CN105872253B - A kind of live sound processing method and mobile terminal - Google Patents

Abstract

The embodiment of the invention discloses a live broadcast sound processing method and a mobile terminal, wherein the live broadcast sound processing method comprises the following steps: acquiring an audio file selected by a user; decoding the audio file to obtain an audio signal; collecting voice signals of the user; the audio signal and the voice signal are mixed and played, and the embodiment of the invention can enrich the live broadcast atmosphere without other external equipment and can improve the quality of the final output signal.

Description

A kind of live sound processing method and mobile terminal

technical field

Embodiments of the present invention relate to the field of communication technologies, and in particular, to a method for processing live sound and a mobile terminal.

Background technique

In recent years, with the continuous development of mobile terminal technologies such as mobile phones, mobile terminals have become more and more intelligent, and mobile terminals have provided more and more functions, such as live broadcast functions. Features are becoming more and more popular with users.

In order to enrich the live broadcast atmosphere, it is sometimes necessary to add some background music during the live broadcast. In the prior art, in order to achieve this purpose, other external playback devices (such as personal computers) are usually required. The specific implementation process is as follows: The speaker of the mobile terminal plays the background music, and the microphone of the mobile terminal simultaneously collects the background music and the voice of the host, and then mixes the two for playback. Since it needs to use other external devices to obtain background music, this method is very inconvenient to implement, and the implementation location is easily limited. Moreover, the background music is collected through the microphone at the same time, and the radio effect is relatively poor. Signal quality is poor.

SUMMARY OF THE INVENTION

In view of this, the embodiments of the present invention provide a live broadcast sound processing method and a mobile terminal, which can enrich the live broadcast atmosphere without other external playback devices, and can improve the quality of the final output signal.

The live sound processing method provided by the embodiment of the present invention includes:

Get the audio file selected by the user;

Decode the audio file to obtain an audio signal;

collecting the voice signal of the user;

The audio signal and the speech signal are mixed and played.

The mobile terminal provided by the embodiment of the present invention includes:

an acquisition unit for acquiring the audio file selected by the user;

a decoding unit for decoding the audio file to obtain an audio signal;

a collection unit, used for collecting the voice signal of the user;

A sound mixing and playing unit, configured to mix and play the audio signal and the voice signal.

In this embodiment of the present invention, the mobile terminal can acquire the audio file selected by the user, and decode the audio file to obtain the audio signal. Therefore, it is no longer necessary to play the audio signal through other external playback devices, and it is no longer necessary to use the microphone of the mobile terminal to collect the audio signal. After obtaining the audio signal, the microphone of the mobile terminal can be adjusted to focus on collecting the user's voice signal, and then the obtained audio signal and the collected user's voice signal are mixed and played, thereby improving the quality of the final output signal.

Description of drawings

In order to illustrate the technical solutions in the embodiments of the present invention more clearly, the following briefly introduces the accompanying drawings used in the description of the embodiments. Obviously, the accompanying drawings in the following description are only some embodiments of the present invention. For those skilled in the art, other drawings can also be obtained from these drawings without creative effort.

Fig. 1 is a scene schematic diagram of a live broadcast sound processing method provided by an embodiment of the present invention;

2 is a schematic flowchart of a method for processing live sound provided by an embodiment of the present invention;

Fig. 3a is another schematic flow chart of the live sound processing method provided by the embodiment of the present invention;

Fig. 3b is a schematic flow chart of the method for decoding audio file of the present invention;

4a to 4d are the effect display diagrams of the background music provided by the embodiments of the present invention;

5a to 5d are diagrams showing effects of sound effects provided by embodiments of the present invention;

6 is a schematic structural diagram of a mobile terminal provided by an embodiment of the present invention;

FIG. 7 is another schematic structural diagram of a mobile terminal provided by an embodiment of the present invention.

Detailed ways

The technical solutions in the embodiments of the present invention will be clearly and completely described below with reference to the accompanying drawings in the embodiments of the present invention. Obviously, the described embodiments are only a part of the embodiments of the present invention, but not all of the embodiments. Based on the embodiments of the present invention, all other embodiments obtained by those skilled in the art without creative efforts shall fall within the protection scope of the present invention.

In the prior art, when other audio signals are to be added during the live broadcast to enrich the live broadcast atmosphere, other external playback devices are required, and the quality of the final output signal is not ideal. Therefore, the embodiment of the present invention provides a live broadcast The sound processing method and the mobile terminal can enrich the live broadcast atmosphere without the aid of other external playback devices, and can improve the quality of the final output signal. The mobile terminal in the embodiment of the present invention may specifically be a mobile phone, a tablet computer, a notebook computer, and the like. As shown in FIG. 1 , the mobile terminal can acquire the audio file selected by the user, decode the audio file to obtain an audio signal, and the audio signal is used to enrich the live broadcast atmosphere, and then mix the collected user's voice signal with the audio signal. The audio signal is obtained by decoding the audio file in the whole process, so the live broadcast atmosphere can be enriched without other external playback devices, and the quality of the final output signal can be improved.

