CN114333864A - Audio data mixing method, terminal and computer readable storage medium - Google Patents
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Abstract
The invention provides an audio data mixing method, a terminal and a computer readable storage medium, wherein the audio data mixing method comprises the following steps: acquiring the same audio information respectively transmitted by at least two audio channels, wherein the audio information comprises a standard synchronous sound source; determining the time delay of the audio channels according to the time for receiving the standard synchronous sound source transmitted by the at least two audio channels respectively; and respectively determining the buffer length of the queue corresponding to the at least two audio channels according to the delay time length corresponding to the at least two audio channels. According to the method and the device, the delay time of different audio channels is detected, and the buffer length of the queues of other audio channels is determined based on the delay time of all the audio channels, so that the audio mixing module can receive audio data synchronously transmitted by different audio channels, and the audio mixing effect is improved.
Description
Technical Field
The present invention relates to the field of audio data processing technologies, and in particular, to an audio data mixing method, a terminal, and a computer-readable storage medium.
Background
Voice mixing is an important component in multimedia conferences. Because the audio sources of the audio mixing are from different devices and through different transmission paths, the time when the audio data collected by each path really reaches the audio mixing module and the time when the real world sound is generated have a delay. The delay between each path may be greatly different, especially the audio data transmitted through the network has more processing procedures such as encoding, network transmission, decoding and the like than the analog audio acquisition, and the delay is obviously higher than that of the analog audio source. Different audio acquisition devices may have different delay performances due to internal processing flow and network fluctuation. If the delay difference of each audio is not processed and the audio is directly sent to the audio mixing module, the audio data after audio mixing may have the problem of sound superposition, which seriously affects the audio mixing effect.
Disclosure of Invention
The technical problem mainly solved by the invention is to provide an audio data mixing method, a terminal and a computer readable storage medium, which solve the problem of poor mixing effect caused by asynchronous data transmission time of each audio channel in the prior art.
In order to solve the technical problems, the first technical scheme adopted by the invention is as follows: there is provided an audio data mixing method including: acquiring the same audio information respectively transmitted by at least two audio channels, wherein the audio information comprises a standard synchronous sound source; determining the time delay of the audio channels according to the time for receiving the standard synchronous sound sources respectively transmitted through at least two audio channels; and respectively determining the buffer length of the queue corresponding to the at least two audio channels according to the delay time length corresponding to the at least two audio channels.
Wherein, according to the time of receiving the standard synchronization sound source respectively transmitted through at least two audio channels, determining the time delay duration of the audio channels comprises: identifying audio information transmitted by an audio channel and determining the time for receiving a standard synchronous sound source; and determining the time delay of the audio channel according to the time for receiving the standard synchronous sound source.
Wherein, the audio information transmitted by the audio channel is identified, and the time for receiving the standard synchronous sound source is determined, comprising: caching an audio sample with preset duration; converting the audio samples into an audio spectrum; judging whether the similarity of the audio frequency spectrum and a preset frequency spectrum of a standard synchronous sound source exceeds the preset similarity or not; if the similarity of the audio frequency spectrum and the preset frequency spectrum of the standard synchronous sound source exceeds the preset similarity, determining the audio frequency sample as the standard synchronous sound source; the time at which the audio sample is received is determined as the time at which the standard synchronization source is received.
Wherein, the audio sample of duration is predetermine in the buffer memory, still includes before: acquiring audio sample points in the audio information; judging whether the amplitude of the audio sample point exceeds a threshold value; if the amplitude of the audio sample point exceeds the threshold, buffering of the audio sample begins.
Wherein converting the audio samples into an audio spectrum comprises: the audio samples are converted to an audio spectrum by a fast fourier transform.
Wherein, according to the time of receiving the standard synchronous sound source, determining the time delay of the audio channel comprises: the moment of playing the standard synchronous sound source is the first time; the moment of receiving the standard synchronous sound source transmitted by the audio channel is the second time; the difference value between the second time and the first time is the delay time length of the audio channel.
Wherein, the difference between the second time and the first time is the delay duration of the audio channel, and then comprises: judging whether the frequency of playing the standard synchronous sound source exceeds a preset frequency or not; and if the frequency of playing the standard synchronous sound source exceeds the preset frequency, determining the average delay time of the audio channel according to a plurality of delay time obtained by the audio channel.
