CN112887776A - Method and equipment for reducing audio delay and computer-readable storage medium - Google Patents

Method and equipment for reducing audio delay and computer-readable storage medium Download PDF

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CN112887776A
CN112887776A CN202110290076.2A CN202110290076A CN112887776A CN 112887776 A CN112887776 A CN 112887776A CN 202110290076 A CN202110290076 A CN 202110290076A CN 112887776 A CN112887776 A CN 112887776A
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played
audio
data
audio data
queue
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CN112887776B (en
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廖松茂
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Nubia Technology Co Ltd
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Nubia Technology Co Ltd
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    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04NPICTORIAL COMMUNICATION, e.g. TELEVISION
    • H04N21/00Selective content distribution, e.g. interactive television or video on demand [VOD]
    • H04N21/40Client devices specifically adapted for the reception of or interaction with content, e.g. set-top-box [STB]; Operations thereof
    • H04N21/43Processing of content or additional data, e.g. demultiplexing additional data from a digital video stream; Elementary client operations, e.g. monitoring of home network or synchronising decoder's clock; Client middleware
    • H04N21/4302Content synchronisation processes, e.g. decoder synchronisation
    • H04N21/4307Synchronising the rendering of multiple content streams or additional data on devices, e.g. synchronisation of audio on a mobile phone with the video output on the TV screen
    • GPHYSICS
    • G06COMPUTING; CALCULATING OR COUNTING
    • G06FELECTRIC DIGITAL DATA PROCESSING
    • G06F3/00Input arrangements for transferring data to be processed into a form capable of being handled by the computer; Output arrangements for transferring data from processing unit to output unit, e.g. interface arrangements
    • G06F3/14Digital output to display device ; Cooperation and interconnection of the display device with other functional units
    • G06F3/1454Digital output to display device ; Cooperation and interconnection of the display device with other functional units involving copying of the display data of a local workstation or window to a remote workstation or window so that an actual copy of the data is displayed simultaneously on two or more displays, e.g. teledisplay
    • GPHYSICS
    • G06COMPUTING; CALCULATING OR COUNTING
    • G06FELECTRIC DIGITAL DATA PROCESSING
    • G06F3/00Input arrangements for transferring data to be processed into a form capable of being handled by the computer; Output arrangements for transferring data from processing unit to output unit, e.g. interface arrangements
    • G06F3/16Sound input; Sound output
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04NPICTORIAL COMMUNICATION, e.g. TELEVISION
    • H04N21/00Selective content distribution, e.g. interactive television or video on demand [VOD]
    • H04N21/40Client devices specifically adapted for the reception of or interaction with content, e.g. set-top-box [STB]; Operations thereof
    • H04N21/43Processing of content or additional data, e.g. demultiplexing additional data from a digital video stream; Elementary client operations, e.g. monitoring of home network or synchronising decoder's clock; Client middleware
    • H04N21/433Content storage operation, e.g. storage operation in response to a pause request, caching operations
    • H04N21/4334Recording operations
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04NPICTORIAL COMMUNICATION, e.g. TELEVISION
    • H04N21/00Selective content distribution, e.g. interactive television or video on demand [VOD]
    • H04N21/40Client devices specifically adapted for the reception of or interaction with content, e.g. set-top-box [STB]; Operations thereof
    • H04N21/43Processing of content or additional data, e.g. demultiplexing additional data from a digital video stream; Elementary client operations, e.g. monitoring of home network or synchronising decoder's clock; Client middleware
    • H04N21/439Processing of audio elementary streams

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  • Engineering & Computer Science (AREA)
  • Theoretical Computer Science (AREA)
  • Multimedia (AREA)
  • Signal Processing (AREA)
  • Human Computer Interaction (AREA)
  • Physics & Mathematics (AREA)
  • General Engineering & Computer Science (AREA)
  • General Physics & Mathematics (AREA)
  • Health & Medical Sciences (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • General Health & Medical Sciences (AREA)
  • Circuit For Audible Band Transducer (AREA)

Abstract

The invention discloses a method, equipment and a computer readable storage medium for reducing audio delay, wherein the method comprises the following steps: storing the audio data to be played into an audio queue to be played; acquiring the audio data to be played from the audio queue to be played, sending the audio data to be played to a sound card for playing, and deleting the audio data to be played, which is sent to the sound card for playing, from the audio queue to be played; detecting whether the data volume of the audio data to be played remaining in the audio queue to be played is larger than a preset threshold value or not; if yes, the remaining audio data to be played are accelerated and played according to preset multiples. The scheme for reducing the audio delay is realized, the problem of unsynchronized sound and picture in the screen projection process is solved, and the overall quality and the user experience of the screen projection are improved.

Description

Method and equipment for reducing audio delay and computer-readable storage medium
Technical Field
The present invention relates to the field of audio data processing technologies, and in particular, to a method and an apparatus for reducing audio delay, and a computer-readable storage medium.
Background
In the prior art, with the rapid development of the communication technology, the screen projection technology is also improved unprecedentedly, and the diversified screen projection requirements of users can be met. When a screen projection project with an audio playing function is developed, the inventor finds that sound is delayed in the screen projection process, the delay is accumulated continuously in the screen projection process, and when the delay is accumulated to a certain amount, the problem of obvious picture-sound asynchronism occurs, the screen projection quality and the user experience are influenced, and the technical problem to be solved urgently is solved.
Disclosure of Invention
It is a primary object of the present invention to provide a method, apparatus and computer-readable storage medium for reducing audio latency, which at least partially solve the problems in the prior art.
In order to achieve the above object, the present invention provides a method for reducing audio delay, comprising the steps of:
storing the audio data to be played into an audio queue to be played;
acquiring the audio data to be played from the audio queue to be played, sending the audio data to be played to a sound card for playing, and deleting the audio data to be played, which is sent to the sound card for playing, from the audio queue to be played;
detecting whether the data volume of the audio data to be played remaining in the audio queue to be played is larger than a preset threshold value or not;
if yes, the remaining audio data to be played are accelerated and played according to preset multiples.
Optionally, the step of performing accelerated playing on the remaining audio data to be played according to a preset multiple includes:
calling audio frequency speed change software to accelerate the rest audio frequency data to be played according to preset multiples to generate acceleration result data;
and sending the acceleration result data to a sound card for playing.