Detailed descriptions will be given below, and the order of description of each embodiment below does not constitute a limitation on the order of specific implementation.

Example 1

As shown in FIG. 2 , the live sound processing method of this embodiment includes the following steps:

Step 201, obtain the audio file selected by the user;

In the specific implementation, the user can enter the live broadcast interface through the live broadcast software on the mobile terminal, click the atmosphere manager of the live broadcast interface to input the identifier of the selected audio file (for example, the name of the audio file) at the corresponding position, and the mobile terminal can input the audio file according to the user input. Get the audio file selected by the user.

Step 202, decoding the audio file to obtain an audio signal;

The audio signal is used to enrich the live broadcast atmosphere during the live broadcast.

Step 203, collecting the voice signal of the user;

In specific implementation, the microphone of the mobile terminal can be used to collect the user's voice signal. In this embodiment, since the microphone only needs to collect the user's voice signal and does not need to collect other audio signals, the position of the microphone can be adjusted so that it is aimed at the user, Or adjust the mode of the microphone (for example, the noise reduction mode) to focus on collecting the user's voice signal, so as to ensure that the high-quality voice signal is collected.

Step 204: Mix and play the audio signal and the voice signal.

In a specific implementation, steps 201 and 202 may be performed simultaneously with step 203, that is, the audio signal and the voice signal may be acquired synchronously. The audio signal that is about to be mixed is superimposed with the user's voice signal to prevent signal overflow. In this embodiment, the audio signal and the voice signal can be mixed in real time.

In this embodiment, the mobile terminal can acquire the audio file selected by the user, decode the audio file to obtain the audio signal, so it is no longer necessary to play the audio signal through other external playback devices, and it is no longer necessary to use the microphone of the mobile terminal to collect the audio signal, After acquiring the audio signal, the microphone of the mobile terminal can be adjusted to focus on collecting the user's voice signal, and then the acquired audio signal and the collected user's voice signal are mixed and played, thereby improving the quality of the final output signal.

Embodiment 2

The method described in the first embodiment will be further described in this embodiment. As shown in FIG. 3a, the method for processing live sound in this embodiment includes the following steps:

Step 301, obtain the audio file selected by the user;

In the specific implementation, the user can enter the live broadcast interface through the live broadcast software on the mobile terminal, click the atmosphere manager of the live broadcast interface to input the identifier of the selected audio file (for example, the name of the audio file) at the corresponding position, and the mobile terminal can input the audio file according to the user input. Get the audio file selected by the user.

The mobile terminal can acquire the audio file from the server by connecting to the network according to the user's input, and if the audio file is stored in the local storage of the mobile terminal, the mobile terminal can also acquire the audio file from its own local storage to realize the audio file. quick access.

The user-selected audio file may be a background music (eg, certain songs, light music, etc.) file, or the user-selected audio file may be a sound effect (eg, beep, clapping, jumping, etc.) file.

Step 302, decoding the audio file to obtain an audio signal;

Specifically, the decoding method can be as shown in Figure 3b, including the following steps:

Step 3021, parse the audio file to obtain the sampling parameters of the audio file;

Wherein, the sampling parameters of the audio file include sampling rate, sampling precision and channel mode, and these parameters are the main parameters for decoding the audio file.

The adoption rate, also known as sampling speed or sampling frequency, defines the number of samples per unit time, which is expressed in Hertz (Hz). For sound signals, the sampling rate can be used to describe the sound quality and pitch of the sound file, which is a measure of Quality standards for sound cards and sound files. The higher the sampling rate, that is, the shorter the sampling interval, the more sample data can be obtained per unit time, and the more accurate the representation of the signal waveform. There is a certain relationship between the sampling rate and the frequency of the original signal. According to Nyquist theory, only when the sampling rate is higher than twice the highest frequency of the original signal can the signal represented by the digital signal be restored to the original signal.

Sampling accuracy, which reflects the accuracy of measuring the amplitude of the sound waveform, in bits (Bit), such as 8 bits, 16 bits. 8 bits can divide a sound wave into a signal of 256 levels, and 16 bits can divide the same wave into a signal of 65,536 levels. For example, when the sound sample is represented by 16 bits (2 bytes), the measured sound sample value is in the range of 0～65535, and its precision is 1/65536 of the input signal. The size of the sample bits affects the quality of the sound. The more bits, the higher the quality of the sound, and the more storage space required; the fewer the bits, the lower the quality of the sound, and the less storage space required.

Channel mode, divided into single channel or dual channel mode, the value of single channel is 1, and the value of dual channel is 2.

Step 3022, calculate the coding rate of the audio file;

The code rate is also known as the bit rate. The code rate = sampling rate * sampling accuracy * channel mode. The higher the code rate, the larger the transmitted data and the better the sound quality.

Step 3023: Decode the audio file according to the encoding bit rate to obtain the audio signal.

Decoding is the opposite process to encoding, and the audio signal can be obtained by decoding the audio file according to the encoding bit rate.

