CN104506631A - Audio file cache method and audio file cache equipment - Google Patents
Audio file cache method and audio file cache equipment Download PDFInfo
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Abstract
An embodiment of the invention provides an audio file cache method. The method includes: calculating an actual cache speed of a current audio file cached at present in real time to response to a current audio file cache online play request triggered currently; comparing the actual cache speed of the current audio file with a theoretical cache speed of the current audio file; obtaining a fitting tone quality value according to a comparison result, and caching an audio file with the corresponding tone quality value from a server to local according to the fitting tone quality value. By adoption of the audio file cache method, cache tone quality of audio files can be adjusted in real time by an audio client according to network conditions so as to realize adjustment of cache speeds of the audio files, and accordingly influences on online play of the audio files due to changes of network conditions are remarkably reduced to realize better user experience. In addition, the embodiment of the invention further provides audio file cache equipment.
Description
Technical field
Embodiments of the present invention relate to Information Technology Agreement field, and more specifically, embodiments of the present invention relate to a kind of audio file caching method and equipment.
Background technology
This part embodiments of the present invention be intended to for stating in claims provide background or context.Description is not herein because be included in just admit it is prior art in this part.
At present, along with increasing user uses the Internet, the use of audio client is also more and more extensive.Audio client can comprise but be not limited only to audio class mobile phone terminal, web terminal, pc terminal etc., is the Terminal Service product based on audio content.
In prior art, during the displaying audio file online to user of the audio client based on this locality, generally, user can select the tonequality of audio file in advance, such as, standard tonequality or high definition tonequality etc., so, audio client will by the audio content of corresponding tonequality (the audio file size of such as standard tonequality is 3M, and this audio frequency size of high definition tonequality is then 6M) from server end buffer memory to this locality.
In prior art, once have selected tonequality by user, so in the process of the one or more audio file of follow-up play, the tonequality of audio file is all fixing, and after only having user again manually to adjust, tonequality just can change.
But inevitably existing network situation often there will be change, such as, when user selects tonequality time, network condition (includes but not limited to 2/3G, wifi network) very good, so user just selects the online displaying audio file of high definition tonequality, and in playing process, network condition is deteriorated, slowing of the audio file of user cache high definition tonequality will be caused, buffer memory speed may be caused cannot to ensure the audio frequency smooth playing of high definition tonequality, in this case audio frequency is play and be there will be the situation such as Ka Dun or the wait of long-time buffering, also user is made to experience poor for the online broadcasting of audio file.This is very bothersome process.
For this reason, be starved of a kind of method of audio buffer of improvement, to make user when trigger audio file is play online, audio client can regulate the tonequality of audio file buffer memory in real time according to the fine or not situation of current network, thus the phenomenon that the online broadcasting being significantly reduced in the situation subaudio frequency file of network condition instability is smooth not.
In the present context, embodiments of the present invention are expected to provide a kind of audio file caching method and equipment.
In the first aspect of embodiment of the present invention, provide a kind of audio file caching method, comprising: in response to the cache request of the current audio file of current triggering, calculate just in the real cache speed of the current audio file of buffer memory in real time; The size of the theoretical buffer memory speed of more described real cache speed and described current audio file; Result according to described comparison obtains adaptive tonequality value, and according to the tonequality value of this adaptation from the audio file of the corresponding tonequality value of server buffer to this locality.
In one embodiment of the invention, this result according to described comparison obtains adaptive tonequality value, comprise: when the ratio of described real cache speed and theoretical buffer memory speed is less than the first adjustment threshold value, the tonequality value of described adaptation is defined as the first tonequality value regulating threshold value corresponding with described first.
In yet another embodiment of the present invention, this result according to described comparison obtains adaptive tonequality value and comprises: when the ratio of described real cache speed and theoretical buffer memory speed is greater than the second adjustment threshold value, the tonequality value of described adaptation is defined as the second tonequality value regulating threshold value corresponding with described second; Wherein, described first threshold value is regulated to be less than described second adjustment threshold value.
