CN106060740B

CN106060740B - Microphone data processing method and device

Info

Publication number: CN106060740B
Application number: CN201610353174.5A
Authority: CN
Inventors: 冯穗豫; 肖典欢; 张德文
Original assignee: Tencent Technology Shenzhen Co Ltd
Current assignee: Tencent Technology Shenzhen Co Ltd
Priority date: 2016-05-24
Filing date: 2016-05-24
Publication date: 2020-07-17
Anticipated expiration: 2036-05-24
Also published as: CN106060740A

Abstract

The invention relates to a method and a device for processing microphone data, comprising the following steps: acquiring information to be coded related to a microphone; encoding the information to be encoded into an encoded signal sequence formed by signals with different frequencies, wherein the frequency of the encoded signal sequence is greater than the upper limit frequency of human auditory perception; the method comprises the steps of acquiring an original audio signal acquired by a microphone, superposing the coding signal sequence with the original audio signal to obtain an audio signal carrying microphone related information, and sending the audio signal to a terminal so that the terminal can analyze the audio signal carrying the microphone related information to obtain corresponding microphone related information, so that the microphone information can be conveniently transmitted, and the terminal can conveniently acquire the microphone related information.

Description

Microphone data processing method and device

Technical Field

The invention relates to the technical field of computers, in particular to a method and a device for processing microphone data.

Background

With the development of computer technology, various network applications are more and more common in people's daily life, and users play entertainment and work through various network applications, such as recording and sharing songs through a microphone by adopting a karaoke application. Many hardware manufacturers have produced specialized microphones for sound enhancement that have been aesthetically pleasing in all respects.

In the existing microphone data processing method, only analog signals are transmitted by a microphone through an interface, and often only recorded audio data are transmitted to a playing terminal, so that the information of the microphone cannot be identified when the playing terminal plays audio, and corresponding processing cannot be performed.

Disclosure of Invention

Therefore, it is necessary to provide a method and an apparatus for processing microphone data, which can conveniently transfer microphone information and facilitate a terminal to obtain microphone related information.

A method of microphone data processing, the method comprising:

acquiring information to be coded related to a microphone;

encoding the information to be encoded into an encoded signal sequence formed by signals with different frequencies, wherein the frequency of the encoded signal sequence is greater than the upper limit frequency of human auditory perception;

acquiring an original audio signal acquired by a microphone, superposing the coding signal sequence and the original audio signal to obtain an audio signal carrying microphone related information, and sending the audio signal carrying the microphone related information to a terminal so that the terminal can analyze the audio signal carrying the microphone related information to obtain corresponding microphone related information.

An apparatus for microphone data processing, the apparatus comprising:

the acquisition module is used for acquiring information to be coded related to the microphone;

the coding module is used for coding the information to be coded into a coding signal sequence formed by signals with different frequencies, and the frequency of the coding signal sequence is greater than the upper limit frequency of human ear auditory perception;

and the sending module is used for acquiring an original audio signal acquired by a microphone, superposing the coding signal sequence and the original audio signal to obtain an audio signal carrying microphone related information, and sending the audio signal carrying the microphone related information to the terminal so that the terminal can analyze the audio signal carrying the microphone related information to obtain corresponding microphone related information.

The method and the device for processing the microphone data acquire the original audio signals collected by the microphone by acquiring the information to be coded related to the microphone and coding the information to be coded into the coding signal sequence formed by signals with different frequencies, wherein the frequency of the coding signal sequence is greater than the upper limit frequency of auditory perception of human ears, the audio signals carrying the information related to the microphone are obtained by overlapping the coding signal sequence and the original audio signals and are sent to the terminal, so that the terminal analyzes the audio signals carrying the information related to the microphone to obtain the corresponding information related to the microphone, the way of coding the information related to the microphone into the signals with different frequencies is convenient for the microphone interface to output through analog signals, the microphone can output the information related to the microphone under the condition that the microphone does not support digital signal output, and the frequency of the coding signal sequence is greater than the upper limit frequency of auditory perception of human ears, the difficulty of separating the coded signal sequence is reduced, the microphone information can be conveniently transmitted, and the terminal can conveniently acquire the microphone related information.

A method of microphone data processing, the method comprising:

receiving an audio signal sent by a microphone;

if the audio signal is an audio signal carrying microphone related information, acquiring a coding signal sequence in the audio signal, wherein the coding signal sequence is formed by different frequency signals generated by coding microphone related information to be coded, and the frequency of the coding signal sequence is greater than the upper limit frequency of human ear auditory perception;

and analyzing the coded signal sequence, and decoding to obtain the microphone related information.

An apparatus for microphone data processing, the apparatus comprising:

the receiving module is used for receiving the audio signal sent by the microphone;

the encoding signal sequence acquisition module is used for acquiring an encoding signal sequence in the audio signal if the audio signal is the audio signal carrying the microphone related information, wherein the encoding signal sequence is formed by different frequency signals generated by encoding the information to be encoded related to the microphone, and the frequency of the encoding signal sequence is greater than the upper limit frequency of human ear auditory perception;

and the decoding module is used for analyzing the coding signal sequence and decoding to obtain the microphone related information.