In a specific implementation, the audio file can be decoded into a signal in a pulse-code modulation (Pulse-code modulation, PCM) format, or the audio file can be decoded into a signal in a waveform (wave, WAV) format, which is not specifically limited here. The background music signal can be obtained by decoding the background music file, the sound effect signal can be obtained by decoding the sound effect file, and the background music signal or sound effect signal obtained by decoding is used to enrich the live broadcast atmosphere during the live broadcast.

Step 303, collecting the voice signal of the user;

In this embodiment, before collecting the user's voice signal, the audio file selected by the user may be obtained and decoded in advance, and after decoding, the audio file may be stored in the storage, waiting for the trigger of the user's voice signal. When the user's voice signal is collected, the audio signal is directly obtained from the buffer for mixing, so as to improve the mixing speed.

In specific implementation, the microphone of the mobile terminal can be used to collect the user's voice signal. In this embodiment, since the microphone only needs to collect the user's voice signal and does not need to collect other audio signals, the position of the microphone can be adjusted so that it is aimed at the user, Or adjust the mode of the microphone (for example, the noise reduction mode) to focus on collecting the user's voice signal, so as to ensure that the high-quality voice signal is collected.

In this embodiment, the sampling parameters for the microphone to collect the user's voice signal can be set according to the sampling parameters of the audio file obtained in step 3021, and the voice signal from the microphone is obtained according to the set sampling parameters, so that the audio signal and the voice signal are obtained. The voice signals have the same sampling parameters for the convenience of subsequent mixing processing.

Of course, the sampling parameter of the voice signal and the sampling parameter of the audio signal may also be different. If they are not the same, the sampling parameter conversion of the audio signal or the voice signal needs to be performed before the subsequent sound mixing processing. , so that the two sampling parameters are the same.

Step 304, adjusting the volume of the audio signal according to the change in the state of the volume button;

In the specific implementation, the volume control of background music, sound effects, etc., and the volume control of the microphone can be independently designed, so that the volume adjustment of the background music and sound effects does not affect the recording of the microphone sound track. During the playback of background music or sound effects, the user can touch the volume buttons of the background music or sound effects according to actual needs. When the volume is changed, the volume can be gradually increased according to the preset fade-in step, and when the volume is lowered, the volume can be gradually decreased according to the preset fade-out step, so that when the volume switches between different states, there is a fade-in and fade-out effect for transition. The fade-in step size and the preset fade-out step size can be set according to actual needs, and the specific values of the two can be the same or different.

Step 305, performing voice change, filtering, and noise reduction processing on the audio signal or the voice signal;

In specific implementation, tools such as a voice changer and a sound filter can be used to change the audio signal or the voice signal (for example, change the timbre, tone, etc. of the sound), filter (filter out signals other than the preset frequency band), Noise reduction processing (removing digital noise, etc.) to obtain higher quality audio signals or speech signals.

Step 306: Mix and play the audio signal and the voice signal.

Mixing is to combine the sounds of multiple sources into a stereo track (Stereo) or monophonic track (Mono) for output. Specifically, the following two methods can be used for mixing:

The first is to adjust the amplitude of the audio signal and the voice signal, for example, adjust the amplitude of the audio signal to a first amplitude threshold, adjust the amplitude of the voice signal to a second amplitude threshold, and adjust the amplitude of the audio signal to a first amplitude threshold. The threshold and the second amplitude threshold can be the same or different, the first amplitude threshold can be greater than the second amplitude threshold if the audio signal is to be highlighted, and the second amplitude threshold can be greater than the second amplitude threshold if the speech signal is to be highlighted. an amplitude threshold, in addition, in order to ensure that the signal does not overflow after mixing, the first amplitude threshold and the second amplitude threshold can be set according to the overflow amplitude limit; then the audio signal after the amplitude adjustment and the voice signal are superimposed as All the way, play the signal superimposed as one way.

The second is to directly superimpose the audio signal and the voice signal into one channel, and determine whether the superimposed signal overflows. If no overflow occurs, no other processing is performed; if an overflow occurs, the superimposed overflow The signal is attenuated, for example, the attenuation factor corresponding to the overflow signal value is calculated, the overflow signal is attenuated according to the attenuation factor, and finally the superimposed signal is played.

The following describes the live broadcast sound processing method provided by the embodiment of the present invention when the audio signal is background music and when the audio signal is sound effect.