Optionally, the step of calling audio frequency speed change software to accelerate the remaining audio data to be played according to a preset multiple to generate acceleration result data includes:
integrating the rest audio data to be played to an audio buffer to be accelerated;
and the audio frequency speed changing software calls the data in the audio frequency buffer to be accelerated and accelerates the data according to a preset multiple to generate acceleration result data, and the acceleration result data is stored in the acceleration result audio frequency buffer.
Optionally, after the step of detecting whether the data amount of the remaining audio data to be played in the audio queue to be played is greater than a preset threshold, the method further includes:
if not, returning to the step of detecting whether the data volume of the audio data to be played remaining in the audio queue to be played is larger than a preset threshold value.
Optionally, before the step of storing the audio data to be played into the audio queue to be played, the method further includes:
receiving encoded audio data;
and decoding the encoded audio data to generate audio data to be played.
The present invention also provides an apparatus for reducing audio latency, the apparatus comprising a memory, a processor, and a computer program stored on the memory and executable on the processor, the computer program when executed by the processor implementing:
storing the audio data to be played into an audio queue to be played;
acquiring the audio data to be played from the audio queue to be played, sending the audio data to be played to a sound card for playing, and deleting the audio data to be played, which is sent to the sound card for playing, from the audio queue to be played;
detecting whether the data volume of the audio data to be played remaining in the audio queue to be played is larger than a preset threshold value or not;
if yes, the remaining audio data to be played are accelerated and played according to preset multiples.
Optionally, the computer program when executed by the processor implements:
calling audio frequency speed change software to accelerate the rest audio frequency data to be played according to preset multiples to generate acceleration result data;
and sending the acceleration result data to a sound card for playing.
Optionally, the computer program when executed by the processor implements:
integrating the rest audio data to be played to an audio buffer (buffer register) to be accelerated;
and the audio frequency speed changing software calls the data in the audio frequency buffer to be accelerated and accelerates the data according to a preset multiple to generate acceleration result data, and the acceleration result data is stored in the acceleration result audio frequency buffer.
Optionally, the computer program when executed by the processor implements:
if not, returning to the step of detecting whether the data volume of the audio data to be played remaining in the audio queue to be played is larger than a preset threshold value;
receiving encoded audio data;
and decoding the encoded audio data to generate audio data to be played.
The present invention also provides a computer readable storage medium, having stored thereon a program for reducing audio latency, which when executed by a processor, implements the steps of the method for reducing audio latency as set forth in any one of the preceding claims.
The invention has the beneficial effects that: the invention provides a method, equipment and a computer readable storage medium for reducing audio delay, which are characterized in that audio data to be played are stored in an audio queue to be played; acquiring the audio data to be played from the audio queue to be played, sending the audio data to be played to a sound card for playing, and deleting the audio data to be played, which is sent to the sound card for playing, from the audio queue to be played; detecting whether the data volume of the audio data to be played remaining in the audio queue to be played is larger than a preset threshold value or not; if yes, the remaining audio data to be played are accelerated and played according to preset multiples. The scheme for reducing the audio delay is realized, the problem of unsynchronized sound and picture in the screen projection process is solved, and the overall quality and the user experience of the screen projection are improved.
Drawings
The invention will be further described with reference to the accompanying drawings and examples, in which:
fig. 1 is a schematic diagram of a hardware structure of a mobile terminal according to the present invention;
fig. 2 is a communication network system architecture diagram provided by an embodiment of the present invention;
FIG. 3 is a flow chart of a first embodiment of the method for reducing audio delay of the present invention;
FIG. 4 is a flow chart of a second embodiment of the method of reducing audio delay of the present invention;
FIG. 5 is a flow chart of a third embodiment of the method for reducing audio delay of the present invention;
FIG. 6 is a flow chart of a fourth embodiment of the method for reducing audio delay of the present invention;
fig. 7 is a flowchart of a fifth embodiment of the method for reducing audio delay according to the present invention.
Detailed Description
It should be understood that the specific embodiments described herein are merely illustrative of the invention and are not intended to limit the invention.
In the following description, suffixes such as "module", "component", or "unit" used to denote elements are used only for facilitating the explanation of the present invention, and have no specific meaning in itself. Thus, "module", "component" or "unit" may be used mixedly.
The terminal may be implemented in various forms. For example, the terminal described in the present invention may include a mobile terminal such as a mobile phone, a tablet computer, a notebook computer, a palmtop computer, a Personal Digital Assistant (PDA), a Portable Media Player (PMP), a navigation device, a wearable device, a smart band, a pedometer, and the like, and a fixed terminal such as a Digital TV, a desktop computer, and the like.
The following description will be given by way of example of a mobile terminal, and it will be understood by those skilled in the art that the construction according to the embodiment of the present invention can be applied to a fixed type terminal, in addition to elements particularly used for mobile purposes.
Referring to fig. 1, which is a schematic diagram of a hardware structure of a mobile terminal for implementing various embodiments of the present invention, the mobile terminal 100 may include: RF (Radio Frequency) unit 101, WiFi module 102, audio output unit 103, a/V (audio/video) input unit 104, sensor 105, display unit 106, user input unit 107, interface unit 108, memory 109, processor 110, and power supply 111. Those skilled in the art will appreciate that the mobile terminal architecture shown in fig. 1 is not intended to be limiting of mobile terminals, which may include more or fewer components than those shown, or some components may be combined, or a different arrangement of components.
The following describes each component of the mobile terminal in detail with reference to fig. 1:
the radio frequency unit 101 may be configured to receive and transmit signals during information transmission and reception or during a call, and specifically, receive downlink information of a base station and then process the downlink information to the processor 110; in addition, the uplink data is transmitted to the base station. Typically, radio frequency unit 101 includes, but is not limited to, an antenna, at least one amplifier, a transceiver, a coupler, a low noise amplifier, a duplexer, and the like. In addition, the radio frequency unit 101 can also communicate with a network and other devices through wireless communication. The wireless communication may use any communication standard or protocol, including but not limited to GSM (Global System for mobile communications), GPRS (General Packet Radio Service), CDMA2000(Code Division Multiple Access 2000), WCDMA (Wideband Code Division Multiple Access), TD-SCDMA (Time Division-Synchronous Code Division Multiple Access), FDD-LTE (Frequency Division duplex Long Term Evolution), and TDD-LTE (Time Division duplex Long Term Evolution).