Wherein, according to the delay duration corresponding to at least two audio channels, respectively determining the buffer length of the queue corresponding to at least two audio channels, comprises: selecting the maximum delay time length in the delay time lengths corresponding to at least two audio channels; and respectively calculating the difference between the maximum delay time length and other delay time lengths except the maximum delay time length in the delay time lengths corresponding to the at least two audio channels so as to determine the buffer length of the queue corresponding to each audio channel.
The method for determining the buffer length of the queue corresponding to each audio channel includes the following steps: calculating to obtain a difference value between the delay time length corresponding to the audio channel and the maximum delay time length; and determining the buffer lengths of the queues respectively corresponding to the audio channels according to the difference value and the sampling rate of the acquired audio information.
In order to solve the above technical problems, the second technical solution adopted by the present invention is: there is provided a terminal comprising a memory, a processor and a computer program stored in the memory and running on the processor, the processor being configured to execute the sequence data to implement the steps in the audio data mixing method described above.
In order to solve the above technical problems, the third technical solution adopted by the present invention is: there is provided a computer-readable storage medium having stored thereon a computer program which, when executed by a processor, implements the steps in the above-described audio data mixing method.
The invention has the beneficial effects that: different from the prior art, the audio data mixing method, the terminal and the computer readable storage medium are provided, and the audio data mixing method comprises the following steps: acquiring the same audio information respectively transmitted by at least two audio channels, wherein the audio information comprises a standard synchronous sound source; determining the time delay of the audio channels according to the time for receiving the standard synchronous sound sources respectively transmitted by the at least two audio channels; and respectively determining the buffer length of the queue corresponding to the at least two audio channels according to the delay time length corresponding to the at least two audio channels. According to the method and the device, the delay time of different audio channels is detected, and the buffer length of the queue corresponding to other audio channels is determined based on the delay time of all the audio channels, so that the audio mixing module can receive audio data synchronously transmitted by different audio channels, and the audio mixing effect is improved.
Drawings
Fig. 1 is a schematic flow chart of an audio data mixing method provided by the present invention;
FIG. 2 is a flowchart illustrating an audio data mixing method according to an embodiment of the present invention;
FIG. 3 is a schematic block diagram of one embodiment of a terminal provided by the present invention;
FIG. 4 is a schematic block diagram of one embodiment of a computer-readable storage medium provided by the present invention.
Detailed Description
The following describes in detail the embodiments of the present application with reference to the drawings attached hereto.
In the following description, for purposes of explanation and not limitation, specific details are set forth such as particular system structures, interfaces, techniques, etc. in order to provide a thorough understanding of the present application.
The technical solutions in the embodiments of the present application will be clearly and completely described below with reference to the drawings in the embodiments of the present application, and it is obvious that the described embodiments are only a part of the embodiments of the present application, and not all of the embodiments. All other embodiments, which can be derived by a person skilled in the art from the embodiments given herein without making any creative effort, shall fall within the protection scope of the present application.
The terms "first", "second" and "third" in this application are used for descriptive purposes only and are not to be construed as indicating or implying relative importance or implying any indication of the number of technical features indicated. Thus, a feature defined as "first," "second," or "third" may explicitly or implicitly include at least one of the feature. In the description of the present application, "plurality" means at least two, e.g., two, three, etc., unless explicitly specifically limited otherwise. All directional indications (such as up, down, left, right, front, and rear … …) in the embodiments of the present application are only used to explain the relative positional relationship between the components, the movement, and the like in a specific posture (as shown in the drawings), and if the specific posture is changed, the directional indication is changed accordingly. Furthermore, the terms "include" and "have," as well as any variations thereof, are intended to cover non-exclusive inclusions. For example, a process, method, system, article, or apparatus that comprises a list of steps or elements is not limited to only those steps or elements listed, but may alternatively include other steps or elements not listed, or inherent to such process, method, article, or apparatus.
Reference herein to "an embodiment" means that a particular feature, structure, or characteristic described in connection with the embodiment can be included in at least one embodiment of the application. The appearances of the phrase in various places in the specification are not necessarily all referring to the same embodiment, nor are separate or alternative embodiments mutually exclusive of other embodiments. It is explicitly and implicitly understood by one skilled in the art that the embodiments described herein can be combined with other embodiments.