Referring to FIGS. 4 a to 4 d , FIGS. 4 a to 4 d are diagrams showing effects when the audio signal selected by the user is background music. In the live broadcast interface shown in Figure 4a, an atmosphere manager can be set on the live broadcast interface. Users can click the atmosphere manager on the live broadcast interface to enter the atmosphere management interface. The atmosphere management interface is shown in Figure 4b, including background music options and sound effects options. The user determines the type of the selected audio file. After the user selects the background music option, the interface can be shown in Figure 4c. The background music file; in the specific implementation, the mobile terminal can connect the network to obtain the background music file from the server according to the user's input, and if the local storage of the mobile terminal stores the background music file, the mobile terminal can also obtain the background music file from its own local The background music file is acquired in the storage, so as to realize the rapid acquisition of the background music file. After the background music file is acquired, it is saved in the file list corresponding to the background music option. After the mobile terminal acquires the background music file, the user can click to play the background music file, and the mobile terminal decodes the background music file to obtain the background music. Specifically, the background music file can be decoded into a pulse-code modulation (PCM) format. signal, the playing interface of background music can be shown in Figure 4d.

At the same time, the mobile terminal can use the microphone to collect the user's voice signal, and use a preset tool to perform audio processing on the background music or the voice signal. The preset tool includes a voice changer and a voice filter, so as to obtain high-quality sound A background music signal or a voice signal, the background music signal and the voice signal are mixed and played.

If the background music file is not decoded and cached in advance, but is decoded in real time during the live broadcast, the background music file can be decoded in segments. Since it is time-consuming to decode background music files, for example, MP3 files are used as background music files, and it takes about three seconds to fully decode a three-minute MP3 file. Make users feel that there is a long time delay, therefore, the background music file can be decoded in segments, and the background music signal with preset length can be obtained by decoding first. It takes tens of milliseconds to decode the background music signal of the preset length, mix the user's voice signal with the background music signal and play it. The speed of playing music is generally slower than the decoding speed. Therefore, after decoding the background music signal with a preset length, you can play it while decoding. This solves the problem that the user feels a long delay. The user can select the background music to play. You can hear the background music playing immediately after the function is activated.

During the playing process, as shown in Figure 4d, the playing progress of the background music can be displayed, and the playing of the background music can also be paused. In addition, the user can touch the volume button of the background music according to actual needs, and the mobile terminal adjusts the volume of the background music according to the change of the state of the volume button. When the volume is increased, the volume can be gradually increased according to the preset fade-in step. When lowering the volume, you can gradually reduce the volume according to the preset fade-out step, so that when the volume is switched between different states, there is a fade-in and fade-out effect as a transition. The volume control of the background music and the volume control of the microphone can be independent to The adjustment of the background music volume does not affect the recording of the microphone audio track.

When the user wants to delete a certain background music file, he can click the atmosphere manager, enter the background music option, view all the background music files in the list, and delete the background music file that needs to be deleted.

5a to 5d are diagrams showing effects when the audio signal selected by the user is a sound effect. In the live broadcast interface shown in Figure 5a, an atmosphere manager can be set on the live broadcast interface. Users can click the atmosphere manager on the live broadcast interface to enter the atmosphere management interface. The atmosphere management interface is shown in Figure 5b, including background music options and sound effects options. The user determines the type of the selected audio file. When the user selects the sound effect option, the interface can be shown in Figure 5c, including the high-quality sound effect files screened out by the mobile terminal from the server or local storage according to the operation recommendation, and the user can add the selected sound effect files. Commonly used sound effects files, such as adding life rewards, jumping sounds, eating gold coins, etc. are commonly used sound effects files. The mobile terminal saves the sound effect files added by the user in the file list corresponding to the sound effect option. The user can click to play the desired sound effect file, move The terminal decodes the sound effect file to obtain a sound effect signal. Specifically, the sound effect file may be decoded into a signal in a pulse-code modulation (Pulse-codemodulation, PCM) format. The sound effect playback interface is shown in FIG. 5d.

At the same time, the mobile terminal can use the microphone to collect the user's voice signal, and use a preset tool to perform audio processing on the sound effect or the voice signal. The preset tool includes a voice changer and a voice filter to obtain high-quality sound effects. signal or voice signal, mix the sound effect signal with the voice signal and play.

During the playback, as shown in FIG. 5d, the playback progress of the sound effect can be displayed, and the playback of the sound effect can also be paused. In addition, the user can touch the volume button of the sound effect according to actual needs, and the mobile terminal adjusts the volume of the sound effect according to the change of the state of the volume button. When the volume is used, the volume can be gradually reduced according to the preset fade-out step, so that when the volume is switched between different states, there is a fade-in and fade-out effect for transition. Adjustments do not affect microphone audio track recording.

When the user wants to cancel a sound effect file, he can click the atmosphere manager, enter the sound effect option, view all the sound effect files in the list, and click Cancel to cancel the corresponding sound effect file.

It should be noted that the above effect diagrams are only schematic representations, and do not constitute a final limitation on the manner of displaying the effects of the present invention.

That is, in this embodiment, when the user wants to use background music or sound effects to enrich the atmosphere of the live broadcast, he can directly select the desired audio file in the atmosphere manager of the live broadcast interface, and click to play. It is transparent to the user and does not affect the entire live broadcast process. During the entire live broadcast process, the user only needs to operate the mobile terminal and does not need to operate other external devices.