WiFi belongs to short-distance wireless transmission technology, and the mobile terminal can help a user to receive and send e-mails, browse webpages, access streaming media and the like through the WiFi module 102, and provides wireless broadband internet access for the user. Although fig. 1 shows the WiFi module 102, it is understood that it does not belong to the essential constitution of the mobile terminal, and may be omitted entirely as needed within the scope not changing the essence of the invention.
The audio output unit 103 may convert audio data received by the radio frequency unit 101 or the WiFi module 102 or stored in the memory 109 into an audio signal and output as sound when the mobile terminal 100 is in a call signal reception mode, a call mode, a recording mode, a voice recognition mode, a broadcast reception mode, or the like. Also, the audio output unit 103 may also provide audio output related to a specific function performed by the mobile terminal 100 (e.g., a call signal reception sound, a message reception sound, etc.). The audio output unit 103 may include a speaker, a buzzer, and the like.
The a/V input unit 104 is used to receive audio or video signals. The a/V input Unit 104 may include a Graphics Processing Unit (GPU) 1041 and a microphone 1042, the Graphics processor 1041 Processing image data of still pictures or video obtained by an image capturing device (e.g., a camera) in a video capturing mode or an image capturing mode. The processed image frames may be displayed on the display unit 106. The image frames processed by the graphic processor 1041 may be stored in the memory 109 (or other storage medium) or transmitted via the radio frequency unit 101 or the WiFi module 102. The microphone 1042 may receive sounds (audio data) via the microphone 1042 in a phone call mode, a recording mode, a voice recognition mode, or the like, and may be capable of processing such sounds into audio data. The processed audio (voice) data may be converted into a format output transmittable to a mobile communication base station via the radio frequency unit 101 in case of a phone call mode. The microphone 1042 may implement various types of noise cancellation (or suppression) algorithms to cancel (or suppress) noise or interference generated in the course of receiving and transmitting audio signals.
The mobile terminal 100 also includes at least one sensor 105, such as a light sensor, a motion sensor, and other sensors. Specifically, the light sensor includes an ambient light sensor that can adjust the brightness of the display panel 1061 according to the brightness of ambient light, and a proximity sensor that can turn off the display panel 1061 and/or a backlight when the mobile terminal 100 is moved to the ear. As one of the motion sensors, the accelerometer sensor can detect the magnitude of acceleration in each direction (generally, three axes), can detect the magnitude and direction of gravity when stationary, and can be used for applications of recognizing the posture of a mobile phone (such as horizontal and vertical screen switching, related games, magnetometer posture calibration), vibration recognition related functions (such as pedometer and tapping), and the like; as for other sensors such as a fingerprint sensor, a pressure sensor, an iris sensor, a molecular sensor, a gyroscope, a barometer, a hygrometer, a thermometer, and an infrared sensor, which can be configured on the mobile phone, further description is omitted here.
The display unit 106 is used to display information input by a user or information provided to the user. The Display unit 106 may include a Display panel 1061, and the Display panel 1061 may be configured in the form of a Liquid Crystal Display (LCD), an Organic Light-Emitting Diode (OLED), or the like.
The user input unit 107 may be used to receive input numeric or character information and generate key signal inputs related to user settings and function control of the mobile terminal. Specifically, the user input unit 107 may include a touch panel 1071 and other input devices 1072. The touch panel 1071, also referred to as a touch screen, may collect a touch operation performed by a user on or near the touch panel 1071 (e.g., an operation performed by the user on or near the touch panel 1071 using a finger, a stylus, or any other suitable object or accessory), and drive a corresponding connection device according to a predetermined program. The touch panel 1071 may include two parts of a touch detection device and a touch controller. The touch detection device detects the touch direction of a user, detects a signal brought by touch operation and transmits the signal to the touch controller; the touch controller receives touch information from the touch sensing device, converts the touch information into touch point coordinates, sends the touch point coordinates to the processor 110, and can receive and execute commands sent by the processor 110. In addition, the touch panel 1071 may be implemented in various types, such as a resistive type, a capacitive type, an infrared ray, and a surface acoustic wave. In addition to the touch panel 1071, the user input unit 107 may include other input devices 1072. In particular, other input devices 1072 may include, but are not limited to, one or more of a physical keyboard, function keys (e.g., volume control keys, switch keys, etc.), a trackball, a mouse, a joystick, and the like, and are not limited to these specific examples.
Further, the touch panel 1071 may cover the display panel 1061, and when the touch panel 1071 detects a touch operation thereon or nearby, the touch panel 1071 transmits the touch operation to the processor 110 to determine the type of the touch event, and then the processor 110 provides a corresponding visual output on the display panel 1061 according to the type of the touch event. Although the touch panel 1071 and the display panel 1061 are shown in fig. 1 as two separate components to implement the input and output functions of the mobile terminal, in some embodiments, the touch panel 1071 and the display panel 1061 may be integrated to implement the input and output functions of the mobile terminal, and is not limited herein.
The interface unit 108 serves as an interface through which at least one external device is connected to the mobile terminal 100. For example, the external device may include a wired or wireless headset port, an external power supply (or battery charger) port, a wired or wireless data port, a memory card port, a port for connecting a device having an identification module, an audio input/output (I/O) port, a video I/O port, an earphone port, and the like. The interface unit 108 may be used to receive input (e.g., data information, power, etc.) from external devices and transmit the received input to one or more elements within the mobile terminal 100 or may be used to transmit data between the mobile terminal 100 and external devices.
The memory 109 may be used to store software programs as well as various data. The memory 109 may mainly include a storage program area and a storage data area, wherein the storage program area may store an operating system, an application program required by at least one function (such as a sound playing function, an image playing function, etc.), and the like; the storage data area may store data (such as audio data, a phonebook, etc.) created according to the use of the cellular phone, and the like. Further, the memory 109 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.
The processor 110 is a control center of the mobile terminal, connects various parts of the entire mobile terminal using various interfaces and lines, and performs various functions of the mobile terminal and processes data by operating or executing software programs and/or modules stored in the memory 109 and calling data stored in the memory 109, thereby performing overall monitoring of the mobile terminal. Processor 110 may include one or more processing units; preferably, the processor 110 may integrate an application processor, which mainly handles operating systems, user interfaces, application programs, etc., and a modem processor, which mainly handles wireless communications. It will be appreciated that the modem processor described above may not be integrated into the processor 110.
