CN113450807B - Method and device for compressing voice digital signal - Google Patents
Method and device for compressing voice digital signal Download PDFInfo
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Abstract
The invention relates to the technical field of communication, and discloses a method and a device for compressing a voice digital signal, which are used for solving the problem that a lookup table occupies a large amount of storage space. The method comprises the following steps: firstly, obtaining a first voice digital signal of an unsigned bit binary number based on an original voice digital signal; based on normalization rules, determining an index value of the first voice digital signal, calling a second-order fitting curve matched with the index value, and compressing the first voice digital signal to obtain a second voice digital signal; and finally, sequentially carrying out sign bit resetting and redundancy value truncation processing on the second voice digital signal to obtain and output a third voice digital signal. The first voice digital signal is compressed by calling a second-order fitting curve, and compared with the traditional table look-up method, a large amount of storage space can be saved.
Description
Technical Field
The present invention relates to the field of communications technologies, and in particular, to a method and an apparatus for compressing a speech digital signal.
Background
The distributed base station performs analog-to-digital conversion on a voice analog signal sent by the intelligent terminal device to obtain a corresponding voice digital signal, and generally performs companding processing on the voice digital signal in a non-uniform quantization mode in order to save a storage space occupied by storing the voice digital signal and reduce a bandwidth occupied by transmitting the voice digital signal.
Specifically, in a look-up table stored in advance, a voice compression signal matching the voice digital signal is looked up and output. The lookup table stores the mapping relation between each voice digital signal and the voice compressed signal obtained based on the A-law compression algorithm.
As the computing power of distributed base stations increases gradually, the use of table lookup to compress voice digital signals occupies a large amount of storage space.
In view of the above, a new method for compressing a speech digital signal needs to be devised to overcome the above-mentioned drawbacks.
Disclosure of Invention
The embodiment of the invention provides a method and a device for compressing a voice digital signal, which are used for solving the problem that a lookup table occupies a large amount of storage space.
The embodiment of the invention provides the following specific technical scheme:
in a first aspect, the present invention provides a method for compressing a speech digital signal, including:
obtaining a first voice digital signal based on an original voice digital signal, wherein the first voice digital signal is a binary number without a sign bit;
determining an index value of the first voice digital signal based on a preset normalization rule, calling a second-order fitting curve matched with the index value, and compressing the first voice digital signal to obtain a second voice digital signal;
and sequentially carrying out redundancy truncation and sign bit resetting on the second voice digital signal to obtain and output a third voice digital signal.
Optionally, obtaining the first speech digital signal based on the original speech digital signal includes:
taking absolute values of numerical digits of the original voice digital signals to obtain first voice digital signals; or,
when the original voice digital signal is determined to be a positive number, determining the numerical value bit of the original voice digital signal as the first voice digital signal;
and when the original voice digital signal is determined to be a negative number, taking an absolute value of the numerical value of the original voice digital signal to obtain the first voice digital signal.
Optionally, taking an absolute value of the numerical bits of the original digital signal includes:
acquiring a sign bit of the original voice digital signal;
if the sign bit represents that the original voice digital signal is a positive number, performing original code operation on a numerical value bit in the original voice digital signal;
and if the sign bit represents that the original voice digital signal is a negative number, performing complement operation on a numerical value bit in the original voice digital signal.
Optionally, determining the index value of the first speech digital signal based on a preset normalization rule includes:
determining the position of a set index in the first voice digital signal according to a set sequence;
based on the location, an exponent value of the first speech digital signal is determined.
Optionally, performing redundancy truncation processing on the second speech digital signal, including:
and according to a set sequence, when the n continuous numerical digits in the second voice digital signal are determined to be invalid numerical digits, carrying out truncation processing on the n continuous numerical digits.
Optionally, after performing redundancy truncation processing on the second voice digital signal, further performing sign bit resetting processing on the second voice digital signal, where the processing includes:
and adding the sign bit of the original voice digital signal to the second voice digital signal.
In a second aspect, an embodiment of the present invention further provides an apparatus for compressing a speech digital signal, including:
the conversion unit is used for obtaining a first voice digital signal based on an original voice digital signal, wherein the first voice digital signal is a binary number without sign bits;
the compression unit is used for determining an index value of the first voice digital signal based on a preset normalization rule, calling a second-order fitting curve matched with the index value, and compressing the first voice digital signal to obtain a second voice digital signal;
and the processing unit is used for sequentially carrying out redundancy value truncation and sign bit resetting on the second voice digital signal to obtain and output a third voice digital signal.