According to the microphone data processing method and device, whether the audio signal is the audio signal carrying the microphone related information is detected by receiving the audio signal sent by the microphone, if so, a coding signal sequence in the audio signal is obtained, the coding signal sequence is composed of different frequency signals generated by coding the information to be coded related to the microphone, the frequency of the coding signal sequence is larger than the upper limit frequency of human ear auditory perception, the coding signal sequence is analyzed and decoded to obtain the microphone related information, the coding signal sequence is composed of different frequency signals generated by coding the information to be coded related to the microphone, the microphone interface can output the audio signal carrying the microphone related information through analog signals, and the microphone can output the microphone related information under the condition that the microphone does not support digital signal output. And the frequency of the coding signal sequence is greater than the upper limit frequency of human auditory perception, so that the coding signal sequence can be conveniently separated from the original audio signal without influencing the original audio signal, microphone information can be conveniently transmitted, and a terminal can conveniently acquire microphone related information.

Drawings

FIG. 1 is a diagram of an exemplary implementation of a method for processing microphone data;

FIG. 2 is a diagram illustrating the internal structure of the microphone of FIG. 1 according to one embodiment;

FIG. 3 is a diagram illustrating an internal structure of the terminal of FIG. 1 according to one embodiment;

FIG. 4 is a flow diagram of a method of microphone data processing in one embodiment;

FIG. 5 is a flow diagram of encoded signal sequence generation in one embodiment;

FIG. 6 is a spectrum diagram of a minimum coding unit according to an embodiment;

FIG. 7 is a spectrum diagram of another exemplary minimum coding unit;

FIG. 8 is a spectrum diagram of an embodiment of a coded signal sequence consisting of minimum coding units;

FIG. 9 is a diagram of a spectrum formed by the superposition of an encoded signal sequence and an original audio signal according to an embodiment;

FIG. 10 is a flow diagram of another method of microphone data processing in one embodiment;

FIG. 11 is a flow diagram of decoding to obtain microphone related information in one embodiment;

FIG. 12 is a block diagram of an apparatus for microphone data processing in one embodiment;

FIG. 13 is a block diagram of the structure of an encoding module in one embodiment;

FIG. 14 is a block diagram of an alternative embodiment of an apparatus for microphone data processing;

FIG. 15 is a block diagram of a decoding module in one embodiment;

fig. 16 is a block diagram of another microphone data processing apparatus according to still another embodiment.

Detailed Description

FIG. 1 is a diagram of an application environment in which a method for controlling an application operates, according to one embodiment. As shown in fig. 1, the application environment includes a microphone 110, a terminal 120, wherein the microphone 110 is connected to the terminal 120 in a plug-in manner, or the microphone 110 and the terminal 120 communicate through a network.

The microphone 110 may be a moving coil type, a crystal type, a carbon particle type, an aluminum tape type, a capacitor type, and the like, and the terminal 110 may be a smart phone, a tablet computer, a notebook computer, a desktop computer, and the like, but is not limited thereto. The microphone 110 may be plugged into the terminal 120 or wirelessly, or may send data to the terminal 120 through a network, and the terminal 120 may receive the data transmitted by the microphone 110 and perform parsing to obtain microphone related information.

In one embodiment, the internal structure of the microphone 110 shown in fig. 1 is as shown in fig. 2, and includes an input device, a processor, a storage medium, a memory, and an interface, wherein the input device is used for receiving vibration input of sound, forming a varying current, the varying current is amplified and processed by the processor, and the storage medium stores a first microphone data processing device, which is used for implementing a microphone data processing method suitable for a microphone. The processor is used to provide computational and control capabilities, supporting the operation of the entire microphone. The memory provides an environment for the operation of the first microphone data processing device in the storage medium, the interface is used for data transmission with the terminal, and the interface can be a data line interface or a wireless interface.

In one embodiment, the internal structure of the terminal 120 in fig. 1 is as shown in fig. 3, and the terminal 120 includes a processor, a graphic processing unit, a storage medium, a memory, a network interface, a display screen, and an input device, which are connected through a system bus. The storage medium of the terminal 120 stores an operating system, and further includes a second microphone data processing device, which is used to implement a microphone data processing method suitable for the terminal. The processor is used to provide computational and control capabilities that support the operation of the entire terminal 120. The graphic processing unit in the terminal 120 is configured to at least provide a rendering capability of a display interface, the memory provides an environment for the operation of the second microphone data processing apparatus in the storage medium, and the network interface is configured to perform network communication with the microphone 110, such as receiving audio data sent by the microphone 110. The display screen is used for displaying an application interface and the like, such as displaying a playing interface and the like, and the input device is used for receiving commands or data and the like input by a user. The terminal 110 is provided with a touch screen, and the display screen and the input device may be a touch screen.

As shown in fig. 4, in one embodiment, a method for processing microphone data is provided, which is exemplified by a microphone applied in the above application environment, and includes the following steps:

in step S210, information to be encoded related to the microphone is acquired.

Specifically, the information to be encoded related to the microphone may be attribute information of the microphone, operation state information of the microphone, function information, and the like. For example, the attribute information of the microphone may be the model and manufacturer information of the microphone, the operation state information of the microphone may be the position information of the clipper of the microphone, and the like, and the function information may be the reverberation depth information, the volume information, the corresponding relationship between the clipper and the function, and the like, for example, the clipper 3 is the volume in the first scene, and the clipper 3 is the reverberation in the second scene.

Step S220, the information to be coded is coded into a coded signal sequence formed by signals with different frequencies, and the frequency of the coded signal sequence is greater than the upper limit frequency of human auditory perception.

Specifically, the upper limit frequency of human auditory perception is 20KHz, and signals above 20KHz of the audio signal are weak or almost none. Because the frequency of the coded signal sequence coded by the information to be coded is greater than the upper limit frequency of human auditory perception, the coded signal sequence can be conveniently separated from the original audio signal without influencing the original audio signal. And even if the coded signal sequence is not separated, the human ear cannot perceive the coded signal sequence after playing because the frequency of the coded signal sequence is greater than the upper limit frequency of human ear auditory perception, thereby reducing the separation difficulty and ensuring the tone quality effect.