Referring to fig. 1, fig. 1 is a flow chart illustrating an audio data mixing method according to the present invention. The embodiment provides an audio data mixing method, which can be applied to recorded broadcast products in the education industry and can also be applied to industries such as conference recorded broadcast products and the like. The audio data mixing method includes the following steps.
S11: and acquiring the same audio information respectively transmitted by at least two audio channels, wherein the audio information comprises a standard synchronous sound source.
Specifically, a standard synchronous sound source played by the recording and broadcasting host is collected through at least two audio channels, and the audio channels collecting the standard synchronous sound source transmit audio data to a sound mixing module in the recording and broadcasting host. Wherein, the audio channel is with audio data transmission to the transmission methods such as recorded broadcast host computer's audio mixing module including wired transmission, bluetooth transmission, WIFI transmission, transmission such as Zigbee transmission, because different audio data can be through the audio mixing module of transmission methods in with the standard synchronization sound source transmission to the recorded broadcast host computer of gathering of different transmission methods, different transmission methods' transmission speed is different, and in the transmission process of standard synchronization sound source, encode to standard synchronization sound source, the mode of decoding is also different, make the time of different audio channel transmission to audio mixing module appear the time delay, make different audio channel with the standard synchronization sound source of gathering can not synchronous transmission to audio mixing module.
S12: and determining the time delay of the audio channels according to the time for receiving the standard synchronous sound sources respectively transmitted by the at least two audio channels.
Specifically, identifying audio information transmitted by an audio channel, and determining the time for receiving a standard synchronous sound source; and determining the time delay of the audio channel according to the time for receiving the standard synchronous sound source. In a specific embodiment, audio samples of a preset duration are cached; converting the audio samples into an audio spectrum; judging whether the similarity of the audio frequency spectrum and a preset frequency spectrum of a standard synchronous sound source exceeds the preset similarity or not; if the similarity of the audio frequency spectrum and the preset frequency spectrum exceeds the preset similarity, determining the audio frequency sample as a standard synchronous sound source; the time at which the audio sample is received is determined as the time at which the standard synchronization source is received. Wherein the audio samples are converted to an audio spectrum by a fast fourier transform. In an alternative embodiment, audio sample points in the audio information are obtained; judging whether the amplitude of the audio sample point exceeds a threshold value; if the amplitude of the audio sample point exceeds the threshold, buffering of the audio sample begins.
In one embodiment, the time of transmitting the standard synchronized sound source is a first time; the moment of receiving the standard synchronous sound source transmitted by the audio channel is the second time; the difference value between the second time and the first time is the delay time length of the audio channel.
In one embodiment, whether the number of times of playing the standard synchronous sound source exceeds a preset number of times is judged; and if the frequency of playing the standard synchronous sound source exceeds the preset frequency, determining the average delay time of the audio channel according to a plurality of delay time obtained by the audio channel.
S13: and respectively determining the buffer length of the queue corresponding to the at least two audio channels according to the delay time length corresponding to the at least two audio channels.
Specifically, selecting the maximum delay time length from the delay time lengths corresponding to at least two audio channels; and respectively calculating the difference between the maximum delay time length and other delay time lengths except the maximum delay time length in the delay time lengths corresponding to the at least two audio channels so as to determine the buffer length of the queue corresponding to each audio channel.
The embodiment provides an audio data mixing method, which includes acquiring the same audio information respectively transmitted by at least two audio channels, wherein the audio information includes a standard synchronous sound source; determining the time delay of the audio channels according to the time for receiving the standard synchronous sound sources respectively transmitted by the at least two audio channels; and respectively determining the buffer length of the queue corresponding to the at least two audio channels according to the delay time length corresponding to the at least two audio channels. According to the method and the device, the delay time of different audio channels is detected, and the buffer length of the queue corresponding to other audio channels is determined based on the delay time of all the audio channels, so that the audio mixing module can receive audio data synchronously transmitted by different audio channels, and the audio mixing effect is improved.
Referring to fig. 2, fig. 2 is a flowchart illustrating an audio data mixing method according to an embodiment of the present invention. The embodiment provides an audio data mixing method, which can be applied to recorded broadcast products in the education industry and can also be applied to industries such as conference recorded broadcast products and the like. The audio data mixing method includes the following steps.
S201: at least two audio channels simultaneously acquire the same standard synchronous sound source.