Optionally, the first speech digital signal is obtained based on an original speech digital signal, and the conversion unit is configured to:
taking an absolute value of the numerical digit of the original voice digital signal to obtain the first voice digital signal; or,
when the original voice digital signal is determined to be a positive number, determining the numerical digit of the original voice digital signal as the first voice digital signal;
and when the original voice digital signal is determined to be a negative number, taking an absolute value of the numerical value of the original voice digital signal to obtain the first voice digital signal.
Optionally, the conversion unit is configured to, for each digital bit of the original digital signal, obtain an absolute value:
acquiring a sign bit of the original voice digital signal;
if the sign bit represents that the original voice digital signal is a positive number, performing original code operation on a numerical value bit in the original voice digital signal;
and if the sign bit represents that the original voice digital signal is a negative number, performing complement operation on a numerical value bit in the original voice digital signal.
Optionally, based on a preset normalization rule, determining an index value of the first speech digital signal, where the compression unit is configured to:
determining the position of a set index in the first voice digital signal according to a set sequence;
based on the location, an exponent value of the first speech digital signal is determined.
Optionally, the processing unit is configured to perform redundancy truncation processing on the second speech digital signal, and is configured to:
and according to a set sequence, when the n continuous numerical digits in the second voice digital signal are determined to be invalid numerical digits, carrying out truncation processing on the n continuous numerical digits.
Optionally, after performing redundancy truncation processing on the second voice digital signal, further performing sign bit resetting processing on the second voice digital signal, where the processing unit is configured to:
and adding the sign bit of the original voice digital signal to the second voice digital signal.
In a third aspect, an embodiment of the present invention further provides a computing device, including:
a memory for storing program instructions;
and the processor is used for calling the program instructions stored in the memory and executing any one of the compression methods of the voice digital signals according to the obtained program.
In a fourth aspect, an embodiment of the present invention further provides a storage medium, which includes computer readable instructions, and when the computer readable instructions are read and executed by a computer, the computer is caused to execute any one of the above methods for compressing a speech digital signal.
The invention has the following beneficial effects:
in the embodiment of the invention, a first voice digital signal of binary number without sign bit is obtained based on an original voice digital signal; based on normalization rules, determining an index value of the first voice digital signal, calling a second-order fitting curve matched with the index value, and compressing the first voice digital signal to obtain a second voice digital signal; and finally, sequentially carrying out sign bit resetting and redundancy value truncation processing on the second voice digital signal to obtain and output a third voice digital signal. The first voice digital signal is compressed by calling a second-order fitting curve, and compared with the traditional table look-up method, a large amount of storage space can be saved.
Drawings
FIG. 1 is a flow chart of compressing a speech digital signal according to an embodiment of the present invention;
FIG. 2 is a schematic flow chart of compressing an original speech digital signal 2 according to an embodiment of the present invention;
fig. 3 is a schematic structural diagram of a device for compressing a speech digital signal according to an embodiment of the present invention;
fig. 4 is a schematic structural diagram of a computing device according to an embodiment of the present invention.
Detailed Description
In order to solve the problem that the lookup table occupies a large amount of storage space, a new technical scheme is provided in the embodiment of the application. The scheme comprises the following steps: firstly, obtaining a first voice digital signal of an unsigned bit binary number based on an original voice digital signal; based on normalization rules, determining an index value of the first voice digital signal, calling a second-order fitting curve matched with the index value, and compressing the first voice digital signal to obtain a second voice digital signal; and finally, sequentially carrying out sign bit resetting and redundancy value truncation processing on the second voice digital signal to obtain and output a third voice digital signal.
A user may communicate with other users via a telephone network or a network communication system. Specifically, a distributed base station in a telephone network or a network communication system needs to perform analog-to-digital conversion on a voice analog signal sent by a user to obtain a corresponding voice digital signal, and in order to save a storage space occupied by storing the voice digital signal and reduce a bandwidth occupied by transmitting the voice digital signal, a non-uniform quantization mode is adopted, that is, a corresponding second-order fitting curve is called to perform companding processing on the voice digital signal.
Preferred embodiments of the present invention will be described in detail below with reference to the accompanying drawings.