The frequency of the encoded signal sequence can be between 20KHz and 22KHz, and common lossy encoding, such as MP3, ogg (oggvrbis), AAC (Advanced Audio Coding) removes frequencies above 20KHz, so that basically no additional filtering process is required when generating Audio data. And for the situation that the recorded CD sound quality is released, the terminal can also use a low-pass filter to filter and store the information after the information related to the microphone is obtained by analysis. The information to be coded is converted into the arrangement combination of the minimum coding units with limited quantity according to the coding algorithm, the specific conversion algorithm can be customized according to the needs, for example, the specific conversion algorithm is determined by a preset binary coding method, such as a hexadecimal coding method, an octal coding method, a stroke coding method and the like, different coding conversion methods correspond to the minimum coding units with different numbers and types, for example, for the hexadecimal coding method, the minimum coding unit is 16 characters in total from 0 to F, for the octal coding method, the minimum coding unit is 7 characters in total from 0 to 7, and for the stroke coding method, the minimum coding unit is the minimum strokes such as "-", "|", and the like. Each minimum coding unit is pre-allocated with corresponding different frequency information, one minimum coding unit corresponds to one frequency during allocation, and one minimum coding unit can also be represented by two different frequencies, so that the utilization rate of the frequencies is reduced, the space between different frequencies is ensured, and the interference generated between different frequencies is avoided. The sine wave signal y is obtained by a formula y 32768 x 0.5 x sin (2 x PI t Freq/Samplerate), wherein PI is a circumferential rate, t represents time, Freq represents frequency, and Samplerate represents a sampling rate of sound, such as 44100, wherein 0.5 corresponds to a signal of-6 dB, so that the signal corresponding to the minimum generated coding unit is not too weak or too strong, the signal is too strong to generate overflow in superposition, such as clipping or sound breaking, and the signal is too weak to be easily distinguished from the original recorded sound in analysis, so that the corresponding sine wave signal can be obtained according to the frequency corresponding to the minimum coding unit. Each character in the information to be coded can be converted into a coded character consisting of minimum coding units, and the minimum coding units correspond to signals with different frequencies, so that the information to be coded is converted into a coded signal sequence consisting of signals with different frequencies.

Step S230, acquiring an original audio signal collected by the microphone, superimposing the encoded signal sequence and the original audio signal to obtain an audio signal carrying the microphone related information, and sending the audio signal to the terminal, so that the terminal analyzes the audio signal carrying the microphone related information to obtain the corresponding microphone related information.

Specifically, the coding signal sequence carries microphone related information, and the way of coding the microphone related information into signals of different frequencies facilitates the microphone interface to output the microphone related information through analog signals, so that the microphone can also output the microphone related information under the condition that the microphone does not support digital signal output. In one embodiment, in order to avoid the existence of an interference signal in a portion of the original audio signal having a frequency greater than the upper limit frequency of human auditory perception, the original audio signal may be subjected to low-pass filtering with a preset cut-off frequency, such as 20KHz, to remove a high-frequency signal, and then the original audio signal may be superimposed with the encoded signal sequence to obtain an audio signal carrying microphone related information, so as to facilitate subsequent extraction of the encoded signal sequence without the interference signal from the audio signal. After receiving the audio signal carrying the microphone related information, the terminal may detect the encoded signal sequence therein, where the microphone may carry identification information in the encoded signal sequence, or appoint the position, time, etc. of the encoded signal sequence with the terminal, for example, the insertion position of the encoded signal sequence is a preset proportion of the total length of the audio signal, or a preset proportion of the total time of the audio signal, in order to facilitate the terminal to detect the complete encoded signal sequence. After the terminal separates out the coded signal sequence, the corresponding microphone related information can be obtained through a decoding algorithm corresponding to the microphone.

In the embodiment, the information to be coded related to the microphone is obtained, the information to be coded is coded into a coding signal sequence formed by signals with different frequencies, the frequency of the coding signal sequence is greater than the upper limit frequency of human auditory perception, the original audio signal collected by the microphone is obtained, the coding signal sequence and the original audio signal are superposed to obtain an audio signal carrying the information related to the microphone and the audio signal is sent to the terminal, so that the terminal analyzes the audio signal carrying the information related to the microphone to obtain corresponding information related to the microphone, the microphone related information is coded into signals with different frequencies, the microphone interface can conveniently output the information related to the microphone through analog signals, the microphone can also output the information related to the microphone under the condition that the microphone does not support digital signal output, the frequency of the coding signal sequence is greater than the upper limit frequency of human auditory perception, and the difficulty in separating the coding signal sequence is reduced, microphone information can be conveniently transmitted, and the terminal can conveniently acquire microphone related information.

In one embodiment, as shown in fig. 5, step S220 includes:

step S221, the information to be coded is divided and coded into coding sub-characters corresponding to a preset coding system, and the coding sub-characters are formed by minimum coding units in sequence.

Specifically, the information to be encoded is divided into characters to be encoded according to a preset unit, for example, the characters are divided into the characters to be encoded by taking the characters as a unit, for example, "middle 3" can be divided into "middle" and "3", the encoded sub-characters obtained from the characters to be encoded are also different according to the difference of the preset encoding system, for example, the preset encoding system is hexadecimal, the encoded sub-character corresponding to "middle" is 0x4E2D, the encoded sub-character corresponding to "3" is 0x0030, the encoded sub-characters are sequentially composed of minimum encoding units, for example, 0x4E2D is composed of 4 minimum encoding units, namely "4", "E", "2", "D".