Specifically, the recording and broadcasting host sends out a standard synchronous sound source, and the time for the recording and broadcasting host to send out the standard synchronous sound source is recorded as T0. And simultaneously acquiring a standard synchronous sound source emitted by the recording and broadcasting host through at least two audio channels. The number of the audio channels may be two or more. The audio channel may be a sound pickup, an analog high-definition Camera, an internet protocol Camera (IP Camera, IPC), and/or an internet dome Camera, or may be other audio acquisition devices, which is not limited herein. The standard synchronous sound source is buzzing with fixed frequency and fixed length, and can also be other audio data. When the recording and broadcasting host computer sends the standard synchronous sound source, the environment needs to be kept quiet, and the standard synchronous sound source needs to be played at a large volume, so that the collected audio information can be conveniently identified subsequently.
S202: audio sample points in the audio information are obtained.
Specifically, when receiving audio information sent by an audio channel, the recording and playing host acquires audio sample points in the audio information in real time.
S203: it is determined whether the amplitude of the audio sample point exceeds a threshold.
Specifically, the audio information is screened first to filter the audio before the standard synchronous sound source, so that the workload of converting the audio sample into an audio frequency spectrum subsequently is reduced. And comparing the audio sample points in the acquired audio information with a threshold value to further judge whether the audio sample points exceed the threshold value. Wherein the threshold is determined according to the amplitude of the standard synchronous sound source to distinguish the standard synchronous sound source from noise.
If the audio sample point in the audio information exceeds the threshold value, directly jumping to the step S204; if the audio sample point in the audio information does not exceed the threshold, the process directly jumps to step S202.
S204: the buffering of the audio samples is started.
If the amplitude of the audio sample point exceeds the threshold, buffering of the audio sample begins. Specifically, the recording and broadcasting host receives audio information transmitted by the audio channel, and the recording and broadcasting host caches audio samples with preset duration in real time. Wherein the audio samples may comprise at least part of a standard synchronized audio source.
S205: the audio samples are converted to an audio spectrum.
Specifically, in order to compare the collected audio sample with a standard synchronous sound source, the audio sample is more conveniently identified, and the audio sample is converted into an audio frequency spectrum through fast Fourier transform.
S206: and judging whether the similarity of the audio frequency spectrum and a preset frequency spectrum of the standard synchronous sound source exceeds the preset similarity or not.
Specifically, the audio frequency spectrum converted from the audio sample is compared with a preset frequency spectrum of the standard synchronous sound source, and then whether the acquired audio sample is the standard synchronous sound source or not is judged, or whether the acquired audio sample comprises the standard synchronous sound source or not is judged. In one embodiment, the audio sample is converted by comparing the amplitude at the frequency point of the audio spectrum with the amplitude at the frequency point of the predetermined spectrum of the standard synchronized sound source. The comparison can also be performed by other mathematical methods, which are not limited herein, as long as the comparison between the audio frequency spectrum and the preset frequency spectrum can be achieved.
If the similarity of the audio frequency spectrum and the preset frequency spectrum of the standard synchronous sound source exceeds the preset similarity, directly jumping to the step S207; if the similarity between the audio frequency spectrum and the preset frequency spectrum of the standard synchronous sound source does not exceed the preset similarity, the step S202 is directly skipped.
S207: determining an audio sample as a standard synchronous sound source; and determines the time at which the audio sample is received as the time at which the standard synchronization source is received.
Specifically, if the similarity of the audio frequency spectrum and the preset frequency spectrum of the standard synchronous sound source exceeds the preset similarity, the audio frequency sample is indicated as the standard synchronous sound source; and determining the time of reception of the received audio samples as the time T of reception of the standard synchronization tone source1And further, the time for transmitting the collected standard synchronous sound source to the sound mixing module by the audio channel is obtained.
S208: and determining the time delay of the audio channel according to the time of sending the standard synchronous sound source by the received audio channel.
Specifically, the time when the recording and broadcasting host sends the standard synchronous sound source is the first time and is recorded as T0(ii) a The moment when the standard synchronous sound source is received is the second time and is recorded as T1(ii) a Determining the delay time length of the audio channel as DT according to the difference value between the second time and the first time, namely DT-T1-T0。
S209: and judging whether the frequency of playing the standard synchronous sound source exceeds a preset frequency.