The mobile terminal 100 may further include a power supply 111 (e.g., a battery) for supplying power to various components, and preferably, the power supply 111 may be logically connected to the processor 110 via a power management system, so as to manage charging, discharging, and power consumption management functions via the power management system.
Although not shown in fig. 1, the mobile terminal 100 may further include a bluetooth module or the like, which is not described in detail herein.
In order to facilitate understanding of the embodiments of the present invention, a communication network system on which the mobile terminal of the present invention is based is described below.
Referring to fig. 2, fig. 2 is an architecture diagram of a communication Network system according to an embodiment of the present invention, where the communication Network system is an LTE system of a universal mobile telecommunications technology, and the LTE system includes a UE (User Equipment) 201, an E-UTRAN (Evolved UMTS Terrestrial Radio Access Network) 202, an EPC (Evolved Packet Core) 203, and an IP service 204 of an operator, which are in communication connection in sequence.
Specifically, the UE201 may be the terminal 100 described above, and is not described herein again.
The E-UTRAN202 includes eNodeB2021 and other eNodeBs 2022, among others. Among them, the eNodeB2021 may be connected with other eNodeB2022 through backhaul (e.g., X2 interface), the eNodeB2021 is connected to the EPC203, and the eNodeB2021 may provide the UE201 access to the EPC 203.
The EPC203 may include an MME (Mobility Management Entity) 2031, an HSS (Home Subscriber Server) 2032, other MMEs 2033, an SGW (Serving gateway) 2034, a PGW (PDN gateway) 2035, and a PCRF (Policy and Charging Rules Function) 2036, and the like. The MME2031 is a control node that handles signaling between the UE201 and the EPC203, and provides bearer and connection management. HSS2032 is used to provide registers to manage functions such as home location register (not shown) and holds subscriber specific information about service characteristics, data rates, etc. All user data may be sent through SGW2034, PGW2035 may provide IP address assignment for UE201 and other functions, and PCRF2036 is a policy and charging control policy decision point for traffic data flow and IP bearer resources, which selects and provides available policy and charging control decisions for a policy and charging enforcement function (not shown).
The IP services 204 may include the internet, intranets, IMS (IP Multimedia Subsystem), or other IP services, among others.
Although the LTE system is described as an example, it should be understood by those skilled in the art that the present invention is not limited to the LTE system, but may also be applied to other wireless communication systems, such as GSM, CDMA2000, WCDMA, TD-SCDMA, and future new network systems.
Based on the above mobile terminal hardware structure and communication network system, the present invention provides various embodiments of the method.
Example one
Fig. 3 is a flowchart of a first embodiment of a method for reducing audio delay according to the present invention, and the method for reducing audio delay includes the steps of:
s1, storing audio data to be played into an audio queue to be played;
s2, acquiring the audio data to be played from the audio queue to be played, sending the audio data to be played to a sound card for playing, and deleting the audio data to be played, which is sent to the sound card for playing, from the audio queue to be played;
s3, detecting whether the data volume of the audio data to be played remaining in the audio queue to be played is larger than a preset threshold value or not;
and S4, if so, carrying out accelerated playing on the remaining audio data to be played according to a preset multiple.
In this embodiment, first, the audio data to be played is stored in the audio queue to be played; then, the audio data to be played is acquired from the audio queue to be played, sent to a sound card for playing, and the audio data to be played, which is sent to the sound card for playing, is deleted from the audio queue to be played; then, detecting whether the data volume of the audio data to be played remaining in the audio queue to be played is larger than a preset threshold value; and finally, if so, carrying out accelerated playing on the residual audio data to be played according to a preset multiple.
In this embodiment, it should be noted that, based on the prior art, a delay may occur in a screen projection process, and the delay may be continuously accumulated in the screen projection process, and when the delay is accumulated to a certain amount, a problem of obvious asynchronism between sound and picture may occur, which affects screen projection quality and user experience. Therefore, in the present embodiment, in order to solve the above technical problem, the audio data to be played is stored in the audio queue to be played; acquiring the audio data to be played from the audio queue to be played, sending the audio data to be played to a sound card for playing, and deleting the audio data to be played, which is sent to the sound card for playing, from the audio queue to be played; detecting whether the data volume of the audio data to be played remaining in the audio queue to be played is larger than a preset threshold value or not; if yes, the remaining audio data to be played are accelerated and played according to preset multiples. The residual audio data to be played, which are stored in the audio queue to be played but not sent to the sound card for playing, are played in an accelerated manner, so that the playing time is saved, and the audio delay is reduced, thereby realizing a scheme for reducing the audio delay, solving the problem of unsynchronized sound and picture in the screen projection process, and improving the overall quality and user experience of the screen projection.
Specifically, in this embodiment, in the step of storing the audio data to be played into the audio queue to be played, the format of the audio data to be played includes an audio data format that can be identified and played by the sound card, and the specific format of the audio data is not limited in this embodiment.
Optionally, the audio data to be played is audio data in a PCM (pulse code modulation) format.
Specifically, in this embodiment, in the step of obtaining the audio data to be played from the audio queue to be played, sending the audio data to be played to a sound card for playing, and deleting the audio data to be played, which is sent to the sound card for playing, from the audio queue to be played, the audio data to be played, which is sent to the sound card for playing, is deleted from the audio queue to be played, so that only the audio data to be played, which is to be played waiting for playing but cannot be played actually, is left in the audio queue to be played, that is, delayed audio data is generated, so as to detect whether the data amount of the audio data to be played remaining in the audio queue to be played is greater than a preset threshold value or not.
Specifically, in this embodiment, the audio delay can be found and quantified in time through the step of detecting whether the data amount of the remaining audio data to be played in the audio queue to be played is greater than the preset threshold, so that when the accumulation of the remaining audio data to be played reaches the preset threshold, the audio delay is reduced by performing accelerated playing on the remaining audio data to be played according to a preset multiple. It will be appreciated that the preset threshold may be set according to the specific audio delay requirements.
Optionally, the detecting whether the data amount of the remaining audio data to be played in the audio queue to be played is greater than a preset threshold is performed in real time. By detecting whether the data quantity of the audio data to be played remaining in the audio queue to be played is larger than a preset threshold value or not in real time, the problem of audio delay can be found more timely, and the sensitivity of audio delay detection is improved.