Step S222, sequentially obtaining the high frequency and the low frequency corresponding to each minimum coding unit, generating the coded sub-signals corresponding to the minimum coding units according to the high frequency and the low frequency, and forming the coded signal sequence by the sequentially generated coded sub-signals.

Specifically, a minimum coding unit is represented by high-frequency and low-frequency, and a minimum coding unit is represented by two frequencies, so that the distance between the frequencies is large as much as possible, and interference is avoided. The correspondence between the high-frequency and the low-frequency and the minimum coding unit may be preset, so that the high-frequency and the low-frequency corresponding to the minimum coding unit are obtained according to the correspondence. And obtaining a corresponding sine wave signal or cosine wave signal according to the high-frequency and the low-frequency through a formula, generating a coding sub-signal corresponding to the minimum coding unit, and forming a coding signal sequence by sequentially generating the coding sub-signals. The time interval between each encoded sub-signal can be customized as desired, e.g., defined to be more than 30 milliseconds.

In one embodiment, step S222 includes: and acquiring a coding table corresponding to a preset coding system, and acquiring the high-frequency and the low-frequency corresponding to the minimum coding unit according to a table look-up mode.

Specifically, a table of the corresponding relationship between the high frequency and the low frequency and the minimum coding unit is determined in advance according to a preset coding system, so that the high frequency and the low frequency corresponding to the minimum coding unit can be quickly obtained in a table look-up mode. As shown in table 1 below, a table corresponding to the hexadecimal coding method is shown, wherein the horizontal axis represents low frequency, the vertical axis represents high frequency, and the characters in the table are the minimum coding units corresponding to the hexadecimal coding method.

TABLE 1

	20200	20400	20600	20800
					21000	0	1	2	3
21200	4	5	6	7
					21400	8	9	A	B
21600	C	D	E	F

If "4" is assigned to the high frequency 21200Hz and the low frequency 20200Hz, the signal assigned to "4" is formed by 21200Hz and 20200 Hz. It can be understood that the specific values and the interval gaps of the high frequency and the low frequency in the table can be customized as required, and the high frequency and the low frequency at reasonable intervals are designed according to different preset code systems. In one embodiment, the table corresponding to the octal coding method is shown in table 2:

TABLE 2

	20200	20550	20900	21250
					21600	0	1	2	3
21950	4	5	6	7

In one embodiment, the encoded signal sequence is generated periodically, the encoded signal sequence carrying the header identification information and the trailer identification information in a predetermined format.

Specifically, the coded signal sequence is generated periodically without the need of the microphone and the terminal to appoint the specific position or the specific time of the coded signal sequence, which is simple and convenient, and only the head identification information of the preset format is added at the head part of the coded signal sequence and the tail identification information of the preset format is added at the tail part of the coded signal sequence. The specific format and characters of the header identification information and the tail identification information in the preset format can be customized according to needs.

In a specific embodiment, the corresponding relationship in table 1 is adopted for encoding, an encoded sub-character corresponding to the information to be encoded is "4E", after the encoded sub-signal corresponding to 4 is constructed, the corresponding spectrogram is shown in fig. 6, and after the encoded sub-signal corresponding to E is constructed, the corresponding spectrogram is shown in fig. 7. A spectrogram corresponding to an encoded signal sequence formed by a predetermined time interval "4E" is shown in fig. 8. The encoded signal sequence "4E" is generated periodically, e.g., once in 10 seconds. The spectrogram formed by overlapping the encoded signal sequence and the original audio signal is shown in fig. 9, and it can be seen that the encoded signal sequence 310 is obviously above the frequency spectrum, and the original audio signal 320 is concentrated below the frequency spectrum, which is convenient for terminal separation and extraction.

In one embodiment, as shown in fig. 10, a method for processing microphone data is provided, which is exemplified by a terminal applied in the above application environment, and includes the following steps:

step S410, receiving the audio signal sent by the microphone.

Specifically, the audio signal transmitted by the microphone may carry microphone-related information, and the microphone-related information may be attribute information of the microphone, operation state information of the microphone, function information, and the like. For example, the attribute information of the microphone may be the model and manufacturer information of the microphone, the operation state information of the microphone may be the position information of the clipper of the microphone, and the like, and the function information may be the reverberation depth information, the volume information, the corresponding relationship between the clipper and the function, and the like, for example, the clipper 3 is the volume in the first scene, and the clipper 3 is the reverberation in the second scene.

Step S420, if the audio signal is an audio signal carrying microphone related information, acquiring a coding signal sequence in the audio signal, where the coding signal sequence is formed by different frequency signals generated by coding microphone related information to be coded, and the frequency of the coding signal sequence is greater than the upper limit frequency of human ear auditory perception.

Specifically, whether the audio signal carries microphone related information may be detected by using a flag bit or by using a plurality of methods, for example, if the flag bit is 1, the microphone related information is carried, and the encoded signal sequence may be extracted according to a position, time, and the like of the encoded signal sequence agreed with the microphone, for example, an insertion position of the encoded signal sequence is a preset proportion of the total length of the audio signal, or a preset proportion of the total time of the audio signal. Or by detecting whether the audio signal carries preset identification information, for example, the header identification information and the tail identification information carry preset format, the part between the header identification information and the tail identification information is a coding signal sequence. The coded signal sequence carries microphone related information, the coded signal sequence is formed by different frequency signals generated by coding the information to be coded related to the microphone, and therefore the microphone interface can conveniently output audio signals carrying the microphone related information through analog signals, and the microphone can also output the microphone related information under the condition that the microphone does not support digital signal output. And the frequency of the coding signal sequence is greater than the upper limit frequency of human auditory perception, so that the coding signal sequence can be conveniently separated from the original audio signal without influencing the original audio signal. And even if the coded signal sequence is not separated, the human ear cannot perceive the coded signal sequence after playing because the frequency of the coded signal sequence is greater than the upper limit frequency of human ear auditory perception, thereby reducing the separation difficulty and ensuring the tone quality effect.