Referring to fig. 1, in the embodiment of the present invention, the process of compressing the speech digital signal is as follows:
s101: and obtaining a first voice digital signal based on the original voice digital signal, wherein the first voice digital signal is a binary number without sign bits.
The distributed base station performs analog-to-digital conversion processing on a voice analog signal sent by a user to obtain an original voice digital signal, wherein the original voice digital signal is a binary number with sign bits, the leftmost bit of the digital signal is usually determined as the sign bit, and the rest bits are determined as numerical value bits, wherein the second bit on the left side is the highest numerical value bit, and the rightmost bit is the lowest numerical value bit.
Optionally, the first speech digital signal may be obtained by the following two ways:
the first method is as follows: and taking an absolute value of the numerical digit of the original voice digital signal to obtain a first voice digital signal.
Specifically, a sign bit of an original voice digital signal is obtained, and if the sign bit represents that the original voice digital signal is a positive number, an original code operation is performed on a numerical value bit in the original voice digital signal; and if the sign bit represents that the original voice digital signal is a negative number, performing complement operation on the numerical value bit in the original voice digital signal.
For example, the original speech digital signal 1 is 00001010111101010, the sign bit of the original speech digital signal 1 is 0, the original speech digital signal 1 is determined to be a positive number, and the original code operation is performed on 0001010111101010, so that the first speech digital signal 1 is 0001010111101010.
For another example, the original speech digital signal 2 is 10001010011011101, the sign bit is 1, the original speech digital signal 2 is determined to be a negative number, and the complementary operation is performed on 0001010011011101, so that the first speech digital signal 2 is 1110101100100011.
The second method comprises the following steps: when the original voice digital signal is determined to be a positive number, determining the numerical digit of the original voice digital signal as a first voice digital signal; and when the original voice digital signal is determined to be a negative number, taking an absolute value of the numerical digit of the original voice digital signal to obtain a first voice digital signal.
For example, the original voice digital signal 3 is 00001110111001010, the sign bit is 0, the original voice digital signal 3 is determined to be a positive number, and the numeric bit of the original voice digital signal 3 is directly determined to be the first voice digital signal, so the first voice digital signal 3 is 0001110111001010.
For another example, the original speech digital signal 4 is 10101010111001101, the sign bit is 1, the original speech digital signal 4 is determined to be a negative number, and the complementary operation is performed on 0101010111001101, so that the first speech digital signal 4 is 1010101000110011.
No matter the first or second method is adopted, the absolute value of the original voice digital signal of the negative number needs to be obtained, so that the quantization error brought by directly compressing the negative value can be reduced.
S102: and determining an index value of the first voice digital signal based on a preset normalization rule, calling a second-order fitting curve matched with the index value, and compressing the first voice digital signal to obtain a second voice digital signal.
Before step 102 is executed, a second-order fitting curve of each index value needs to be calculated respectively based on an a-law companding formula.
The formula for A-law companding is as follows:
wherein A is a compression parameter, x is a normalized first speech digital signal, y is a companded second speech digital signal, and sgn (x) is a sign function of the parameter x. After the value of the compression parameter A is set and the first voice digital signal x is obtained, a second-order fitting curve corresponding to each index value can be calculated according to a formula (1) so as to compress the voice digital signals divided into different index values in a segmented approximate compression mode, so that the compression efficiency is improved, and the quantization error caused by compression can be reduced.
Alternatively, the process of determining the exponent value of the first speech digital signal is described as follows:
a1, determining the position of a set index in the first voice digital signal according to a set sequence.
And A2, determining an index value of the first voice digital signal based on the position.
In the embodiment of the invention, all numerical digits of the first voice digital signal are read in sequence from left to right, and the first numerical digit which is 1 is determined as a set index; and determining the position of the numerical digit with the first value of 1 as the index value of the first voice digital signal.
For example, the first speech digital signal 1 is 0001010111101010, the first numerical bit in the first speech digital signal 1 is set as the 0 th bit, and so on, the last numerical bit is the 15 th bit, so the index position is set to be 3, that is, the index value of the first speech digital signal 1 is 3.
S103: and sequentially carrying out redundancy value truncation and sign bit resetting on the second voice digital signal to obtain and output a third voice digital signal.