In this embodiment, the mobile terminal can acquire the audio file selected by the user, decode the audio file to obtain the audio signal, so it is no longer necessary to play the audio signal through other external playback devices, and it is no longer necessary to use the microphone of the mobile terminal to collect the audio signal, After acquiring the audio signal, the microphone of the mobile terminal can be adjusted to focus on collecting the user's voice signal, and then the acquired audio signal and the collected user's voice signal are mixed and played, thereby improving the quality of the final output signal.

Embodiment 3

In order to better implement the above method, the present invention also provides a mobile terminal. As shown in FIG. 6 , the mobile terminal in this embodiment includes:

Obtaining unit 601, for obtaining the audio file selected by the user;

a decoding unit 602, for decoding the audio file to obtain an audio signal;

a collection unit 603, configured to collect the voice signal of the user;

The first processing unit 604 is configured to mix and play the audio signal and the voice signal.

In a possible implementation manner, the obtaining unit 601 is specifically configured to obtain the audio file selected by the user from a server through a network, or obtain the audio file selected by the user from the local storage of the mobile terminal.

In a possible implementation manner, the decoding unit 602 includes:

a parsing subunit for parsing the audio file to obtain sampling parameters of the audio file;

A decoding subunit, configured to decode the audio file according to the sampling parameter of the audio file to obtain the audio signal.

In a possible implementation manner, the adopted parameters of the audio file include sampling rate, sampling precision and channel mode;

The decoding subunit is specifically used to calculate the encoding bit rate of the audio file, where the encoding bit rate is the product of the sampling rate, sampling precision, and channel mode, that is, bit rate=sampling rate*sampling precision * Channel mode; decoding is the reverse process of encoding, and the audio signal can be obtained by decoding the audio file according to the encoding bit rate.

Among them, the adoption rate, also known as sampling speed or sampling frequency, defines the number of samples per unit time, which is expressed in Hertz (Hz). For sound signals, the sampling rate can be used to describe the sound quality and pitch of the sound file. It is a quality standard for measuring sound cards and sound files. The higher the sampling rate, that is, the shorter the sampling interval, the more sample data can be obtained per unit time, and the more accurate the representation of the signal waveform. There is a certain relationship between the sampling rate and the frequency of the original signal. According to Nyquist theory, only when the sampling rate is higher than twice the highest frequency of the original signal can the signal represented by the digital signal be restored to the original signal.

Sampling accuracy, which reflects the accuracy of measuring the amplitude of the sound waveform, in bits (Bit), such as 8 bits, 16 bits. 8 bits can divide a sound wave into 256 levels, and 16 bits can divide the same wave into a signal of 65,536 levels. For example, each sound sample is represented by 16 bits (2 bytes), the measured sound sample value is in the range of 0 to 65535, and its precision is 1/65536 of the input signal. The size of the sample bits affects the quality of the sound. The more bits, the higher the quality of the sound, and the more storage space required; the fewer the bits, the lower the quality of the sound, and the less storage space required.

Channel mode, divided into single channel or dual channel, the value of single channel is 1, and the value of dual channel is 2.

After obtaining the sampling parameters of the audio file, the collection unit 603 may set the sampling parameters for the microphone to collect the user's voice signal according to the obtained sampling parameters of the audio file, and obtain the voice signal from the microphone according to the set sampling parameters, so that The audio signal and the speech signal have the same sampling parameters, so as to facilitate subsequent sound mixing processing.

In a possible implementation manner, the format of the audio signal includes a pulse code modulation PCM format and a waveform WAV format, and the audio signal includes background music or sound effects.

In a possible implementation manner, the mobile terminal further includes:

The volume adjustment unit adjusts the volume of the background music or sound effect according to the change of the state of the volume button, when the volume is increased, the volume is gradually increased according to the preset fade-in step size, and when the volume is decreased, the volume is gradually increased according to the preset fade-out step size Gradually decrease the volume.

In a possible implementation manner, the mobile terminal further includes:

The second processing unit is configured to perform voice changing, filtering and noise reduction processing on the audio signal or the voice signal.

In a possible implementation manner, the first processing unit 604 includes:

an amplitude adjustment subunit, configured to adjust the amplitude of the audio signal to a first amplitude threshold, and to adjust the amplitude of the speech signal to a second amplitude threshold;

A superimposing subunit, used for superimposing the audio signal and the voice signal after the amplitude adjustment into one channel;

The playback sub-unit is used to play the signals that are superimposed into one channel.

In a possible implementation manner, the first processing unit 604 includes:

a superimposing subunit, used for superimposing the audio signal and the voice signal into one channel;

The attenuation subunit is used to attenuate the overflowed signal after superposition;

The playback sub-unit is used to play the signals that are superimposed into one channel.

It should be noted that, when the mobile terminal provided in the above-mentioned embodiment processes live sound, only the division of the above-mentioned functional modules is used as an example. The internal structure of the device is divided into different functional modules to complete all or part of the functions described above. In addition, the mobile terminal and the live sound processing method provided by the above embodiments belong to the same concept, and the specific implementation process is detailed in the method embodiment, which will not be repeated here.