And step S430, analyzing the coded signal sequence, and decoding to obtain microphone related information.

Specifically, a decoding algorithm of the terminal corresponds to an encoding algorithm of the microphone, and the encoding sub-signals corresponding to the encoding signal sequence are obtained first, and because the encoding sub-signals are separated by a preset time interval, the encoding sub-signals can be sequentially obtained according to a time sequence. The frequencies of the encoded sub-signals are obtained, and depending on the representation method of the encoded sub-signals, one encoded sub-signal may comprise one or more frequencies. And acquiring a corresponding minimum decoding unit according to the frequency, and if the minimum decoding unit is a plurality of frequencies, simultaneously determining the minimum decoding unit according to the plurality of frequencies. The corresponding relation between the frequency and the minimum decoding unit corresponds to a coding end, the number and the type of the minimum decoding unit also correspond to a coding algorithm, such as a preset binary coding method including a hexadecimal coding method, an octal coding method, a stroke coding method and the like, different coding methods correspond to the minimum coding units with different numbers and types, for example, for the hexadecimal coding method, the minimum coding unit is 16 characters in total from 0 to F, for the octal coding method, the minimum coding unit is 7 characters in total from 0 to 7, and for the stroke coding method, the minimum coding unit is the minimum strokes of "-", "|" and the like. The coding algorithm can be acquired in a form appointed with a microphone, or coding algorithm information is directly carried in an audio signal, and the terminal acquires the coding algorithm information in an extraction mode, so that a decoding algorithm corresponding to the coding algorithm can be determined. After the frequency of the encoded sub-signal is obtained, the minimum decoding unit corresponding to the frequency can be obtained according to the corresponding relation between the frequency and the minimum decoding unit. Different minimum decoding units are sequentially arranged, and as a plurality of continuous minimum decoding units form a decoding sub-character generally and have specific meanings, each minimum decoding unit needs to be separated to obtain a corresponding decoding sub-character. If the encoding method is a preset system encoding method, the preset digit corresponding to the preset encoding system can be directly obtained for separation, and if the preset digit corresponding to the hexadecimal system is 4 digits. If the stroke coding algorithm is adopted, the separation position of each decoding sub-character needs to be written in the coding end in advance, the separation position can be written by a method of presetting head and tail marks, and each minimum decoding unit is separated into the decoding sub-characters according to the preset head and tail marks during decoding. And obtaining corresponding microphone related information according to the decoded sub-characters, wherein each decoded sub-character has a specific decoding relationship with the decoded information, for example, the corresponding decoded information in the hexadecimal decoded sub-character 4E2D is 'middle', so that each decoded information is obtained to form the microphone related information.

In this embodiment, whether the audio signal is an audio signal carrying microphone related information is detected by receiving an audio signal sent by a microphone, if so, a coding signal sequence in the audio signal is obtained, where the coding signal sequence is formed by different frequency signals generated by coding information to be coded related to the microphone, the frequency of the coding signal sequence is greater than an upper limit frequency of human ear auditory perception, the coding signal sequence is analyzed and decoded to obtain the microphone related information, and the coding signal sequence is formed by different frequency signals generated by coding information to be coded related to the microphone, so that a microphone interface outputs the audio signal carrying the microphone related information through an analog signal, and the microphone can also output the microphone related information under the condition that the microphone does not support digital signal output. And the frequency of the coding signal sequence is greater than the upper limit frequency of human auditory perception, so that the coding signal sequence can be conveniently separated from the original audio signal without influencing the original audio signal, microphone information can be conveniently transmitted, and a terminal can conveniently acquire microphone related information.

In one embodiment, as shown in fig. 11, step S430 includes:

in step S431, the encoded sub-signal corresponding to the encoded signal sequence is obtained, and the high frequency and the low frequency of the encoded sub-signal are obtained.

Specifically, the encoding sub-signal is composed of a high-frequency and a low-frequency, so that the distance between the frequencies is large as much as possible, and interference is avoided.

Step S432, a corresponding minimum decoding unit is obtained according to the high frequency and the low frequency.

Specifically, a preset correspondence between the high frequency and the low frequency and the minimum decoding unit is obtained, so that the corresponding minimum decoding unit is obtained according to the high frequency and the low frequency. The decoding algorithm is different, and the obtained minimum decoding unit is also different.

And step S433, arranging different minimum decoding units in sequence, separating the minimum decoding units according to the number of bits corresponding to a preset coding system to obtain corresponding decoding sub-characters, and obtaining corresponding microphone related information according to the decoding sub-characters.

Specifically, different preset coding systems correspond to different separation bit numbers, for example, different minimum decoding units are sequentially arranged to obtain "4E 2D 0030", and the preset coding system is hexadecimal, so that every 4 bits are separated to obtain two decoding sub-characters "4E 2D", "0030". Each decoded sub-character can determine the corresponding decoded character, and the information corresponding to the 4E2D is 'middle', the information corresponding to the 0030 is '3', and the weather-related information is 'middle 3'.

