PCM coding methods that further advances this differential PCM coding method, achieves a bit rate of 32 Kbps. There is.

In addition, APC-AB (Adaptive Predication), which utilizes the non-stationarity and linear predictability of audio signals,
tion Co-ding with Adaptiv
e Bit A11 location) method, or LSP (Line Spectrum) method using speech analysis and synthesis method.
umPair) method, etc.

However, such ADPCMf method, APC-
The AB method and the LSP method have disadvantages in that the encoding and decoding processes are very complicated, and the equipment for implementing them is very expensive.

On the other hand, one of the high-quality PCM audio transmission systems for broadcasting satellites is the quasi-instantaneous companding system. This quasi-instantaneous companding method is P
Divide the CM encoded audio data into blocks of a predetermined number in time series, identify scale data representing the most significant digit corresponding to the maximum value of the signal absolute value in each block, and include the most significant digit. This formats data of a predetermined number of bits into code data, and the encoding process is relatively simple, and the number of bits of one sample can be easily reduced. However, such quasi-instantaneous companding method is not efficient enough.

Therefore, a method to improve the efficiency of this quasi-instantaneous companding method is to ``combine the differential PCM method and quasi-instantaneous companding,'' but in general, simply applying quasi-instantaneous companding to the differential PCM method does not allow compression. Missing bits in time cause transmission errors, and the errors accumulate in the integrator on the receiving side, making reception impossible.

One way to solve this problem is to use differential companding PCM (D
C-PCM) J (Takahashi et al., Transactions of the Institute of Electronics and Communication Engineers '8
4/10 Vol. J67-B No. 10) has been proposed.

However, this method is effective when compressing differential data of about 15 bits to about 8 bits, and cannot be applied to a low bit rate encoding method that compresses differential data of 8 bits to about 3 bits.

[Purpose] The present invention was made in order to solve the above-mentioned disadvantages of the conventional technology, and its purpose is to provide an audio encoding method that can reproduce high-quality audio at a low bit rate and through simple processing. It is said that

[Configuration] In the DC-PCM encoding method, the present invention limits the absolute value of the difference value to a predetermined value, and also sets the cumulative value of missing bits at the end of the previous block to the first difference value of the next block. If the code data causes a carry as a result of adding 1 to the code data due to a carry of the accumulated value of missing bits, change the value of the scale data to one larger value. The above objective is achieved by repeatedly performing the encoding process on the block.

Embodiments of the present invention will be described in detail below with reference to the accompanying drawings.

First, the principle of the present invention will be explained.

For example, an audio signal as shown by the solid line in Fig. 1(a) is sampled at a sampling frequency of 8K)lz, and each sample is converted into 12-bit digital data (PCM).
Consider converting this digital data into a code (code) and then semi-instantaneous companding of this digital data. However, in order to prevent fluctuations in the scale value, the maximum value of this digital data is limited to 7 bits. The conditions for quasi-instantaneous companding are that the number of samples constituting one block is 8, each sample is compressed to 3 bits, and the scale value is 3 bits.

Note that the 7-bit digital data formed with a limited maximum value is digital data written in two's complement, and the most significant digit is a sign bit (sign bit for identifying positive/negative).

Here, it is assumed that the digital data of samples #1 to #8 are obtained as shown in FIG. 2(a).

In this block consisting of 8 samples, the data of sample #1 has the maximum absolute value, so the most significant digit position of the bit pattern, that is, bit 4
is set as the scale position POS for quasi-instantaneous companding.

And regarding each sample 11 to #8.

The lower three digits from bit 5, which is one digit higher than this scale position, are extracted as transmission bits, that is, code data.

Therefore, at the MSB (most significant digit) of code data, there is a sign bit (sign bit) that indicates whether the code data is positive or negative.

As a result, encoded data obtained by quasi-instantaneous companding is obtained, as shown in FIG. 2(b). That is, the data of scale position [PO5 is placed in the first 3 bits, followed by the 3-bit code data extracted from samples #1 to #18 in sequence.

At this time, in order to realize the above-mentioned l) C-PCM,
The data of the lower bits not included in the code data (hereinafter referred to as missing bits) are accumulated in sequence for each sample, and when a carry occurs in the accumulated value, the LSB of the code data is
Add 1 to . In this case.

Sample #3J6 with “-” mark. This addition occurs in #8.

In this way, code data reflecting the contents of the missing bits is formed.

When decoding this encoded data to reproduce an audio signal, first, the scale position pos is identified from the contents of the first three bits.

Next, the following data is sequentially divided into code data of 3 bits each, and the code data is divided into 7-bit data so that the MSB of the code data is located one digit higher than the previously identified scale position MPOS. Place.