Specifically, in this embodiment, if yes, the step of performing accelerated playing on the remaining audio data to be played according to a preset multiple accelerates and then plays the remaining audio data to be played that is not played in time due to audio delay, so that the audio playing time is saved, and the audio delay is reduced. For example, when the preset multiple is 2 times, the remaining audio data to be played is played at twice speed, and accordingly, half of the playing time is saved, thereby reducing the audio delay. It can be understood that the specific value of the preset multiple may be determined according to the requirement of the specific application environment on the audio delay, and the specific value is not limited in this embodiment.
The method has the advantages that the audio data to be played are stored in the audio queue to be played; acquiring the audio data to be played from the audio queue to be played, sending the audio data to be played to a sound card for playing, and deleting the audio data to be played, which is sent to the sound card for playing, from the audio queue to be played; detecting whether the data volume of the audio data to be played remaining in the audio queue to be played is larger than a preset threshold value or not; if yes, the remaining audio data to be played are accelerated and played according to preset multiples. The scheme for reducing the audio delay is realized, the problem of unsynchronized sound and picture in the screen projection process is solved, and the overall quality and the user experience of the screen projection are improved.
Example two
Based on the above embodiment, in order to more conveniently implement the accelerated playing of the remaining audio data to be played according to the preset multiple, as shown in fig. 4, the step of accelerated playing the remaining audio data to be played according to the preset multiple includes the following steps:
s41, calling audio frequency speed change software to accelerate the rest audio frequency data to be played according to preset multiples to generate acceleration result data;
and S42, sending the acceleration result data to a sound card for playing.
Specifically, in this embodiment, the step of calling the audio speed changing software to accelerate the remaining audio data to be played according to a preset multiple and generating acceleration result data realizes the function of accelerating the remaining audio data to be played according to the preset multiple. The audio frequency speed changing software is software capable of accelerating the input residual audio frequency data to be played, and for convenience of use, the audio frequency speed changing software can be existing audio frequency speed changing software, such as soundmouth software, the soundmouth software is an open-source audio frequency processing library, and the audio frequency speed changing software mainly realizes three functional modules including speed changing, tone changing, simultaneous speed changing and tone changing, and can operate on media streams in real time and audio frequency files. The present embodiment does not limit the specific type of audio frequency shifting software. It can be understood that after the step of calling audio frequency speed change software to accelerate the remaining audio frequency data to be played according to the preset multiple and generate the acceleration result data, the acceleration result data can be sent to the sound card to be played, so as to achieve the purpose of reducing the audio frequency delay.
The method has the advantages that the audio data to be played are stored in the audio queue to be played; acquiring the audio data to be played from the audio queue to be played, sending the audio data to be played to a sound card for playing, and deleting the audio data to be played, which is sent to the sound card for playing, from the audio queue to be played; detecting whether the data volume of the audio data to be played remaining in the audio queue to be played is larger than a preset threshold value or not; if so, calling audio frequency speed change software to accelerate the remaining audio frequency data to be played according to a preset multiple to generate acceleration result data; and sending the acceleration result data to a sound card for playing. The scheme for reducing the audio delay is realized, the problem of unsynchronized sound and picture in the screen projection process is solved, and the overall quality and the user experience of the screen projection are improved.
EXAMPLE III
Based on the above embodiment, in order to more conveniently realize that the calling audio frequency speed changing software accelerates the remaining audio data to be played by a preset multiple to generate acceleration result data, as shown in fig. 5, the step of the calling audio frequency speed changing software accelerating the remaining audio data to be played by a preset multiple to generate acceleration result data includes the following steps:
s411, integrating the rest audio data to be played to an audio buffer to be accelerated;
and S412, the audio frequency speed changing software calls the data in the audio frequency buffer to be accelerated and accelerates the data according to a preset multiple to generate acceleration result data, and the acceleration result data is stored in the acceleration result audio frequency buffer.
Specifically, in this embodiment, the remaining audio data to be played is integrated into an audio buffer to be accelerated; and calling the data in the audio buffer to be accelerated by the audio speed change software, accelerating the data according to a preset multiple to generate acceleration result data, and storing the acceleration result data into the acceleration result audio buffer, so that the acceleration of the remaining audio data to be played is realized, and the acceleration result data is sent to a sound card to be played subsequently, so that the audio playing time can be saved, and the audio delay is reduced.
The method has the advantages that the audio data to be played are stored in the audio queue to be played; acquiring the audio data to be played from the audio queue to be played, sending the audio data to be played to a sound card for playing, and deleting the audio data to be played, which is sent to the sound card for playing, from the audio queue to be played; detecting whether the data volume of the audio data to be played remaining in the audio queue to be played is larger than a preset threshold value or not; if so, integrating the remaining audio data to be played to an audio buffer to be accelerated; the audio frequency speed changing software calls the data in the audio frequency buffer to be accelerated and accelerates the data according to a preset multiple to generate acceleration result data, and the acceleration result data is stored in the acceleration result audio frequency buffer; and sending the acceleration result data to a sound card for playing. The scheme for reducing the audio delay is realized, the problem of unsynchronized sound and picture in the screen projection process is solved, and the overall quality and the user experience of the screen projection are improved.
Example four
Based on the above embodiment, in order to obtain the degree of audio delay in time, as shown in fig. 6, after the step of detecting whether the data amount of the audio data to be played remaining in the audio queue to be played is greater than a preset threshold, the method further includes the following steps:
and S5, if not, returning to the step of detecting whether the data volume of the audio data to be played remaining in the audio queue to be played is larger than a preset threshold value.
Specifically, in this embodiment, it can be understood that the sequence of the step S4 and the step S5 is determined according to the detection result of the step S3, and if the detection result indicates that the data amount of the audio data to be played remaining in the audio queue to be played is greater than the preset threshold, the step S4 is executed first; otherwise, step S5 is executed first.
Specifically, in this embodiment, when the detection result is that the data amount of the remaining audio data to be played in the audio queue to be played is not greater than the preset threshold, the step S3 is returned to, and whether the data amount of the remaining audio data to be played in the audio queue to be played is greater than the preset threshold is continuously detected, so that the degree of audio delay can be obtained in time, and the audio delay is further reduced in time.