In one embodiment, step S432 includes: and acquiring a decoding table corresponding to a preset coding system, and acquiring a minimum decoding unit corresponding to the high-frequency and the low-frequency according to a table look-up mode.

Specifically, the size of the preset code system is agreed with the microphone or extracted from the audio signal, and different preset code systems have decoding algorithms corresponding to the code systems. By acquiring a decoding table of the relationship between the high-frequency and the low-frequency corresponding to the preset coding system and the minimum coding unit, the minimum decoding unit corresponding to the high-frequency and the low-frequency can be quickly acquired in a table look-up mode. As shown in table 3 below, a decoding table corresponding to the hexadecimal decoding method is shown, in which the horizontal axis represents low frequency, the vertical axis represents high frequency, and the characters in the table are the minimum decoding units corresponding to the hexadecimal decoding method.

TABLE 3

	20200	20400	20600	20800
					21000	0	1	2	3
21200	4	5	6	7
					21400	8	9	A	B
21600	C	D	E	F

For example, the high frequency 21200Hz and the low frequency 20200Hz correspond to the minimum decoding unit "4", and it can be understood that the specific values and the interval gaps of the high frequency and the low frequency in the table can be customized as required and are consistent with the encoding algorithm. In one embodiment, the table corresponding to the octal decoding method is shown in table 4:

TABLE 4

	20200	20550	20900	21250
					21600	0	1	2	3
21950	4	5	6	7

In one embodiment, step S420 includes: detecting whether the audio signal has the header identification information and the tail identification information in the preset format, and if so, acquiring data between the header identification information and the tail identification information to obtain a coding signal sequence in the audio signal.

Specifically, the coded signal sequence can be quickly extracted through the head identification information and the tail identification information in the preset format, the specific position or the specific time of the coded signal sequence does not need to be appointed with a microphone, and the method is simple and convenient.

In one embodiment, after step S430, the method further includes: and inserting the microphone related information into the service data or adjusting the corresponding playing parameters according to the microphone related information.

Specifically, the service data refers to data related to a service, such as audio sharing data, audio recommendation data, and the like, and by inserting the microphone related information into the service data, a user related to the service can quickly know the microphone related information, so that the information is quickly spread and popularized. If information can be added after the released works recorded by the microphone, the effect of rapidly popularizing information by a microphone hardware manufacturer is achieved by displaying that the work is recorded by the first type of microphone. The playing parameters include volume, reverberation depth, position of the clipper, etc., for example, if the microphone related information is "clipper 3-50" indicating that the 3 # clipper is pushed to 50% of the position, the corresponding position of the clipper can be adjusted according to the microphone related information. For example, the microphone related information is a "first type microphone", and since the first type microphone has a special sound effect, the playing sound effect of the playing terminal can be controlled to be closed during playing, so that the played sound quality effect is ensured.

In one embodiment, as shown in fig. 12, there is provided an apparatus for microphone data processing, including:

an obtaining module 510, configured to obtain information to be encoded related to the microphone.

And the encoding module 520 is configured to encode the information to be encoded into an encoded signal sequence formed by signals with different frequencies, where the frequency of the encoded signal sequence is greater than the upper limit frequency of human auditory perception.

The sending module 530 is configured to obtain an original audio signal collected by a microphone, superimpose the encoded signal sequence and the original audio signal to obtain an audio signal carrying microphone related information, and send the audio signal to the terminal, so that the terminal analyzes the audio signal carrying the microphone related information to obtain corresponding microphone related information.

In one embodiment, as shown in fig. 13, the encoding module 520 includes:

and the division encoding unit 521 is used for dividing and encoding the information to be encoded into encoding sub-characters corresponding to a preset encoding system, wherein the encoding sub-characters are sequentially formed by minimum encoding units.

The encoded signal sequence generating unit 522 is configured to sequentially obtain the high frequency and the low frequency corresponding to each minimum coding unit, generate the encoded sub-signals corresponding to the minimum coding unit according to the high frequency and the low frequency, and form an encoded signal sequence by the sequentially generated encoded sub-signals.

In an embodiment, the code signal sequence generating unit is further configured to obtain a code table corresponding to a preset code system, and obtain the high frequency and the low frequency corresponding to the minimum coding unit according to a table look-up manner.

In one embodiment, as shown in fig. 14, there is provided an apparatus for microphone data processing, including:

the receiving module 610 is configured to receive an audio signal sent by a microphone.

And the encoding signal sequence obtaining module 620 is configured to obtain an encoding signal sequence in the audio signal if the audio signal is an audio signal carrying microphone-related information, where the encoding signal sequence is formed by different frequency signals generated by encoding microphone-related information to be encoded, and a frequency of the encoding signal sequence is greater than an upper limit frequency of human auditory perception.

And the decoding module 630 is configured to parse the encoded signal sequence and decode the encoded signal sequence to obtain microphone related information.

In one embodiment, as shown in fig. 15, the decoding module 630 includes:

the minimum decoding unit acquiring unit 631 is configured to acquire the encoded sub-signal corresponding to the encoded signal sequence, acquire the high frequency and the low frequency of the encoded sub-signal, and acquire the corresponding minimum decoding unit according to the high frequency and the low frequency.

The microphone related information decoding unit 632 is configured to sequentially arrange different minimum decoding units and divide the minimum decoding units according to the number of bits corresponding to the preset coding system to obtain corresponding decoding sub-characters, and obtain corresponding microphone related information according to the decoding sub-characters.