The MS of the code data is in the upper digits than the code data.
By arranging the contents of B (that is, the contents of the sign bit) and placing 0 in the lower digits than the sign data, 6
The bit difference data is expanded (see (c) in the same figure).

Then, the audio signal is reproduced by integrating this difference data, converting the integrated value (12 bits) into digital/analog, and shaping the waveform with a low-pass filter.

The audio signal encoded and reproduced by C-PCH in this way is as shown by the dotted chain line in FIG. 1(b).
The signal changes at a position lower in level than the original audio signal shown by the solid line in the figure.

By the way, when the number of bits of the code data is as small as 3 bits, there is a possibility that a carry may occur in the code data as a result of adding 1 to the code data due to a carry of the accumulated value of missing bits.

Accordingly, in the present invention, when such a situation occurs, the value of the scale data is increased by one, and the encoding process of the block is performed again.

FIG. 3 shows a speech encoding device according to an embodiment of the present invention. The conditions for quasi-instantaneous companding in this audio encoding device are as described above.

In the figure, an input audio signal SS is band-limited by a low-pass filter 1, and then applied to an analog/digital converter 2 with a sampling frequency of 8 KHz, where it is converted into a 12-bit digital signal DS. is an adder/subtractor 3 for forming difference data.
is added to the positive input end of.

The difference data DDs output from this adder/subtractor 3 is as follows.

The maximum value limiting circuit 4 converts it into 7-bit differential data DD. The reason why the maximum value of differential data is limited in this way is as follows.

In other words, when the differential data DDs is compressed quasi-instantaneously as is, a correspondingly large scale value is set in a block that suddenly contains large differential data, and this makes it difficult for the data after quasi-instantaneous compression to follow other small differential data. Sexuality worsens. As a result, the restored audio signal gives an aurally jerky feel. Therefore, by limiting the maximum value of the difference data to a certain extent in this way, such auditory problems can be solved.

This difference data DD is applied to one input terminal of the adder 5 and one input terminal of the adder 6.

The output of adder 6 is added to register 7, and the output of register 7 is applied to the minus input terminal of adder/subtractor 3 and adder 6.
has been added to the low input end. In this way, the differential data D limited to 7 bits by the maximum value limiting circuit 4
An integrated value of D is formed by the adder 8, and this data is used as the data of the immediately preceding sample for forming the difference data DDs.

Further, the output of the adder 5 is applied to a buffer memory 8 having a storage capacity of 8 samples and a scale value setting section 9 for setting a scale value for quasi-instantaneous compression.

The data stored in the buffer memory 8 is applied to a quasi-instantaneous compression unit 10 that performs quasi-instantaneous compression encoding for each sample, and an accumulator 11 that accumulates missing bits not included in the encoded data.

The scale value setting unit 9 identifies the one with the largest absolute value among the eight consecutive samples of the difference data DD output from the adder 5, determines the most significant digit of the bit pattern, and determines the bit position. Output as 3-bit scale data DK.

This scale data DK is applied to one input terminal of a quasi-instantaneous compression section 10, an accumulator 11, and a multiplexer 12 for shaping one block's worth of data into a predetermined signal format.

For each sample of the 7-bit difference data DD added from the buffer memory 8, the quasi-instantaneous compression section 10 converts the scale data DK added from the scale value setting section 9 into one bit higher than the scale position. The 3-bit data designated as MSB is extracted and outputted to the adder 13 as compressed differential data DC.

The contents of the accumulator 11 are cleared at each start of a block, and the accumulator 11 identifies missing bits in the digital signal DS added from the buffer memory 6 based on the scale data DK added from the scale value setting section 9, and When the missing bits are accumulated and a carry occurs in the accumulated value, a carry signal CC is generated.
is output to the adder 13. Further, when finishing the processing in each block, the accumulated value at that time is outputted to the low input terminal of the adder 5. This makes it possible to eliminate the accumulation of errors due to missing bits, which is unique to quasi-instantaneous companding, when forming the first sample data of the 5th order block.

The adder 13 receives a carry signal C from the accumulator 11.
For samples to which C is input, quasi-instantaneous compression unit 1
1 is added to the compressed difference data DC added from 0 and the result is outputted to the multiplexer 12 as coded data DCo, and for other samples, the compressed difference data DC is directly outputted to the multiplexer 12 as coded data DCo.

Further, if a carry occurs as a result of adding 1 to the compressed difference data DC, the adder 13 outputs a carry signal CY to the scale value setting section 9 and the multiplexer 12.

As a result, the multiplexer 12 does not output one block of data (described later) that was input at that time, and the scale value setting section 9 changes the set scale value to one larger value. As a result, the quasi-instantaneous compression process is executed again on the data of the same block.

