CN109062537B - Audio delay reduction method, device, medium and equipment - Google Patents
Audio delay reduction method, device, medium and equipment Download PDFInfo
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- CN109062537B CN109062537B CN201811004108.2A CN201811004108A CN109062537B CN 109062537 B CN109062537 B CN 109062537B CN 201811004108 A CN201811004108 A CN 201811004108A CN 109062537 B CN109062537 B CN 109062537B
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Abstract
The invention provides a method, a device, a medium and equipment for reducing audio delay. The method comprises the following steps: acquiring input audio data to be processed; storing the audio data to be processed into a buffer area; calling the audio data to be processed from the buffer area and outputting the audio data to an output channel; converting the audio data to be processed into an analog signal played by a loudspeaker by using the output channel; and driving a loudspeaker to play the analog signal. By caching data at the time of writing data, the data volume of waiting for alignment in an output path can be reduced, and further the audio transmission or playing delay can be reduced; in addition, the invention does not need extra hardware cost and has stronger compatibility.
Description
Technical Field
The present invention relates to the field of audio processing technologies, and in particular, to a method, an apparatus, a medium, and a device for reducing audio delay.
Background
In the application scene of digital audio transmission or playing of intelligent products, external input audio data sometimes needs to be played as soon as possible, and when the situation that playing is not timely enough occurs, human ears can feel sound delay (experience data is more than 50 milliseconds), so that experience is seriously influenced. In the prior art, the processing mode for the application is to directly call an output interface of a system to process data after acquiring input audio data so as to drive external devices such as a loudspeaker to sound, and a playing channel is not optimized, so that the effect is poor. In addition, there is an audio scheme that implements low delay in hardware, but this scheme requires hardware cost increase on the smart device and cannot be compatible with the products already sold.
Disclosure of Invention
Aiming at the defects in the prior art, the invention provides a method for reducing audio delay, which can reduce the data volume waiting for alignment in an output channel, and further can reduce the audio transmission or playing delay; in addition, the invention does not need extra hardware cost and has stronger compatibility.
In a first aspect, the present invention provides a method for reducing audio delay, including:
acquiring input audio data to be processed;
storing the audio data to be processed into a buffer area;
calling the audio data to be processed from the buffer area and outputting the audio data to an output channel;
converting the audio data to be processed into an analog signal played by a loudspeaker by using the output channel;
and driving a loudspeaker to play the analog signal.
Optionally, before the step of retrieving the audio data to be processed from the buffer and outputting the audio data to an output path, the method further includes:
calculating the total data volume of each stage of output waiting queues in the output path in real time;
judging whether the total data volume is smaller than a preset threshold value or not;
if the current time is less than the preset time, calling the audio data to be processed from the buffer area and outputting the audio data to an output channel;
if not, waiting for the output path to process the data of the waiting queue; and continuing to execute the step of calculating the total data volume of each stage of output waiting queues in the output path in real time.
Optionally, the retrieving the audio data to be processed from the buffer area and outputting the audio data to an output path includes:
and calling a specified number of audio data to be processed from the buffer area and outputting the audio data to an output path.
Optionally, storing the audio data to be processed in a buffer includes:
storing the audio data to be processed into a buffer area; and if the buffer zone is saturated, discarding the overflowing audio data to be processed.
Optionally, before the step of retrieving the audio data to be processed from the buffer and outputting the audio data to an output path, the method further includes:
judging whether the buffer area has data or not;
if not, the audio data to be processed is called from the buffer area and output to an output path, and the method comprises the following steps: outputting the spoofed data to a first level output in an output path;
if yes, the step of calling the audio data to be processed from the buffer area and outputting the audio data to an output channel is executed.
In a second aspect, the present invention provides an apparatus for reducing audio delay, including:
the data acquisition module is used for acquiring input audio data to be processed;
the data buffer module is used for storing the audio data to be processed into a buffer area;
the data output module is used for calling the audio data to be processed from the buffer area and outputting the audio data to an output channel;
the data conversion module is used for converting the audio data to be processed into an analog signal played by a loudspeaker by using the output channel;
and the audio playing module is used for driving a loudspeaker to play the analog signal.
In a third aspect, the present invention provides a computer-readable storage medium having stored thereon a computer program which, when executed by a processor, implements a method of reducing audio delay in the first aspect.
In a fourth aspect, the present invention provides an audio delay reduction apparatus, comprising: memory, processor and computer program stored on the memory and executable on the processor, the processor implementing a method of audio delay reduction in a first aspect when executing the program.