Specifically, in order to improve the accuracy of detecting the delay time when the audio channel transmits the audio information, the recording and broadcasting host needs to send out the standard synchronous sound source for multiple times, that is, the above steps S201 to S208 are repeated, and multiple delay times need to be obtained for the same audio channel. And accumulating the times of repeating the steps, and judging whether the times of repeating the steps exceed the preset times.
If the number of times of repeating the steps exceeds the preset number, directly jumping to the step S210; if the number of times of repeating the above steps does not exceed the preset number of times, directly jumping to step S201.
S210: and determining the average delay time of the audio channel according to the plurality of delay time obtained by the audio channel.
Specifically, in order to improve the stability of the time delay duration of audio information transmission in the audio channel, if the number of times of sending the standard synchronization sound source exceeds the preset number of times, the average time delay duration of the audio channel is determined according to a plurality of time delay durations obtained by the audio channel. Namely, the average delay time length of a plurality of delay time lengths corresponding to the same audio channel is obtained through calculation.
At this time, the average delay time corresponding to different audio channels is obtained.
S211: and selecting the maximum delay time length in the delay time lengths corresponding to the at least two audio channels.
Specifically, the average delay time lengths corresponding to different audio channels are obtained through the steps, the average delay time lengths corresponding to the different audio channels are sorted from large to small, and the maximum average delay time length is selected as the maximum delay time length corresponding to the multiple audio channels. For example, the maximum delay time duration of each audio channel is ordered as follows: DavgT0(channel 0), DavgT1(channel 1), …, DavgTn(channel n), taking the maximum channel delay time: DTmax＝Max{DavgT0，DavgT1，…，DavgTnAnd recording the corresponding audio channel number as m.
S212: and respectively calculating the difference between the maximum delay time length and other delay time lengths except the maximum delay time length in the delay time lengths corresponding to the at least two audio channels so as to determine the buffer length of the queue corresponding to each audio channel.
Specifically, the buffer lengths of the queues respectively corresponding to the other audio channels are adjusted according to the maximum delay time. I.e. such that the delay time duration of the other audio channels needs to be consistent with the delay time duration of audio channel m. Respectively calculating the difference T between the corresponding delay time and the maximum delay time of the audio channeli＝DTmax-DavgTiWhere i is audioThe channel number. Wherein the buffer length of the queue represents a time length of the audio samples that need to be buffered in the audio channel. In this embodiment, the audio sample may be blank data, i.e., air-sound data.
Determining the buffer length L of the queue of the audio channel according to the difference value between the corresponding delay time length and the maximum delay time length of the audio channel and the sampling rate of the obtained audio informationiI.e. Li＝TiF. Wherein L isiFor the number of audio sample points to be buffered, LiAnd taking a positive integer, wherein i is the number of the audio channel, and F is the sampling rate of the audio channel for acquiring the audio information.
The embodiment provides an audio data mixing method, which includes acquiring the same audio information respectively transmitted by at least two audio channels, wherein the audio information includes a standard synchronous sound source; determining the time delay of the audio channels according to the time for receiving the standard synchronous sound sources respectively transmitted by the at least two audio channels; and respectively determining the buffer length of the queue corresponding to the at least two audio channels according to the delay time length corresponding to the at least two audio channels. According to the method and the device, the delay time of different audio channels is detected, and the buffer length of the queue corresponding to other audio channels is determined based on the delay time of all the audio channels, so that the audio mixing module can receive audio data synchronously transmitted by different audio channels, and the audio mixing effect is improved.
Referring to fig. 3, fig. 3 is a schematic block diagram of an embodiment of a terminal provided in the present invention. The terminal 70 in this embodiment includes: the processor 71, the memory 72, and a computer program stored in the memory 72 and capable of running on the processor 71 are not repeated herein to avoid repetition in the above audio data mixing method when the computer program is executed by the processor 71.
Referring to fig. 4, fig. 4 is a schematic block diagram of an embodiment of a computer-readable storage medium provided by the present invention.
A computer-readable storage medium 90 is further provided in the embodiments of the present application, the computer-readable storage medium 90 stores a computer program 901, the computer program 901 includes program instructions, and a processor executes the program instructions to implement the audio data mixing method provided in the embodiments of the present application.
The computer-readable storage medium 90 may be an internal storage unit of the computer device of the foregoing embodiment, such as a hard disk or a memory of the computer device. The computer-readable storage medium 90 may also be an external storage device of the computer device, such as a plug-in hard disk provided on the computer device, a Smart Media Card (SMC), a Secure Digital (SD) Card, a Flash memory Card (Flash Card), and the like.