In this embodiment, the acquisition unit of the mobile terminal can acquire the audio file selected by the user, and the decoding unit decodes the audio file to obtain the audio signal, so it is no longer necessary to play the audio signal through other external playback devices, and it is no longer necessary to use the microphone of the mobile terminal After collecting the audio signal, after obtaining the audio signal, the collecting unit can adjust the microphone of the mobile terminal to focus on collecting the user's voice signal, and then the mixing and playing unit mixes the obtained audio signal with the collected user's voice signal and mixes it. playback, thus improving the quality of the final output signal.

Embodiment 4

Correspondingly, an embodiment of the present invention further provides a mobile terminal. As shown in FIG. 7 , the mobile terminal may include a radio frequency (RF, Radio Frequency) circuit 701 and a memory 702 including one or more computer-readable storage media. , an input unit 703, a display unit 704, a sensor 705, an audio circuit 706, a wireless fidelity (WiFi, Wireless Fidelity) module 707, a processor 708 including one or more processing cores, and a power supply 709 and other components. Those skilled in the art can understand that the structure of the mobile terminal shown in FIG. 7 does not constitute a limitation on the terminal, and may include more or less components than those shown in the figure, or combine some components, or arrange different components. in:

The RF circuit 701 can be used for receiving and sending signals during the process of sending and receiving information or talking. In particular, after receiving the downlink information of the base station, it is handed over to one or more processors 708 for processing; in addition, it sends the data related to the uplink to the base station. . Typically, the RF circuit 701 includes, but is not limited to, an antenna, at least one amplifier, a tuner, one or more oscillators, a Subscriber Identity Module (SIM) card, a transceiver, a coupler, a low noise amplifier (LNA, Low Noise Amplifier), duplexer, etc. In addition, the RF circuit 701 can also communicate with the network and other devices through wireless communication. The wireless communication can use any communication standard or protocol, including but not limited to Global System for Mobile Communication (GSM, Global System of Mobile communication), General Packet Radio Service (GPRS, General Packet Radio Service), Code Division Multiple Access (CDMA, Code Division Multiple Access), Wideband Code Division Multiple Access (WCDMA, Wideband Code Division Multiple Access), Long Term Evolution (LTE, Long TermEvolution), email, Short Messaging Service (SMS, Short Messaging Service), etc.

The memory 702 can be used to store software programs and modules, and the processor 708 executes various functional applications and data processing by running the software programs and modules stored in the memory 702 . The memory 702 may mainly include a stored program area and a stored data area, wherein the stored program area may store an operating system, an application program (such as a sound playback function, an image playback function, etc.) required for at least one function, and the like; Data (such as audio data, phonebook, etc.) created by the use of the mobile terminal, etc. Additionally, memory 702 may include high-speed random access memory, and may also include non-volatile memory, such as at least one magnetic disk storage device, flash memory device, or other volatile solid state storage device. Accordingly, memory 702 may also include a memory controller to provide processor 708 and input unit 703 access to memory 702.

The input unit 703 can be used to receive input numerical or character information, and generate keyboard, mouse, joystick, optical or trackball signal input related to user settings and function control. Specifically, in a specific embodiment, the input unit 703 may include a touch-sensitive surface as well as other input devices. A touch-sensitive surface, also known as a touch display or trackpad, collects the user's touch operations on or near it (such as the user's finger, stylus, etc., any suitable operation near the surface), and drive the corresponding connection device according to the preset program. Alternatively, the touch-sensitive surface may include two parts, a touch detection device and a touch controller. Among them, the touch detection device detects the user's touch orientation, detects the signal brought by the touch operation, and transmits the signal to the touch controller; the touch controller receives the touch information from the touch detection device, converts it into contact coordinates, and then sends it to the touch controller. To the processor 708, and can receive the command sent by the processor 708 and execute it. Additionally, touch-sensitive surfaces can be implemented using resistive, capacitive, infrared, and surface acoustic wave types. In addition to touch-sensitive surfaces, input unit 703 may also include other input devices. Specifically, other input devices may include, but are not limited to, one or more of physical keyboards, function keys (such as volume control keys, switch keys, etc.), trackballs, mice, joysticks, and the like.

The display unit 704 may be used to display information input by the user or information provided to the user and various graphical user interfaces of the terminal, which may be composed of graphics, text, icons, videos and any combination thereof. The display unit 704 may include a display panel, and optionally, the display panel may be configured in the form of a liquid crystal display (LCD, Liquid Crystal Display), an organic light-emitting diode (OLED, Organic Light-Emitting Diode), or the like. Further, the touch-sensitive surface may cover the display panel, and when the touch-sensitive surface detects a touch operation on or near it, it is transmitted to the processor 708 to determine the type of the touch event, and then the processor 708 displays the touch event according to the type of the touch event. The corresponding visual output is provided on the panel. Although in FIG. 7 the touch-sensitive surface and the display panel are implemented as two separate components to implement the input and input functions, in some embodiments, the touch-sensitive surface and the display panel may be integrated to implement the input and output functions.