The method has the advantages that the audio data to be played are stored in the audio queue to be played; acquiring the audio data to be played from the audio queue to be played, sending the audio data to be played to a sound card for playing, and deleting the audio data to be played, which is sent to the sound card for playing, from the audio queue to be played; detecting whether the data volume of the audio data to be played remaining in the audio queue to be played is larger than a preset threshold value or not; if not, returning to the step of detecting whether the data volume of the audio data to be played remaining in the audio queue to be played is larger than a preset threshold value; if yes, the remaining audio data to be played are accelerated and played according to preset multiples. The scheme for reducing the audio delay is realized, the problem of unsynchronized sound and picture in the screen projection process is solved, and the overall quality and the user experience of the screen projection are improved.
EXAMPLE five
Based on the above embodiment, in order to obtain the audio data to be played, as shown in fig. 7, before the step of storing the audio data to be played into the audio queue to be played, the method further includes the following steps:
s01, receiving coded audio data;
s02, decoding the coded audio data to generate audio data to be played.
Specifically, in this embodiment, the step of receiving encoded audio data is to obtain the encoded audio data from a network or a hardware device communicatively connected to the network by means of wired communication or wireless communication, where the encoded audio data refers to data that can generate audio data to be played by decoding, for example, audio data in an ACC (advanced audio coding) format. The data format of the encoded audio data is not limited in this embodiment.
Specifically, in this embodiment, the step of decoding the encoded audio data to generate the audio data to be played includes selecting a corresponding decoding method for decoding the received encoded audio data according to the data format of the encoded audio data to generate the audio data to be played.
The present embodiment has an advantageous effect in that by receiving encoded audio data; decoding the encoded audio data to generate audio data to be played; storing the audio data to be played into an audio queue to be played; acquiring the audio data to be played from the audio queue to be played, sending the audio data to be played to a sound card for playing, and deleting the audio data to be played, which is sent to the sound card for playing, from the audio queue to be played; detecting whether the data volume of the audio data to be played remaining in the audio queue to be played is larger than a preset threshold value or not; if yes, the remaining audio data to be played are accelerated and played according to preset multiples. The scheme for reducing the audio delay is realized, the problem of unsynchronized sound and picture in the screen projection process is solved, and the overall quality and the user experience of the screen projection are improved.
EXAMPLE six
Based on the above embodiments, the present invention further provides an apparatus for reducing audio latency, the apparatus comprising a memory, a processor, and a computer program stored on the memory and executable on the processor, wherein when the computer program is executed by the processor, the apparatus implements:
storing the audio data to be played into an audio queue to be played;
acquiring the audio data to be played from the audio queue to be played, sending the audio data to be played to a sound card for playing, and deleting the audio data to be played, which is sent to the sound card for playing, from the audio queue to be played;
detecting whether the data volume of the audio data to be played remaining in the audio queue to be played is larger than a preset threshold value or not;
if yes, the remaining audio data to be played are accelerated and played according to preset multiples.
In this embodiment, first, the audio data to be played is stored in the audio queue to be played; then, the audio data to be played is acquired from the audio queue to be played, sent to a sound card for playing, and the audio data to be played, which is sent to the sound card for playing, is deleted from the audio queue to be played; then, detecting whether the data volume of the audio data to be played remaining in the audio queue to be played is larger than a preset threshold value; and finally, if so, carrying out accelerated playing on the residual audio data to be played according to a preset multiple.
In this embodiment, it should be noted that, based on the prior art, a delay may occur in a screen projection process, and the delay may be continuously accumulated in the screen projection process, and when the delay is accumulated to a certain amount, a problem of obvious asynchronism between sound and picture may occur, which affects screen projection quality and user experience. Therefore, in the present embodiment, in order to solve the above technical problem, the audio data to be played is stored in the audio queue to be played; acquiring the audio data to be played from the audio queue to be played, sending the audio data to be played to a sound card for playing, and deleting the audio data to be played, which is sent to the sound card for playing, from the audio queue to be played; detecting whether the data volume of the audio data to be played remaining in the audio queue to be played is larger than a preset threshold value or not; if yes, the remaining audio data to be played are accelerated and played according to preset multiples. The residual audio data to be played, which are stored in the audio queue to be played but not sent to the sound card for playing, are played in an accelerated manner, so that the playing time is saved, and the audio delay is reduced, thereby realizing a scheme for reducing the audio delay, solving the problem of unsynchronized sound and picture in the screen projection process, and improving the overall quality and user experience of the screen projection.
Specifically, in this embodiment, in the step of storing the audio data to be played into the audio queue to be played, the format of the audio data to be played includes an audio data format that can be identified and played by the sound card, and the specific format of the audio data is not limited in this embodiment.
Optionally, the audio data to be played is audio data in a PCM format.
Specifically, in this embodiment, in the step of obtaining the audio data to be played from the audio queue to be played, sending the audio data to be played to a sound card for playing, and deleting the audio data to be played, which is sent to the sound card for playing, from the audio queue to be played, the audio data to be played, which is sent to the sound card for playing, is deleted from the audio queue to be played, so that only the audio data to be played, which is to be played waiting for playing but cannot be played actually, is left in the audio queue to be played, that is, delayed audio data is generated, so as to detect whether the data amount of the audio data to be played remaining in the audio queue to be played is greater than a preset threshold value or not.
Specifically, in this embodiment, the audio delay can be found and quantified in time through the step of detecting whether the data amount of the remaining audio data to be played in the audio queue to be played is greater than the preset threshold, so that when the accumulation of the remaining audio data to be played reaches the preset threshold, the audio delay is reduced by performing accelerated playing on the remaining audio data to be played according to a preset multiple. It will be appreciated that the preset threshold may be set according to the specific audio delay requirements.
Optionally, the detecting whether the data amount of the remaining audio data to be played in the audio queue to be played is greater than a preset threshold is performed in real time. By detecting whether the data quantity of the audio data to be played remaining in the audio queue to be played is larger than a preset threshold value or not in real time, the problem of audio delay can be found more timely, and the sensitivity of audio delay detection is improved.
Specifically, in this embodiment, if yes, the step of performing accelerated playing on the remaining audio data to be played according to a preset multiple accelerates and then plays the remaining audio data to be played that is not played in time due to audio delay, so that the audio playing time is saved, and the audio delay is reduced. For example, when the preset multiple is 2 times, the remaining audio data to be played is played at twice speed, and accordingly, half of the playing time is saved, thereby reducing the audio delay. It can be understood that the specific value of the preset multiple may be determined according to the requirement of the specific application environment on the audio delay, and the specific value is not limited in this embodiment.