The above description is only an embodiment of the present invention, and not intended to limit the scope of the present invention, and all modifications of equivalent structures and equivalent processes performed by the present specification and drawings, or directly or indirectly applied to other related technical fields, are included in the scope of the present invention.
Claims (11)
1. An audio data mixing method, characterized in that the audio data mixing method comprises:
acquiring the same audio information respectively transmitted by at least two audio channels, wherein the audio information comprises a standard synchronous sound source;
determining the time delay duration of the audio channels according to the time for receiving the standard synchronous sound source respectively transmitted through the at least two audio channels;
and respectively determining the buffer length of the queue corresponding to the at least two audio channels according to the delay time length corresponding to the at least two audio channels.
2. The audio data mixing method according to claim 1,
the determining the delay duration of the audio channel according to the time of receiving the standard synchronization sound source respectively transmitted through the at least two audio channels includes:
identifying the audio information transmitted by the audio channel and determining the time for receiving the standard synchronous sound source;
and determining the time delay of the audio channel according to the time for receiving the standard synchronous sound source.
3. The audio data mixing method according to claim 2,
said identifying said audio information transmitted by said audio channel and determining a time at which said standard synchronization source is received comprises:
caching an audio sample with preset duration;
converting the audio samples into an audio spectrum;
judging whether the similarity of the audio frequency spectrum and a preset frequency spectrum of the standard synchronous sound source exceeds a preset similarity or not;
if the similarity of the audio frequency spectrum and a preset frequency spectrum of the standard synchronous sound source exceeds the preset similarity, determining that the audio frequency sample is the standard synchronous sound source;
determining a time at which the audio sample is received as a time at which the standard synchronization source is received.
4. The audio data mixing method according to claim 3,
the caching of the audio samples with the preset duration further comprises the following steps:
acquiring audio sample points in the audio information;
judging whether the amplitude of the audio sample point exceeds a threshold value;
if the magnitude of the audio sample point exceeds the threshold, beginning buffering the audio sample.
5. The audio data mixing method according to claim 3,
the converting the audio samples into an audio spectrum comprises:
the audio samples are converted to an audio spectrum by a fast fourier transform.
6. The audio data mixing method according to claim 2,
the determining the delay time duration of the audio channel according to the time for receiving the standard synchronous sound source includes:
the moment of playing the standard synchronous sound source is the first time;
the moment of receiving the standard synchronous sound source transmitted by the audio channel is a second time;
and the difference value of the second time and the first time is the delay time of the audio channel.
7. The audio data mixing method according to claim 6,
the difference between the second time and the first time is the delay duration of the audio channel, and then the method comprises the following steps:
judging whether the frequency of playing the standard synchronous sound source exceeds a preset frequency or not;
and if the frequency of playing the standard synchronous sound source exceeds the preset frequency, determining the average delay time of the audio channel according to the plurality of delay time obtained by the audio channel.
8. The audio data mixing method according to claim 1,
the determining, according to the delay durations corresponding to the at least two audio channels, the buffer lengths of the queues corresponding to the at least two audio channels respectively includes:
selecting the maximum delay time length in the delay time lengths corresponding to the at least two audio channels;
and respectively calculating the difference between the other delay time lengths except the maximum delay time length in the delay time lengths corresponding to the at least two audio channels and the maximum delay time length so as to determine the buffer length of the queue corresponding to each audio channel.
9. The audio data mixing method according to claim 8,
the calculating the difference between the maximum delay time length and the other delay time lengths of the delay time lengths corresponding to the at least two audio channels except the maximum delay time length respectively to determine the buffer length of the queue corresponding to each audio channel respectively includes:
calculating to obtain a difference value between the delay time corresponding to the audio channel and the maximum delay time;
and determining the buffer lengths of the queues respectively corresponding to the audio channels according to the difference and the sampling rate of the acquired audio information.
10. A terminal, characterized in that the terminal comprises a memory, a processor and a computer program stored in the memory and run on the processor, the processor being configured to execute sequential data to implement the steps in the audio data mixing method according to any of claims 1-9.
11. A computer-readable storage medium, having stored thereon a computer program which, when being executed by a processor, carries out the steps of the audio data mixing method according to any one of claims 1 to 9.
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