The terminal may also include at least one sensor 705, such as a light sensor, a motion sensor, and other sensors. Specifically, the light sensor may include an ambient light sensor and a proximity sensor, wherein the ambient light sensor may adjust the brightness of the display panel according to the brightness of the ambient light, and the proximity sensor may turn off the display panel and/or the backlight when the terminal is moved to the ear . As a kind of motion sensor, the gravitational acceleration sensor can detect the magnitude of acceleration in all directions (usually three axes), and can detect the magnitude and direction of gravity when stationary, and can be used for applications that recognize the attitude of mobile phones (such as horizontal and vertical screen switching, related games, magnetometer attitude calibration), vibration recognition related functions (such as pedometer, tapping), etc.; as for other sensors such as gyroscopes, barometers, hygrometers, thermometers, infrared sensors, etc. Repeat.

Audio circuit 706, speakers, and microphones may provide an audio interface between the user and the terminal. The audio circuit 706 can convert the received audio data into an electrical signal, and transmit it to the speaker, which is converted into a sound signal for output; on the other hand, the microphone converts the collected sound signal into an electrical signal, which is converted after being received by the audio circuit 706 In the form of audio data, the audio data is output to the processor 708 for processing, and then sent to, for example, another terminal via the RF circuit 701, or the audio data is output to the memory 702 for further processing. The audio circuit 706 may also include an earphone jack to provide communication between peripheral headphones and the terminal.

WiFi is a short-distance wireless transmission technology, and the mobile terminal can help users to send and receive emails, browse web pages, access streaming media, etc. through the WiFi module 707, which provides users with wireless broadband Internet access. Although FIG. 7 shows the WiFi module 707, it can be understood that it is not a necessary component of the terminal, and can be completely omitted as required within the scope of not changing the essence of the invention.

The processor 708 is the control center of the mobile terminal, uses various interfaces and lines to connect various parts of the entire mobile terminal, runs or executes the software programs and/or modules stored in the memory 702, and calls the data stored in the memory 702. , perform various functions of the terminal and process data, so as to monitor the mobile phone as a whole. Optionally, the processor 708 may include one or more processing cores; preferably, the processor 708 may integrate an application processor and a modem processor, wherein the application processor mainly processes the operating system, user interface, and application programs, etc. , the modem processor mainly deals with wireless communication. It can be understood that the above-mentioned modulation and demodulation processor may not be integrated into the processor 708 .

The mobile terminal also includes a power supply 709 (such as a battery) for supplying power to various components. Preferably, the power supply can be logically connected to the processor 708 through a power management system, so as to manage charging, discharging, and power consumption management functions through the power management system. Power supply 709 may also include one or more DC or AC power sources, recharging systems, power failure detection circuits, power converters or inverters, power status indicators, and any other components.

Although not shown, the mobile terminal may also include a camera, a Bluetooth module, and the like, which will not be repeated here. Specifically, in this embodiment, the processor 708 in the mobile terminal loads the executable files corresponding to the processes of one or more application programs into the memory 702 according to the following instructions, and the processor 708 executes them and stores them in the memory 702. The application program in the memory 702, thereby realizing various functions:

Acquire the audio file selected by the user; decode the audio file to obtain an audio signal; collect the user's voice signal; mix the audio signal and the voice signal and play.

The audio file may be acquired from a server through a network, or the audio file may be acquired from a local storage of the mobile terminal, and the audio signal includes background music or sound effects.

Those of ordinary skill in the art can understand that all or part of the steps in the various methods of the above embodiments can be completed by instructing relevant hardware through a program, and the program can be stored in a computer-readable storage medium, and the storage medium can include: Read Only Memory (ROM, Read Only Memory), Random Access Memory (RAM, Random Access Memory), magnetic disk or optical disk, etc.

The above provides a detailed introduction to a live sound processing method and a mobile terminal provided by the embodiments of the present invention. The principles and implementations of the present invention are described in this paper by using specific examples. The descriptions of the above embodiments are only used to help understanding The method of the present invention and its core idea; at the same time, for those skilled in the art, according to the idea of the present invention, there will be changes in the specific implementation and application scope. In summary, the content of this description should not be understood to limit the present invention.