The method has the advantages that the audio data to be played are stored in the audio queue to be played; acquiring the audio data to be played from the audio queue to be played, sending the audio data to be played to a sound card for playing, and deleting the audio data to be played, which is sent to the sound card for playing, from the audio queue to be played; detecting whether the data volume of the audio data to be played remaining in the audio queue to be played is larger than a preset threshold value or not; if yes, the remaining audio data to be played are accelerated and played according to preset multiples. The scheme for reducing the audio delay is realized, the problem of unsynchronized sound and picture in the screen projection process is solved, and the overall quality and the user experience of the screen projection are improved.
EXAMPLE seven
Based on the above embodiments, the computer program when executed by the processor implements:
calling audio frequency speed change software to accelerate the rest audio frequency data to be played according to preset multiples to generate acceleration result data;
and sending the acceleration result data to a sound card for playing.
Specifically, in this embodiment, the step of calling the audio speed changing software to accelerate the remaining audio data to be played according to a preset multiple and generating acceleration result data realizes the function of accelerating the remaining audio data to be played according to the preset multiple. The audio frequency speed changing software may be software capable of accelerating the input remaining audio frequency data to be played, and for convenience of use, the audio frequency speed changing software may be existing audio frequency speed changing software, such as soundport software. The present embodiment does not limit the specific type of audio frequency shifting software. It can be understood that after the step of calling audio frequency speed change software to accelerate the remaining audio frequency data to be played according to the preset multiple and generate the acceleration result data, the acceleration result data can be sent to the sound card to be played, so as to achieve the purpose of reducing the audio frequency delay.
The method has the advantages that the audio data to be played are stored in the audio queue to be played; acquiring the audio data to be played from the audio queue to be played, sending the audio data to be played to a sound card for playing, and deleting the audio data to be played, which is sent to the sound card for playing, from the audio queue to be played; detecting whether the data volume of the audio data to be played remaining in the audio queue to be played is larger than a preset threshold value or not; if so, calling audio frequency speed change software to accelerate the remaining audio frequency data to be played according to a preset multiple to generate acceleration result data; and sending the acceleration result data to a sound card for playing. The scheme for reducing the audio delay is realized, the problem of unsynchronized sound and picture in the screen projection process is solved, and the overall quality and the user experience of the screen projection are improved.
Example eight
Based on the above embodiments, the computer program when executed by the processor implements:
integrating the rest audio data to be played to an audio buffer to be accelerated;
and the audio frequency speed changing software calls the data in the audio frequency buffer to be accelerated and accelerates the data according to a preset multiple to generate acceleration result data, and the acceleration result data is stored in the acceleration result audio frequency buffer.
Specifically, in this embodiment, the remaining audio data to be played is integrated into an audio buffer to be accelerated; and calling the data in the audio buffer to be accelerated by the audio speed change software, accelerating the data according to a preset multiple to generate acceleration result data, and storing the acceleration result data into the acceleration result audio buffer, so that the acceleration of the remaining audio data to be played is realized, and the acceleration result data is sent to a sound card to be played subsequently, so that the audio playing time can be saved, and the audio delay is reduced.
The method has the advantages that the audio data to be played are stored in the audio queue to be played; acquiring the audio data to be played from the audio queue to be played, sending the audio data to be played to a sound card for playing, and deleting the audio data to be played, which is sent to the sound card for playing, from the audio queue to be played; detecting whether the data volume of the audio data to be played remaining in the audio queue to be played is larger than a preset threshold value or not; if so, integrating the remaining audio data to be played to an audio buffer to be accelerated; the audio frequency speed changing software calls the data in the audio frequency buffer to be accelerated and accelerates the data according to a preset multiple to generate acceleration result data, and the acceleration result data is stored in the acceleration result audio frequency buffer; and sending the acceleration result data to a sound card for playing. The scheme for reducing the audio delay is realized, the problem of unsynchronized sound and picture in the screen projection process is solved, and the overall quality and the user experience of the screen projection are improved.
Example nine
Based on the above embodiments, the computer program when executed by the processor implements:
if not, returning to the step of detecting whether the data volume of the audio data to be played remaining in the audio queue to be played is larger than a preset threshold value.
Specifically, in this embodiment, it can be understood that, if the number of the audio data to be played is greater than the preset number, the step of performing accelerated playing on the remaining audio data to be played according to the preset number is performed, and if the number of the audio data to be played remaining in the audio queue to be played is not greater than the preset threshold, the step of returning to the step of detecting whether the data amount of the audio data to be played remaining in the audio queue to be played is greater than the preset threshold is performed according to the sequence of the step of detecting whether the data amount of the audio data to be played remaining in the audio queue to be played is greater than the preset threshold, and if the detection result is that the data amount of the audio data to be played remaining in the audio queue to be played is greater than the preset threshold, the step of performing accelerated playing on the remaining audio data to be played according to the preset number is performed first; otherwise, if not, returning to the step of detecting whether the data volume of the audio data to be played remaining in the audio queue to be played is larger than a preset threshold value.
Specifically, in this embodiment, when the detection result is that the data amount of the remaining audio data to be played in the audio queue to be played is not greater than the preset threshold, the step of detecting whether the data amount of the remaining audio data to be played in the audio queue to be played is greater than the preset threshold is returned, and whether the data amount of the remaining audio data to be played in the audio queue to be played is greater than the preset threshold is continuously detected, so that the degree of audio delay can be obtained in time, and the audio delay is further reduced in time.
In another embodiment, before storing the audio data to be played in the audio queue to be played:
receiving encoded audio data;
and decoding the encoded audio data to generate audio data to be played.
Specifically, in this embodiment, the step of receiving encoded audio data is to obtain the encoded audio data from a network or a hardware device communicatively connected to the network by means of wired communication or wireless communication, where the encoded audio data refers to data that can generate audio data to be played by decoding, for example, audio data in an ACC (advanced audio coding) format. The data format of the encoded audio data is not limited in this embodiment.
Specifically, in this embodiment, the step of decoding the encoded audio data to generate the audio data to be played includes selecting a corresponding decoding method for decoding the received encoded audio data according to the data format of the encoded audio data to generate the audio data to be played.