In an embodiment, the minimum decoding unit obtaining unit 631 is further configured to obtain a decoding table corresponding to a preset encoding system, and obtain the minimum decoding units corresponding to the high frequency and the low frequency according to a table look-up manner.

In one embodiment, the encoded signal sequence obtaining module 620 is further configured to detect whether the audio signal has the header identification information and the tail identification information in the preset format, and if the audio signal has the header identification information and the tail identification information in the preset format, obtain the encoded signal sequence in the audio signal by obtaining data between the header identification information and the tail identification information.

In one embodiment, as shown in fig. 16, the apparatus further comprises:

the information utilization module 640 is configured to insert the microphone related information into service data or adjust corresponding playing parameters according to the microphone related information.

It will be understood by those skilled in the art that all or part of the processes in the methods of the embodiments described above may be implemented by hardware related to instructions of a computer program, which may be stored in a computer readable storage medium, for example, in the storage medium of a computer system, and executed by at least one processor in the computer system, so as to implement the processes of the embodiments including the methods described above. The storage medium may be a magnetic disk, an optical disk, a Read-Only Memory (ROM), a Random Access Memory (RAM), or the like.

The technical features of the embodiments described above may be arbitrarily combined, and for the sake of brevity, all possible combinations of the technical features in the embodiments described above are not described, but should be considered as being within the scope of the present specification as long as there is no contradiction between the combinations of the technical features.

The above-mentioned embodiments only express several embodiments of the present invention, and the description thereof is more specific and detailed, but not construed as limiting the scope of the invention. It should be noted that, for a person skilled in the art, several variations and modifications can be made without departing from the inventive concept, which falls within the scope of the present invention. Therefore, the protection scope of the present patent shall be subject to the appended claims.

Claims

1. A method of microphone data processing, the method comprising:

acquiring information to be coded related to a microphone;

encoding the information to be encoded into an encoded signal sequence formed by signals with different frequencies, wherein the frequency of the encoded signal sequence is greater than the upper limit frequency of human auditory perception, and the encoded signal sequence is obtained by encoding according to the ratio of the corresponding frequency to the sampling rate of sound;

acquiring an original audio signal acquired by a microphone, superposing the coding signal sequence and the original audio signal to obtain an audio signal carrying microphone related information, and sending the audio signal to a terminal so that the terminal can analyze the audio signal carrying the microphone related information to obtain corresponding microphone related information, wherein the coding signal sequence and the original audio signal are positioned at different positions of a frequency spectrum corresponding to the audio signal obtained by superposing the coding signal sequence and the original audio signal;

the original audio signal is an audio signal which is filtered to remove a signal exceeding a preset cut-off frequency, and the information to be coded is attribute information, operation state information or function information of a microphone, so that the terminal adjusts a corresponding playing parameter according to the microphone related information obtained through analysis, and the audio signal carrying the microphone related information or the original audio signal is played through the adjusted playing parameter.

2. The method of claim 1, wherein the step of encoding the information to be encoded into a coded signal sequence of signals of different frequencies comprises:

dividing and coding the information to be coded into coding sub-characters corresponding to a preset coding system, wherein the coding sub-characters are formed by minimum coding units in sequence;

and sequentially acquiring high-frequency and low-frequency corresponding to each minimum coding unit, generating coding sub-signals corresponding to the minimum coding units according to the high-frequency and the low-frequency, and forming a coding signal sequence by the sequentially generated coding sub-signals.

3. The method according to claim 2, wherein the step of sequentially obtaining the high frequency and the low frequency corresponding to each minimum coding unit comprises:

and acquiring a coding table corresponding to the preset coding system, and acquiring the high-frequency and the low-frequency corresponding to the minimum coding unit according to a table look-up mode.

4. The method according to claim 1, wherein the encoded signal sequence is generated periodically, and the encoded signal sequence carries header identification information and tail identification information in a preset format.

5. A method of microphone data processing, the method comprising:

receiving an audio signal sent by a microphone;

if the audio signal is an audio signal carrying microphone related information, acquiring a coding signal sequence in the audio signal, wherein the coding signal sequence is formed by different frequency signals generated by coding microphone related information to be coded, the frequency of the coding signal sequence is greater than the upper limit frequency of human ear auditory perception, the audio signal carrying microphone related information comprises an original audio signal collected by a microphone, and the coding signal sequence and the original audio signal are positioned at different positions of a frequency spectrum corresponding to the audio signal carrying microphone related information;

analyzing the coded signal sequence, decoding to obtain the microphone related information, and coding the coded signal sequence according to the ratio of the corresponding frequency to the sampling rate of the sound;

the original audio signal is an audio signal which is filtered to remove signals exceeding a preset cut-off frequency, the information to be coded is attribute information, operation state information or function information of a microphone, corresponding playing parameters are adjusted according to the microphone related information obtained through analysis, and the audio signal carrying the microphone related information or the original audio signal is played through the adjusted playing parameters.

6. The method of claim 5, wherein the step of parsing the encoded signal sequence and decoding to obtain the microphone related information comprises:

acquiring a coding sub-signal corresponding to the coding signal sequence;

acquiring the high-frequency and the low-frequency of the coding sub-signal;

acquiring a corresponding minimum decoding unit according to the high-frequency and the low-frequency;

arranging different minimum decoding units in sequence and separating the different minimum decoding units according to the number of bits corresponding to a preset coding system to obtain corresponding decoding sub-characters;

and obtaining corresponding microphone related information according to the decoded sub-characters.