The invention provides a method for reducing audio delay, which comprises the steps of obtaining audio data to be processed, storing the audio data to be processed in a buffer area, calling the audio data to be processed from the buffer area, processing the audio data to be processed, and caching the data at the moment of writing the data, so that the data volume waiting for alignment in an output path can be reduced, and the audio transmission or playing delay can be further reduced; in addition, the invention does not need extra hardware cost and has stronger compatibility.
The invention provides an audio delay reduction device, a computer readable storage medium and an audio delay reduction device, which have the same beneficial effects with the same inventive concept as the audio delay reduction method.
Drawings
In order to more clearly illustrate the detailed description of the invention or the technical solutions in the prior art, the drawings that are needed in the detailed description of the invention or the prior art will be briefly described below. Throughout the drawings, like elements or portions are generally identified by like reference numerals. In the drawings, elements or portions are not necessarily drawn to scale.
Fig. 1 is a flowchart of a method for reducing audio delay according to the present invention;
FIG. 2 is a flow chart of a data processing process provided by the present invention;
fig. 3 is a schematic diagram of an apparatus for reducing audio delay according to the present invention.
Detailed Description
Embodiments of the present invention will be described in detail below with reference to the accompanying drawings. The following examples are only for illustrating the technical solutions of the present invention more clearly, and therefore are only examples, and the protection scope of the present invention is not limited thereby.
It is to be noted that, unless otherwise specified, technical or scientific terms used herein shall have the ordinary meaning as understood by those skilled in the art to which the invention pertains.
The invention provides a method, a device, a medium and equipment for reducing audio delay. Embodiments of the present invention will be described below with reference to the drawings.
In the process of audio transmission and playing, the conventional audio data path is as follows: mic- > 1-level output- > 2-level output- > 3-level output- > … … - > horn- > human ear. Where, Mic represents an output path for acquiring audio data from the outside, and 1-stage output, 2-stage output, etc. are output as processing audio data, and the audio data acquired from the outside can be converted into an analog signal that can be played by a speaker through the one-stage output.
Audio delay refers to the time difference from the point in time when the sound is produced to the point in time when it is played out on the speaker of the smart device to be heard by the human ear.
In the output path, each stage of output will have the audio data queued for processing, that is, the data amount of the queue waiting at each stage is: BufSize _ o1, BufSize _ o2, BufSize _ o3, etc., the amount of waiting queue data for all outputs being: BufSize _ queue ═ BufSize _ o1+ BufSize _ o2+ BufSize _ o3+ … …, so that the amount of delay data BufSize _ lat is equal to the amount of wait queue data BufSize _ queue, with the conventional approach of delay time: BufSize _ lat/sample rate.
In the invention, a buffer area is added in a data path, and the data path is as follows: mic- > buffer- > 1-level output- > 2-level output- > 3-level output- > … … - > horn- > human ear. The output path is: the 1-stage output- > 2-stage output- > 3-stage output- > … …. Since the buffer can buffer Mic data, the amount of delay data is from Mic to buffer to being heard by human ears, and its value is BufSize _ cache + BufSize _ queue, where BufSize _ cache is the amount of data buffered in the buffer, BufSize _ queue is the amount of data in the waiting queue of all outputs, and the amount of delay data BufSize _ lat is BufSize _ cache + BufSize _ queue, and the delay time is: BufSize _ lat/sample rate.
Referring to fig. 1, fig. 1 is a flowchart of a method for reducing audio delay according to an embodiment of the present invention, where the method for reducing audio delay according to the embodiment includes:
step S101: and acquiring input audio data to be processed.
Step S102: and storing the audio data to be processed into a buffer area.
Step S103: and calling the audio data to be processed from the buffer area and outputting the audio data to an output channel.
Step S104: and converting the audio data to be processed into an analog signal played by a loudspeaker by using the output channel.
Step S105: and driving a loudspeaker to play the analog signal.
The execution subject of the invention is a processor, a processing chip and the like.
According to the invention, the audio data to be processed is obtained and stored in the buffer area, the audio data to be processed is called from the buffer area and then processed, and the data is buffered at the moment of writing the data, so that the data volume waiting for alignment in an output path can be reduced, and further the audio transmission or playing delay can be reduced; in addition, the invention does not need extra hardware cost and has stronger compatibility.
The audio data to be processed refers to externally input audio data for digital audio transmission or playing of intelligent products. The intelligent product can be a karaoke host, a television and the like.