The present embodiment has an advantageous effect in that by receiving encoded audio data; decoding the encoded audio data to generate audio data to be played; storing the audio data to be played into an audio queue to be played; acquiring the audio data to be played from the audio queue to be played, sending the audio data to be played to a sound card for playing, and deleting the audio data to be played, which is sent to the sound card for playing, from the audio queue to be played; detecting whether the data volume of the audio data to be played remaining in the audio queue to be played is larger than a preset threshold value or not; if not, returning to the step of detecting whether the data volume of the audio data to be played remaining in the audio queue to be played is larger than a preset threshold value; if yes, the remaining audio data to be played are accelerated and played according to preset multiples. The scheme for reducing the audio delay is realized, the problem of unsynchronized sound and picture in the screen projection process is solved, and the overall quality and the user experience of the screen projection are improved.
Example ten
Based on the above embodiments, the present invention further provides a computer readable storage medium, where a program for reducing audio latency is stored, and when the program for reducing audio latency is executed by a processor, the method for reducing audio latency according to any one of the above embodiments is implemented.
The invention provides a method, equipment and a computer readable storage medium for reducing audio delay, which are characterized in that audio data to be played are stored in an audio queue to be played; acquiring the audio data to be played from the audio queue to be played, sending the audio data to be played to a sound card for playing, and deleting the audio data to be played, which is sent to the sound card for playing, from the audio queue to be played; detecting whether the data volume of the audio data to be played remaining in the audio queue to be played is larger than a preset threshold value or not; if yes, the remaining audio data to be played are accelerated and played according to preset multiples. The scheme for reducing the audio delay is realized, the problem of unsynchronized sound and picture in the screen projection process is solved, and the overall quality and the user experience of the screen projection are improved.
It should be noted that, in this document, the terms "comprises," "comprising," or any other variation thereof, are intended to cover a non-exclusive inclusion, such that a process, method, article, or apparatus that comprises a list of elements does not include only those elements but may include other elements not expressly listed or inherent to such process, method, article, or apparatus. Without further limitation, an element defined by the phrase "comprising an … …" does not exclude the presence of other like elements in a process, method, article, or apparatus that comprises the element.
The above-mentioned serial numbers of the embodiments of the present invention are merely for description and do not represent the merits of the embodiments.
Through the above description of the embodiments, those skilled in the art will clearly understand that the method of the above embodiments can be implemented by software plus a necessary general hardware platform, and certainly can also be implemented by hardware, but in many cases, the former is a better implementation manner. Based on such understanding, the technical solutions of the present invention may be embodied in the form of a software product, which is stored in a storage medium (such as ROM/RAM, magnetic disk, optical disk) and includes instructions for enabling a terminal (such as a mobile phone, a computer, a server, an air conditioner, or a network device) to execute the method according to the embodiments of the present invention.
While the present invention has been described with reference to the embodiments shown in the drawings, the present invention is not limited to the embodiments, which are illustrative and not restrictive, and it will be apparent to those skilled in the art that various changes and modifications can be made therein without departing from the spirit and scope of the invention as defined in the appended claims.

Claims (10)

1. A method for reducing audio latency, the method comprising:
storing the audio data to be played into an audio queue to be played;
acquiring the audio data to be played from the audio queue to be played, sending the audio data to be played to a sound card for playing, and deleting the audio data to be played, which is sent to the sound card for playing, from the audio queue to be played;
detecting whether the data volume of the audio data to be played remaining in the audio queue to be played is larger than a preset threshold value or not;
if yes, the remaining audio data to be played are accelerated and played according to preset multiples.
2. The method of claim 1, wherein the step of accelerating the playing of the remaining audio data to be played by a preset multiple comprises:
calling audio frequency speed change software to accelerate the rest audio frequency data to be played according to preset multiples to generate acceleration result data;
and sending the acceleration result data to a sound card for playing.
3. The method for reducing audio delay according to claim 2, wherein the step of calling audio frequency shifting software to accelerate the remaining audio data to be played according to a preset multiple to generate acceleration result data comprises:
integrating the rest audio data to be played to an audio buffer to be accelerated;
and the audio frequency speed changing software calls the data in the audio frequency buffer to be accelerated and accelerates the data according to a preset multiple to generate acceleration result data, and the acceleration result data is stored in the acceleration result audio frequency buffer.
4. The method of claim 1, further comprising, after the step of detecting whether the data amount of the audio data to be played remaining in the audio queue to be played is greater than a preset threshold, the step of:
if not, returning to the step of detecting whether the data volume of the audio data to be played remaining in the audio queue to be played is larger than a preset threshold value.
5. The method for reducing audio latency according to claim 1, further comprising, before the step of storing the audio data to be played in the audio queue to be played:
receiving encoded audio data;
and decoding the encoded audio data to generate audio data to be played.
6. An apparatus for reducing audio latency, the apparatus comprising a memory, a processor, and a computer program stored on the memory and executable on the processor, the computer program when executed by the processor implementing:
storing the audio data to be played into an audio queue to be played;
acquiring the audio data to be played from the audio queue to be played, sending the audio data to be played to a sound card for playing, and deleting the audio data to be played, which is sent to the sound card for playing, from the audio queue to be played;
detecting whether the data volume of the audio data to be played remaining in the audio queue to be played is larger than a preset threshold value or not;
if yes, the remaining audio data to be played are accelerated and played according to preset multiples.
7. The apparatus for reducing audio latency according to claim 6, wherein the computer program when executed by the processor implements:
calling audio frequency speed change software to accelerate the rest audio frequency data to be played according to preset multiples to generate acceleration result data;
and sending the acceleration result data to a sound card for playing.
8. The apparatus for reducing audio latency of claim 7, wherein the computer program when executed by the processor implements:
integrating the rest audio data to be played to an audio buffer to be accelerated;
and the audio frequency speed changing software calls the data in the audio frequency buffer to be accelerated and accelerates the data according to a preset multiple to generate acceleration result data, and the acceleration result data is stored in the acceleration result audio frequency buffer.
9. The apparatus for reducing audio latency according to claim 6, wherein the computer program when executed by the processor implements:
if not, returning to the step of detecting whether the data volume of the audio data to be played remaining in the audio queue to be played is larger than a preset threshold value;
receiving encoded audio data;
and decoding the encoded audio data to generate audio data to be played.
10. A computer-readable storage medium, on which a program for reducing audio latency is stored, which when executed by a processor implements the steps of the method for reducing audio latency of any one of claims 1 to 5.
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CN113365129B (en) * 2021-08-09 2022-04-26 北京百瑞互联技术有限公司 Bluetooth audio data processing method, transmitter, receiver and transceiving equipment

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