The first voice digital signal is compressed to ensure that the signal is compressed to a plurality of high-value bits, and in order to reduce bandwidth pressure during signal transmission and save the storage space occupied during signal storage, the second voice digital signal needs to be subjected to redundancy truncation processing. Optionally, the process of performing redundancy value truncation processing on the second voice digital signal is as follows: and according to the set sequence, when the continuous n numerical digits in the second voice digital signal are determined to be invalid numerical digits, performing truncation processing on the continuous n numerical digits.
For example, the second speech digital signal 1 (00111110000000000) has n consecutive numerical bits 0 after the 7 th numerical bit, and therefore 7 to 15 numerical bits are truncated.
In the embodiment of the invention, in order to reduce quantization error caused by direct compression of a negative value, an original voice digital signal is processed to obtain a first voice digital signal without a sign bit binary number, a second-order fitting curve is adopted to compress the first voice digital signal, and an obtained second voice digital signal is also without a sign bit binary number, but the voice digital signal finally transmitted to other users needs to be a binary number with a sign bit, so that sign bit resetting processing needs to be carried out on the second voice digital signal after the redundancy value truncation processing. Optionally, the process of performing sign bit reset processing on the second voice digital signal includes: and adding the sign bit of the original voice digital signal into the second voice digital signal to complete the sign bit resetting processing of the second voice digital signal.
For example, the first speech digital signal 1 (0001010111101010) is compressed, and the second speech digital signal 1 subjected to the redundancy truncation process is 0011111, while the sign bit of the original speech digital signal 1 is 0, so that the second speech digital signal 1 subjected to the sign bit resetting process is 00011111.
For another example, since the second speech digital signal 2 subjected to the redundancy truncation process by compressing the first speech digital signal 2 (1110101100100011) is 1110101 and the sign bit of the original speech digital signal 2 is 1, the second speech digital signal 2 subjected to the sign bit resetting process is 11110101.
For the sake of understanding, the process of compressing the speech digital signal will be described by taking a specific embodiment as an example, referring to fig. 2.
S201: the sign bit of the original speech digital signal 2 (10001010011011101) is determined, and the modulo operation is performed on the original speech digital 2 by using the Absolute value (Abs) to obtain the corresponding first speech digital signal 2 (1110101100100011).
S202: a vector normalization (vnorm) operation is performed on the first speech digital signal 2, and an exponent value of the first speech digital signal 2 is determined to be 0.
S203: calling a second-order fitting curve 1-49300 matched with the index value 0 x +19800, the first speech digital signal 2 is compressed to obtain the second speech digital signal 2 (1110101000000000).
S204: and (4) carrying out truncation processing on lower 6-7bit numerical bits in the second voice digital signal 2 to obtain a second voice digital signal 2 after truncation (1110101).
S205: sign bit 1 of the original voice digital signal 2 is added to the truncated second voice digital signal 2 (1110101), and the sign bit reset processed second voice digital signal 2 is 11110101.
S206: the compressed second speech digital signal 2 (11110101) is output.
Based on the same inventive concept, in an embodiment of the present invention, an apparatus for compressing a speech digital signal is provided, as shown in fig. 3, and includes at least a converting unit 301, a compressing unit 302, and a processing unit 303, wherein,
a conversion unit 301, configured to obtain a first voice digital signal based on an original voice digital signal, where the first voice digital signal is a binary number without a sign bit;
a compressing unit 302, configured to determine an index value of the first speech digital signal based on a preset normalization rule, call a second-order fitting curve matched with the index value, and compress the first speech digital signal to obtain a second speech digital signal;
and the processing unit 303 is configured to perform redundancy truncation and sign bit resetting on the second voice digital signal in sequence to obtain and output a third voice digital signal.
Optionally, based on the original speech digital signal, a first speech digital signal is obtained, and the converting unit 301 is configured to:
taking an absolute value of the numerical digit of the original voice digital signal to obtain the first voice digital signal; or,
when the original voice digital signal is determined to be a positive number, determining the numerical digit of the original voice digital signal as the first voice digital signal;
and when the original voice digital signal is determined to be a negative number, taking an absolute value of the numerical value of the original voice digital signal to obtain the first voice digital signal.
Optionally, the conversion unit 301 is configured to take an absolute value of a numerical bit of the original digital signal:
acquiring a sign bit of the original voice digital signal;
if the sign bit represents that the original voice digital signal is a positive number, performing original code operation on a numerical value bit in the original voice digital signal;
and if the sign bit represents that the original voice digital signal is a negative number, performing complement operation on a numerical value bit in the original voice digital signal.