Claims (19)

1. a method for processing live sound, comprising: After the user enters the live broadcast interface through the live broadcast software, acquiring the audio file selected by the user includes: acquiring the audio file selected by the user according to the identifier of the audio file input by the user in the atmosphere manager of the live broadcast interface; Decoding the audio file in real time to obtain an audio signal, comprising: decoding the audio file segmentally to obtain an audio signal of a preset length; simultaneously collecting the user's voice signal; Mixing and playing the audio signal and the voice signal includes: mixing and playing the audio signal of the preset length in real time with the voice signal. 2. The method according to claim 1, wherein the acquiring the audio file selected by the user comprises: acquiring the audio file selected by the user from a server through a network, or acquiring the audio file selected by the local storage of the mobile terminal. the audio file selected by the user. 3. The method according to claim 1, wherein the real-time decoding of the audio file to obtain an audio signal comprises: parsing the audio file to obtain sampling parameters of the audio file; The audio signal is obtained by decoding the audio file according to the sampling parameters of the audio file. 4. The method according to claim 3, wherein the sampling parameters of the audio file include sampling rate, sampling precision and channel mode, and the audio file is decoded according to the sampling parameters of the audio file to obtain the obtained data. The audio signal includes: Calculate the encoding bit rate of the audio file, where the encoding bit rate is the product of the sampling rate, sampling precision, and channel mode; The audio signal is obtained by decoding the audio file according to the encoding bit rate. 5. The method according to any one of claims 1 to 4, wherein the format of the audio signal includes a pulse code modulation PCM format and a waveform WAV format, and the audio signal includes background music or sound effects. 6. The method according to claim 5, wherein after decoding the audio file to obtain the audio signal, the method further comprises: The volume of the background music or sound effect is adjusted according to the change of the state of the volume button, when the volume is increased, the volume is gradually increased according to the preset fade-in step, and when the volume is decreased, the volume is gradually decreased according to the preset fade-out step . 7. The method according to claim 1, wherein before the audio signal and the voice signal are mixed and played, the method further comprises: Perform voice changing, filtering, and noise reduction processing on the audio signal or the speech signal. 8. The method according to claim 1, wherein mixing and playing the audio signal and the voice signal comprises: adjusting the amplitude of the audio signal to a first amplitude threshold, and adjusting the amplitude of the voice signal to a second amplitude threshold; superimposing the audio signal after the amplitude adjustment and the voice signal into one channel; Play the signal superimposed as one. 9. The method according to claim 1, wherein mixing and playing the audio signal and the voice signal comprises: superimposing the audio signal and the voice signal into one channel; Attenuate the overflowed signal after superposition; Play the signal superimposed as one. 10. A mobile terminal, comprising: an obtaining unit, configured to obtain the audio file selected by the user after the user enters the live broadcast interface through the live broadcast software, including: obtaining the user selected audio file according to the identifier of the audio file input by the user in the atmosphere manager of the live broadcast interface audio files; a decoding unit, used for decoding the audio file in real time to obtain an audio signal, including: segment decoding the audio file to obtain an audio signal of a preset length; an acquisition unit for simultaneously acquiring the voice signal of the user; The first processing unit, configured to mix and play the audio signal and the voice signal, includes: mixing and playing the audio signal of the preset length and the voice signal in real time. 11 . The mobile terminal according to claim 10 , wherein the acquiring unit is specifically configured to acquire the audio file selected by the user from a server through a network, or acquire the audio file from a local storage of the mobile terminal. 12 . User selected audio file. 12. The mobile terminal according to claim 10, wherein the decoding unit comprises: a parsing subunit for parsing the audio file to obtain sampling parameters of the audio file; A decoding subunit, configured to decode the audio file according to the sampling parameter of the audio file to obtain the audio signal. 13. The mobile terminal according to claim 12, wherein the sampling parameters of the audio file include sampling rate, sampling precision and channel mode; The decoding subunit is specifically configured to calculate the encoding bit rate of the audio file, where the encoding bit rate is the product of the sampling rate, sampling precision, and channel mode; decode the encoding bit rate according to the encoding bit rate. The audio file gets the audio signal. 14. The mobile terminal according to any one of claims 10 to 13, wherein the format of the audio signal includes a pulse code modulation PCM format and a waveform WAV format, and the audio signal includes background music or sound effects. 15. The mobile terminal according to claim 14, wherein the mobile terminal further comprises: The volume adjustment unit adjusts the volume of the background music or sound effect according to the change of the state of the volume button, when the volume is increased, the volume is gradually increased according to the preset fade-in step size, and when the volume is decreased, the volume is gradually increased according to the preset fade-out step size Gradually decrease the volume. 16. The mobile terminal according to claim 10, wherein the mobile terminal further comprises: The second processing unit is configured to perform voice changing, filtering and noise reduction processing on the audio signal or the voice signal. 17. The mobile terminal according to claim 10, wherein the first processing unit comprises: an amplitude adjustment subunit, configured to adjust the amplitude of the audio signal to a first amplitude threshold, and to adjust the amplitude of the speech signal to a second amplitude threshold; A superimposing subunit, used for superimposing the audio signal and the voice signal after the amplitude adjustment into one channel; The playback sub-unit is used to play the signals that are superimposed into one channel. 18. The mobile terminal according to claim 10, wherein the first processing unit comprises: a superimposing subunit, used for superimposing the audio signal and the voice signal into one channel; The attenuation subunit is used to attenuate the overflowed signal after superposition; The playback sub-unit is used to play the signals that are superimposed into one channel. 19. A storage medium having a computer program stored thereon, characterized in that, when the computer program is run on a computer, the computer is made to execute the live sound processing method according to any one of claims 1 to 9 .

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