When the playing is not timely enough, the human ear can feel the sound delay (the experience data is more than 50 milliseconds), and the experience is seriously influenced.
Wherein, the output path is a path for processing data. The output path comprises a plurality of links, wherein the first link is the first-level output, the second link is the second-level output, and the like. Through multi-stage output, the audio data to be processed can be converted into analog signals played by a loudspeaker.
For example, the output path includes 3 stages of outputs: 1-stage output- > 2-stage output- > 3-stage output.
After the audio data to be processed is called from the buffer area, the audio data to be processed can be output to an output channel by using an output interface for data conversion.
In a specific embodiment provided by the present invention, before the step of retrieving the audio data to be processed from the buffer and outputting the audio data to an output path, the method may further include: calculating the total data volume of each stage of output waiting queues in the output path in real time; judging whether the total data volume is smaller than a preset threshold value or not; if the current time is less than the preset time, calling the audio data to be processed from the buffer area and outputting the audio data to an output channel; if not, waiting for the output path to process the data of the waiting queue; and continuing to execute the step of calculating the total data volume of each stage of output waiting queues in the output path in real time.
Before the audio data to be processed is called from the buffer, the total data volume of each level of output waiting queues in the output path needs to be calculated in real time, whether the total data volume is smaller than a preset threshold value or not is judged, if so, the audio data to be processed is called from the buffer, and the audio data to be processed is output to the output path; if not, waiting is needed, and when the total data volume is smaller than the preset threshold value according to the calculation result, the audio data to be processed is called from the buffer area.
The preset threshold value can be set according to needs.
For example, the amount of data that the buffer can store is 10ms, and the total amount of data of the waiting queue generally needs to be maintained around 20ms, and the preset threshold may be set to 15 ms.
In the prior art, flow control is not added, the sum of data volume output by each level cannot be guaranteed, and the BufSize _ lat is generally close to a full state and is 100ms, which can be easily perceived by human ears and cannot be tolerated; by limiting the total data volume of the waiting queue in the output path, the data volume in the waiting queue in the output path can be controlled to be as small as possible, and the audio delay is effectively reduced.
When the total data volume of each stage of the output waiting queue in the output path is calculated, the total data volume can be calculated at the speed of 1ms once. And when the value is smaller than the preset threshold value, reading data from the buffer area, writing the data into the first-stage output, then sending the data obtained by the first-stage output to the second-stage output, and so on until the data is output to the loudspeaker.
In a specific embodiment provided by the present invention, retrieving the audio data to be processed from the buffer and outputting the audio data to an output path may include: and calling a specified number of audio data to be processed from the buffer area and outputting the audio data to an output path.
When the audio data to be processed is called from the buffer and output to the output path, a specified amount of audio data to be processed can be called each time, so that the total data amount in the output path is prevented from exceeding a preset threshold value.
For example, when the amount of data that can be stored in the buffer is 10ms, the total amount of data in the wait queue generally needs to be maintained around 20ms, and the preset threshold is 15ms, the amount of data written into the output channel at a time may be 2.5 ms.
In a specific embodiment provided by the present invention, storing the audio data to be processed in a buffer may include: storing the audio data to be processed into a buffer area; and if the buffer zone is saturated, discarding the overflowing audio data to be processed.
When the audio data to be processed acquired from the outside is stored in the buffer, if the buffer is saturated, the overflowed audio data to be processed can be discarded. The buffer management module managing the buffer can inform the input module of the input data that the data has been written, but discard the overflowed data. Although this produces very fine noise, this partial discarding occurs only once at the beginning because the data stream velocity is very uniform and stable, and is very fine, so its negative effects can be ignored.
In a specific embodiment provided by the present invention, before the step of retrieving the audio data to be processed from the buffer and outputting the audio data to an output path, the method may further include: judging whether the buffer area has data or not; if not, the audio data to be processed is called from the buffer area and output to an output path, and the method comprises the following steps: outputting the spoofed data to a first level output in an output path; if yes, the step of calling the audio data to be processed from the buffer area and outputting the audio data to an output channel is executed.
And when the buffer does not store data, the calculated total data amount in the output path is less than a preset threshold value, and the audio data to be processed needs to be called from the buffer, adopting the deceptive output to output the deceptive data to the first-stage output in the output path. For example, write 0 to the first stage output. This happens only once, since the data flow rate is very uniform and stable, and there is data every time data is read from the buffer later, so its negative effects are negligible.