Optionally, based on a preset normalization rule, an index value of the first speech digital signal is determined, and the compression unit 302 is configured to:
determining the position of a set index in the first voice digital signal according to a set sequence;
based on the location, an exponent value of the first speech digital signal is determined.
Optionally, the processing unit 303 is configured to perform redundancy truncation processing on the second speech digital signal:
and according to a set sequence, when the n continuous numerical digits in the second voice digital signal are determined to be invalid numerical digits, carrying out truncation processing on the n continuous numerical digits.
Optionally, after performing redundancy value truncation processing on the second voice digital signal, further performing sign bit resetting processing on the second voice digital signal, where the processing unit 303 is configured to:
and adding the sign bit of the original voice digital signal to the second voice digital signal.
Based on the same inventive concept, in the embodiment of the present invention, a computing device is provided, as shown in fig. 4, which at least includes a memory 401 and at least one processor 402, where the memory 401 and the processor 402 complete communication with each other through a communication bus;
the memory 401 is used to store program instructions;
the processor 402 is used for calling the program instructions stored in the memory 401 and executing the aforementioned compression method of the speech digital signal according to the obtained program.
Based on the same inventive concept, in an embodiment of the present invention, a storage medium is provided, which at least includes computer readable instructions, and when the computer reads and executes the computer readable instructions, the computer is caused to execute the foregoing compression method for a speech digital signal.
In summary, the first speech digital signal of binary number without sign bit is obtained based on the original speech digital signal; based on normalization rules, determining an index value of the first voice digital signal, calling a second-order fitting curve matched with the index value, and compressing the first voice digital signal to obtain a second voice digital signal; and finally, sequentially carrying out sign bit resetting and redundancy value truncation processing on the second voice digital signal to obtain and output a third voice digital signal.
The original voice digital signal is converted into a first voice digital signal of a binary number without a sign bit, so that the quantization error caused by direct negative value compression can be reduced; and then calling a second-order fitting curve matched with the index value of the first voice digital signal to compress the first voice digital signal.
As will be appreciated by one skilled in the art, embodiments of the present invention may be provided as a method, system, or computer program product. Accordingly, the present invention may take the form of an entirely hardware embodiment, an entirely software embodiment or an embodiment combining software and hardware aspects. Furthermore, the present invention may take the form of a computer program product embodied on one or more computer-usable storage media (including, but not limited to, disk storage, CD-ROM, optical storage, and the like) having computer-usable program code embodied therein.
The present invention is described with reference to flowchart illustrations and/or block diagrams of methods, apparatus (systems), and computer program products according to embodiments of the invention. It will be understood that each flow and/or block of the flow diagrams and/or block diagrams, and combinations of flows and/or blocks in the flow diagrams and/or block diagrams, can be implemented by computer program instructions. These computer program instructions may be provided to a processor of a general purpose computer, special purpose computer, embedded processor, or other programmable data processing apparatus to produce a machine, such that the instructions, which execute via the processor of the computer or other programmable data processing apparatus, create means for implementing the functions specified in the flowchart flow or flows and/or block diagram block or blocks.
These computer program instructions may also be stored in a computer-readable memory that can direct a computer or other programmable data processing apparatus to function in a particular manner, such that the instructions stored in the computer-readable memory produce an article of manufacture including instruction means which implement the function specified in the flowchart flow or flows and/or block diagram block or blocks.
These computer program instructions may also be loaded onto a computer or other programmable data processing apparatus to cause a series of operational steps to be performed on the computer or other programmable apparatus to produce a computer implemented process such that the instructions which execute on the computer or other programmable apparatus provide steps for implementing the functions specified in the flowchart flow or flows and/or block diagram block or blocks.
While preferred embodiments of the present invention have been described, additional variations and modifications in those embodiments may occur to those skilled in the art once they learn of the basic inventive concepts. Therefore, it is intended that the appended claims be interpreted as including preferred embodiments and all such alterations and modifications as fall within the scope of the invention.
It will be apparent to those skilled in the art that various modifications and variations can be made in the embodiments of the present invention without departing from the spirit or scope of the embodiments of the invention. Thus, if such modifications and variations of the embodiments of the present invention fall within the scope of the claims of the present invention and their equivalents, the present invention is also intended to encompass such modifications and variations.