As shown in FIG. 4, the multiplexer 12 arranges the scale data D output from the scale value setting section 9 at the beginning, and then arranges the code data DCo of each sample.
A signal configured by sequentially arranging the data is formed as one block of encoded data DL and output to the next stage device (for example, a data transmission device or a data storage device, etc.).

In this way, the compressed difference data l)C output from the quasi-instantaneous compression section 10 is corrected and output as encoded data DL.

FIG. 5 shows an example of a voice decoding device according to an embodiment of the present invention. This audio decoding device decodes encoded data DL encoded by the audio encoding device described above and outputs an audio signal.

In the figure, encoded data DL output from a pre-stage device (when shown) such as a data receiving device or a data storage device is applied to a demultiplexer 21, and for each block, the first three bits are converted to a scale value. The other code data (compressed difference data) is identified as SC and added to the scale value input terminal of the quasi-instantaneous decompression unit 22.
is applied to the code data input terminal of the quasi-instantaneous decompression section 22.

The quasi-instantaneous decompression unit 22 divides the added code data into 3 bits each, and adds the 3 bits to the bit position corresponding to the input scale data SC in the 7-bit data.
The bit data is placed, and the sign bit contents are placed in the upper digits than the code data.

0 is placed in the lower digits to expand to 7-bit data, and this 7-bit data is output to an integrating section 23 consisting of an adder and a register.

The integrating section 23 integrates the sequentially input 7-bit data to form a 12-bit signal value for each sample of the audio signal, and outputs this to the digital/analog converter 24.

The digital-to-analog converter 24 converts the received signal value into 8
It is converted into a corresponding analog signal (level signal) at a conversion frequency of KHz, and outputted to the low-pass filter 25. This analog signal is waveform-shaped by a low-pass filter 25 and then output as a reproduced audio signal to a next-stage device (for example, an audio output device, etc.). The configuration of the audio decoding device is extremely simple.

Therefore, for example, this audio decoding device can be realized using a general-purpose 8-bit microprocessor.
Costs can be kept to a minimum.

Note that the various constants such as the number of bits in the embodiments described above are merely examples, and appropriate values can be set.

[Effect] As explained above, according to the present invention, in the DC-PCM encoding method, the absolute value of the difference value is limited to a predetermined value, and the accumulated value of missing bits at the end of the previous block is If the code data causes a carry as a result of adding 1 to the code data due to the carry of the accumulated value of missing bits, the value of the scale data is increased by 1. Since the encoding process is repeatedly performed on the block after changing it to a larger size, it is possible to obtain the excellent effect of reproducing high-quality audio with a low bit rate and simple processing.

[Brief explanation of the drawing]

Figures 1 (a) and (b) are detailed explanatory diagrams of the present invention, Figures 2 (a) to (c) are signal arrangement diagrams for detailed explanation of the present invention, and Figure 3 is a detailed diagram of the present invention. FIG. 4 is a block diagram showing an audio encoding device according to an embodiment, FIG. 4 is a signal arrangement diagram showing an example of encoded data, and FIG. 5 is a block diagram showing an audio decoding device according to the present invention. . 1.25...Low pass filter, 2...Analog/
Digital converter, 3... Adder/subtractor, 4... Maximum value limit circuit, 5.6.13... Adder, 7... Register, 8... Buffer memory, 9... Scale value Setting section, 10... Quasi-instantaneous compression section, 11... Accumulator, 12... Multiplexer, 21... Demultiplexer, 22... Quasi-instantaneous expansion section, 23... Integrating section, 2
4...Digital/analog converter.゛, -1,・' Figure 1 (a) (b) $+1 o2 g3 a4 ub +s5
tt'I n6Figure 2Figure 4L Figure 5

Claims (1)

[Claims] Forms a difference value between adjacent samples of PCM encoded audio data, divides the difference value into a predetermined number of blocks in time series, and corresponds to the maximum absolute value of the difference value in each block. identify the scale data representing the most significant digit, format the data of a predetermined number of bits including the most significant digit into code data, compress the audio data, and accumulate missing bits that were not included in the code data. In an audio encoding method that corrects the coded data by adding 1 to the least significant digit of the coded data of the sample when a value carries over, the absolute value of the difference value is limited to a predetermined value, and the value of the previous value is limited to a predetermined value. If the cumulative value of the missing bits at the end of the block is added to the first difference value of the next block and the above correction is made in this block, and a carry occurs in the code data, the scale data is increased by one. A speech encoding method characterized in that the encoding process is performed again on the block in a state in which the block is encoded.