The invention has the advantages that: the delay problem of audio transmission/playing can be solved only by modifying software without introducing new hardware; the hardware cost and the design cost of equipment are not increased, the products which are sold can be perfectly compatible, and the ultra-low delay is realized; the invention can be used for reducing delay on the intelligent equipment in a plurality of audio application scenes, so that the public entertainment activities are richer, and the user experience is better.
The above is a method for reducing audio delay provided by the present invention.
Based on the same inventive concept as the above-mentioned method for reducing audio delay, the embodiment of the present invention further provides an apparatus for reducing audio delay, as shown in fig. 3. Because the device embodiment is basically similar to the method embodiment, the description is simpler, and the relevant points can be referred to partial description of the method embodiment.
The invention provides a device for reducing audio delay, which comprises:
the data acquisition module 101 is configured to acquire input audio data to be processed;
a data buffer module 102, configured to store the audio data to be processed in a buffer area;
the data output module 103 is configured to retrieve the audio data to be processed from the buffer and output the audio data to an output path;
a data conversion module 104, configured to convert the audio data to be processed into an analog signal played by a speaker by using the output path;
and the audio playing module 105 is used for driving a loudspeaker to play the analog signal.
In a specific embodiment provided by the present invention, the apparatus further includes:
the data volume calculation module is used for calculating the total data volume of each stage of output waiting queues in the output channel in real time;
the data volume judging module is used for judging whether the total data volume is smaller than a preset threshold value;
if the number is less than the preset number, executing the content of the data output module 103;
if not, waiting for the output path to process the data of the waiting queue; continuing to execute the contents of the data volume calculation module.
In an embodiment of the present invention, the data output module 103 is specifically configured to:
and calling a specified number of audio data to be processed from the buffer area and outputting the audio data to an output path.
In an embodiment of the present invention, the data buffering module 102 is specifically configured to:
storing the audio data to be processed into a buffer area; and if the buffer zone is saturated, discarding the overflowing audio data to be processed.
In a specific embodiment provided by the present invention, the apparatus further includes:
the buffer area judging module is used for judging whether the buffer area has data or not;
if not, the data output module 103 is specifically configured to: outputting the spoofed data to a first level output in an output path;
if yes, the content of the data output module 103 is executed.
The above is an audio delay reduction apparatus provided by the present invention.
Based on the same inventive concept as the above-mentioned method for reducing audio delay, correspondingly, embodiments of the present invention further provide a computer-readable storage medium on which a computer program is stored, which, when executed by a processor, implements the above-mentioned method for reducing audio delay.
As can be seen from the foregoing technical solutions, in the computer-readable storage medium provided in this embodiment, a computer program is stored thereon, and when the program is executed by a processor, the computer program can obtain to-be-processed audio data, store the to-be-processed audio data in a buffer, call the to-be-processed audio data from the buffer, process the to-be-processed audio data, and buffer the data at a time of writing the data, so that the amount of data waiting for alignment in an output path can be reduced, and further, audio transmission or playback delay can be reduced; in addition, the invention does not need extra hardware cost and has stronger compatibility.
Based on the same inventive concept as the above-mentioned audio delay reduction method, correspondingly, an embodiment of the present invention further provides an audio delay reduction apparatus, including: memory, processor and computer program stored on the memory and executable on the processor, the processor implementing the above-described method of audio delay reduction when executing the program.
According to the technical scheme, the audio delay reduction device provided by the embodiment can reduce the data volume of the waiting alignment in the output path by acquiring the audio data to be processed, storing the audio data to be processed in the buffer, calling the audio data to be processed from the buffer, processing the audio data to be processed, and buffering the data at the time of writing the data, so that the audio transmission or playing delay can be reduced; in addition, the invention does not need extra hardware cost and has stronger compatibility.
In the description of the present invention, numerous specific details are set forth. It is understood, however, that embodiments of the invention may be practiced without these specific details. In some instances, well-known methods, structures and techniques have not been shown in detail in order not to obscure an understanding of this description.
In the description herein, references to the description of the term "one embodiment," "some embodiments," "an example," "a specific example," or "some examples," etc., mean that a particular feature, structure, material, or characteristic described in connection with the embodiment or example is included in at least one embodiment or example of the invention. In this specification, the schematic representations of the terms used above are not necessarily intended to refer to the same embodiment or example. Furthermore, the particular features, structures, materials, or characteristics described may be combined in any suitable manner in any one or more embodiments or examples. Furthermore, various embodiments or examples and features of different embodiments or examples described in this specification can be combined and combined by one skilled in the art without contradiction.