Claims (12)
1. A method of compressing a speech digital signal, comprising:
obtaining a first voice digital signal based on an original voice digital signal, wherein the first voice digital signal is a binary number without sign bits;
determining an index value of the first voice digital signal based on a preset normalization rule, calling a second-order fitting curve matched with the index value, and compressing the first voice digital signal to obtain a second voice digital signal;
sequentially carrying out redundancy value truncation and sign bit resetting on the second voice digital signal to obtain and output a third voice digital signal;
the determining the index value of the first speech digital signal based on a preset normalization rule comprises:
determining the position of a set index in the first voice digital signal according to a set sequence;
based on the location, an exponent value of the first speech digital signal is determined.
2. The method of claim 1, wherein deriving the first speech digital signal based on an original speech digital signal comprises:
taking absolute values of numerical digits of the original voice digital signals to obtain first voice digital signals; or,
when the original voice digital signal is determined to be a positive number, determining the numerical digit of the original voice digital signal as the first voice digital signal;
and when the original voice digital signal is determined to be a negative number, taking an absolute value of the numerical value of the original voice digital signal to obtain the first voice digital signal.
3. The method of claim 2, wherein taking an absolute value of the numerical bits of the original digital signal comprises:
acquiring a sign bit of the original voice digital signal;
if the sign bit represents that the original voice digital signal is a positive number, performing original code operation on a numerical value bit in the original voice digital signal;
and if the sign bit represents that the original voice digital signal is a negative number, performing complement operation on a numerical value bit in the original voice digital signal.
4. The method of any one of claims 1-3, wherein performing redundancy truncation processing on the second speech digital signal comprises:
and according to a set sequence, when the n continuous numerical digits in the second voice digital signal are determined to be invalid numerical digits, carrying out truncation processing on the n continuous numerical digits.
5. The method of claim 4, wherein after the second voice digital signal is redundancy truncation processed, further performing a sign bit reset process on the second voice digital signal, comprising:
and adding the sign bit of the original voice digital signal to the second voice digital signal.
6. An apparatus for compressing a speech digital signal, comprising:
the conversion unit is used for obtaining a first voice digital signal based on an original voice digital signal, wherein the first voice digital signal is a binary number without sign bits;
the compression unit is used for determining an index value of the first voice digital signal based on a preset normalization rule, calling a second-order fitting curve matched with the index value, and compressing the first voice digital signal to obtain a second voice digital signal;
the processing unit is used for sequentially carrying out redundancy value truncation and sign bit resetting on the second voice digital signal to obtain and output a third voice digital signal;
the compression unit is further configured to:
determining the position of a set index in the first voice digital signal according to a set sequence;
based on the location, an index value of the first speech digital signal is determined.
7. The apparatus of claim 6, wherein the first speech digital signal is derived based on an original speech digital signal, the conversion unit being configured to:
taking absolute values of numerical digits of the original voice digital signals to obtain first voice digital signals; or,
when the original voice digital signal is determined to be a positive number, determining the numerical digit of the original voice digital signal as the first voice digital signal;
and when the original voice digital signal is determined to be a negative number, taking an absolute value of the numerical value of the original voice digital signal to obtain the first voice digital signal.
8. The apparatus of claim 7, wherein the conversion unit is to, for the numerical bits of the original digital signal, take absolute values:
acquiring a sign bit of the original voice digital signal;
if the sign bit represents that the original voice digital signal is a positive number, performing original code operation on a numerical value bit in the original voice digital signal;
and if the sign bit represents that the original voice digital signal is a negative number, performing complement operation on a numerical value bit in the original voice digital signal.
9. The apparatus according to any one of claims 6 to 8, wherein the second speech digital signal is subjected to a redundancy truncation process, and the processing unit is configured to:
and according to a set sequence, when the n continuous numerical digits in the second voice digital signal are determined to be invalid numerical digits, carrying out truncation processing on the n continuous numerical digits.
10. The apparatus of claim 9, wherein after the second speech digital signal is redundancy truncation processed, further sign bit reset processing is performed on the second speech digital signal, the processing unit being configured to:
and adding the sign bit of the original voice digital signal to the second voice digital signal.
11. A computing device, comprising:
a memory for storing program instructions;
a processor for calling program instructions stored in said memory to execute the method of any one of claims 1 to 5 in accordance with the obtained program.
12. A storage medium comprising computer readable instructions which, when read and executed by a computer, cause the computer to perform the method of any one of claims 1 to 5.
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