7. The method of claim 6, wherein the step of obtaining the corresponding minimum decoding unit according to the high frequency and the low frequency comprises:

and acquiring a decoding table corresponding to the preset coding system, and acquiring a minimum decoding unit corresponding to the high-frequency and the low-frequency according to a table look-up mode.

8. The method according to claim 5, wherein the step of obtaining the sequence of encoded signals in the audio signal if the audio signal is an audio signal carrying microphone related information comprises:

detecting whether the audio signal has header identification information and tail identification information in a preset format, and if so, acquiring data between the header identification information and the tail identification information to obtain a coded signal sequence in the audio signal.

9. The method of claim 5, wherein after the step of parsing the encoded signal sequence and decoding to obtain the microphone related information, the method further comprises:

inserting the microphone related information into service data.

10. An apparatus for microphone data processing, the apparatus comprising:

the coding module is used for coding the information to be coded into a coding signal sequence formed by signals with different frequencies, the frequency of the coding signal sequence is greater than the upper limit frequency of human ear auditory perception, and the coding signal sequence is obtained by coding according to the ratio of the corresponding frequency to the sampling rate of sound;

the transmitting module is used for acquiring an original audio signal acquired by a microphone, superposing the coding signal sequence and the original audio signal to obtain an audio signal carrying microphone related information, and transmitting the audio signal carrying microphone related information to a terminal so that the terminal can analyze the audio signal carrying microphone related information to obtain corresponding microphone related information, wherein in a frequency spectrum corresponding to the audio signal obtained by superposing the coding signal sequence and the original audio signal, the coding signal sequence and the original audio signal are positioned at different positions of the frequency spectrum;

11. The apparatus of claim 10, wherein the encoding module comprises:

the segmentation coding unit is used for segmenting and coding the information to be coded into coding sub-characters corresponding to a preset coding system, and the coding sub-characters are sequentially formed by minimum coding units;

and the coding signal sequence generating unit is used for sequentially acquiring the high-frequency and the low-frequency corresponding to each minimum coding unit, generating the coding sub-signals corresponding to the minimum coding units according to the high-frequency and the low-frequency, and forming a coding signal sequence by the sequentially generated coding sub-signals.

12. The apparatus according to claim 11, wherein the encoded signal sequence generating unit is further configured to obtain an encoding table corresponding to the preset encoding system, and obtain the high frequency and the low frequency corresponding to the minimum encoding unit according to a table look-up manner.

13. The apparatus according to claim 10, wherein the encoded signal sequence is generated periodically, and the encoded signal sequence carries header identification information and tail identification information in a preset format.

14. An apparatus for microphone data processing, the apparatus comprising:

the encoding signal sequence acquiring module is configured to acquire an encoding signal sequence in the audio signal if the audio signal is an audio signal carrying microphone-related information, where the encoding signal sequence is formed by different frequency signals generated by encoding microphone-related information to be encoded, and the frequency of the encoding signal sequence is greater than an upper limit frequency of human ear auditory perception, the audio signal carrying microphone-related information includes an original audio signal acquired by the microphone, and the encoding signal sequence and the original audio signal are located at different positions of a frequency spectrum corresponding to the audio signal carrying microphone-related information;

the decoding module is used for analyzing the coding signal sequence, decoding to obtain the microphone related information, and coding the coding signal sequence according to the ratio of the corresponding frequency to the sampling rate of the sound;

the original audio signal is an audio signal which is filtered to remove a signal exceeding a preset cut-off frequency, the information to be coded is attribute information, operation state information or function information of a microphone, corresponding playing parameters are adjusted according to the microphone related information obtained through analysis, and the audio signal carrying the microphone related information or the original audio signal is played through the adjusted playing parameters.

15. The apparatus of claim 14, wherein the decoding module comprises:

a minimum decoding unit obtaining unit, configured to obtain a coded sub-signal corresponding to the coded signal sequence, obtain a high frequency and a low frequency of the coded sub-signal, and obtain a corresponding minimum decoding unit according to the high frequency and the low frequency;

and the microphone related information decoding unit is used for sequentially arranging different minimum decoding units and separating the minimum decoding units according to the bits corresponding to the preset coding system to obtain corresponding decoding sub-characters, and obtaining corresponding microphone related information according to the decoding sub-characters.

16. The apparatus of claim 15, wherein the minimum decoding unit obtaining unit is further configured to obtain a decoding table corresponding to the preset code system, and obtain the minimum decoding units corresponding to the high frequency and the low frequency according to a table look-up manner.

17. The apparatus according to claim 14, wherein the encoded signal sequence obtaining module is further configured to detect whether header identification information and tail identification information in a preset format exist in the audio signal, and if so, obtain data between the header identification information and the tail identification information to obtain the encoded signal sequence in the audio signal.

18. The apparatus of claim 14, further comprising:

and the information utilization module is used for inserting the microphone related information into service data.

19. Microphone, characterized by a storage medium and a processor, in which a computer program is stored which, when being executed by the processor, causes the processor to carry out the steps of the method of microphone data processing according to one of claims 1 to 4.

20. A computer-readable storage medium, characterized in that the storage medium has stored thereon a computer program which, when being executed by a processor, causes the processor to carry out the steps of the method of microphone data processing according to one of claims 1 to 4.

21. A terminal, characterized in that it comprises a storage medium and a processor, the storage medium having stored therein a computer program which, when executed by the processor, causes the processor to carry out the steps of the method of microphone data processing according to any of claims 5 to 9.

22. A computer-readable storage medium, characterized in that the storage medium has stored thereon a computer program which, when being executed by a processor, causes the processor to carry out the steps of the method of microphone data processing according to one of the claims 5 to 9.