Finally, it should be noted that: the above embodiments are only used to illustrate the technical solution of the present invention, and not to limit the same; while the invention has been described in detail and with reference to the foregoing embodiments, it will be understood by those skilled in the art that: the technical solutions described in the foregoing embodiments may still be modified, or some or all of the technical features may be equivalently replaced; such modifications and substitutions do not depart from the spirit and scope of the present invention, and they should be construed as being included in the following claims and description.
Claims (6)
1. A method for audio delay reduction, comprising:
acquiring input audio data to be processed;
storing the audio data to be processed into a buffer area;
calling the audio data to be processed from the buffer area and outputting the audio data to an output channel;
converting the audio data to be processed into an analog signal played by a loudspeaker by using the output channel;
driving a loudspeaker to play the analog signal;
before the step of retrieving the audio data to be processed from the buffer and outputting the audio data to an output path, the method further comprises:
calculating the total data volume of each stage of output waiting queues in the output path in real time;
judging whether the total data volume is smaller than a preset threshold value or not;
if the current time is less than the preset time, calling the audio data to be processed from the buffer area and outputting the audio data to an output channel;
if not, waiting for the output path to process the data of the waiting queue; continuing to execute the step of calculating the total data volume of each stage of output waiting queues in the output path in real time;
the audio data to be processed is called from the buffer area and output to an output path, and the method comprises the following steps:
and calling a specified number of audio data to be processed from the buffer area and outputting the audio data to an output path.
2. The method of claim 1, wherein storing the audio data to be processed in a buffer comprises:
storing the audio data to be processed into a buffer area; and if the buffer zone is saturated, discarding the overflowing audio data to be processed.
3. The method of claim 1, further comprising, prior to the step of retrieving the audio data to be processed from the buffer for output to an output path:
judging whether the buffer area has data or not;
if not, the audio data to be processed is called from the buffer area and output to an output path, and the method comprises the following steps: outputting the spoofed data to a first level output in an output path;
if yes, the step of calling the audio data to be processed from the buffer area and outputting the audio data to an output channel is executed.
4. An apparatus for reducing audio delay, comprising:
the data acquisition module is used for acquiring input audio data to be processed;
the data buffer module is used for storing the audio data to be processed into a buffer area;
the data output module is used for calling the audio data to be processed from the buffer area and outputting the audio data to an output channel;
the data conversion module is used for converting the audio data to be processed into an analog signal played by a loudspeaker by using the output channel;
the audio playing module is used for driving a loudspeaker to play the analog signal;
the data volume calculation module is used for calculating the total data volume of each stage of output waiting queues in the output channel in real time;
the data volume judging module is used for judging whether the total data volume is smaller than a preset threshold value;
if the number of the data output modules is less than the preset number, executing the content of the data output module;
if not, waiting for the output path to process the data of the waiting queue; continuing to execute the contents of the data amount calculation module;
the data output module is specifically configured to:
and calling a specified number of audio data to be processed from the buffer area and outputting the audio data to an output path.
5. A computer-readable storage medium, on which a computer program is stored which, when being executed by a processor, carries out the method of one of claims 1 to 3.
6. An audio delay reduction apparatus comprising: memory, processor and computer program stored on the memory and executable on the processor, characterized in that the processor implements the method of one of claims 1 to 3 when executing the program.
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| CN110175081B (en) * | 2019-05-30 | 2021-05-11 | 睿云联(厦门)网络通讯技术有限公司 | Optimization system and method for Android audio playing |
| CN111526466B (en) * | 2020-04-30 | 2022-07-22 | 成都千立网络科技有限公司 | Real-time audio signal processing method for sound amplification system |
| CN111885415B (en) * | 2020-07-24 | 2023-04-07 | 海信视像科技股份有限公司 | Audio data rapid output method and display device |
| CN112071291A (en) * | 2020-08-28 | 2020-12-11 | 北京戴乐科技有限公司 | Low-delay audio processing device, system and method |
| CN112735463A (en) * | 2020-12-16 | 2021-04-30 | 杭州小伴熊科技有限公司 | Audio playing delay AI correction method and device |
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Effective date of registration: 20211009 Address after: 201601 No. 9, building 31, No. 89, dujiabang Road, Sijing Town, Songjiang District, Shanghai Patentee after: Shanghai Furnace Information Technology Co.,Ltd. Patentee after: Ni Xingwei Address before: 201600 No. 32, Lane 49, Henggang Road, Songjiang District, Shanghai 302 Patentee before: Ni Xingwei |