In yet another embodiment of the present invention, the audio file that should continue the corresponding tonequality value of buffer memory according to the tonequality value of this adaptation from server comprises to this locality: continue from server buffer target audio file to this locality according to the tonequality value of this adaptation, wherein, described target audio file is: the part audio file of also non-buffer memory in described current audio file.
In yet another embodiment of the present invention, the audio file of the corresponding tonequality value of buffer memory should be continued to this locality from server according to the tonequality value of this adaptation, comprise: continue buffer memory subsequent sound frequency file to this locality according to the tonequality value of this adaptation from server, wherein, described subsequent sound frequency file is: in buffer memory order, be positioned at the next audio file after described current audio file.
In yet another embodiment of the present invention, this calculates just in the real cache speed of the current audio file of buffer memory in real time, comprising: the byte number obtaining the buffer memory of described current audio file under current time, and the broadcasting initial time of described current audio file; Calculate the time difference of described current time and described broadcasting initial time; The result that the byte number of described buffer memory and described time difference are divided by is defined as the real cache speed under described current time.
In the second aspect of embodiment of the present invention, provide a kind of audio buffer equipment, comprise: computing module, be configured for the online playing request of the current audio file in response to current triggering, calculate just in the real cache speed of the current audio file of buffer memory in real time; Comparison module, is configured for the size of the theoretical buffer memory speed of more described real cache speed and described current audio file; Acquisition module, the result be configured for according to described comparison obtains adaptive tonequality value; Cache module, is configured for tonequality value according to this adaptation from the audio file of the corresponding tonequality value of server buffer to this locality.
According to audio buffer method and the audio buffer equipment of embodiment of the present invention, can in the process of user cache audio file, calculate real cache speed in real time, and comparing by the theoretical buffer memory speed with this audio file, realize in the process of caching of the subsequent sound frequency file of this audio file or this audio file, automatically to regulate tonequality according to the result that compares, thus significantly reduce and play phenomenon smooth not online at the situation subaudio frequency file of network condition instability.Also therefore, in the process of the online displaying audio file of user, better experience can be play online for user brings.
Summary of the invention
summary of the invention
The present inventor finds, user is when the online displaying audio file of use audio client, and after choosing tonequality, audio client will play the audio file of buffer memory to user according to this kind of tonequality.And the fine or not situation of network condition is real-time change in practical application, likely after user have selected high definition tonequality, network condition there occurs deterioration, smooth not time this will cause audio client displaying audio file, also cause user to play online and experience poor consequence.
And if audio client can regulate the tonequality value of audio file in real time according to network condition, thus adapt with the buffer memory speed under current network, thus ensure the fluency of the online displaying audio file of user.
After describing general principle of the present invention, lower mask body introduces various non-limiting embodiment of the present invention.
application scenarios overview
First with reference to figure 1, user is by the online playing request of audio client 101 trigger audio file, audio client 101 sends the cache request of audio file so that by audio file from server buffer to this locality, to be play to user by the audio file of buffer memory to server 102.Simultaneously, server 102 also can pre-set the functional relation between real cache speed and theoretical buffer memory speed and be synchronized to audio client 101, for audio client 101 reference when carrying out the adjustment of tonequality value, it is local that this functional relation also directly can be stored in audio client 101.
illustrative methods
Below in conjunction with the application scenarios of Fig. 1, be described with reference to Figure 2 the method for audio buffer according to exemplary embodiment of the invention.It should be noted that above-mentioned application scenarios is only that embodiments of the present invention are unrestricted in this regard for the ease of understanding spirit of the present invention and principle and illustrating.On the contrary, embodiments of the present invention can be applied to applicable any scene.
Step 201: in response to the online playing request of the current audio file of current triggering, calculates in real time just in the real cache speed of the current audio file of buffer memory.
Present embodiment can be applied in audio client.The function that audio client provides audio file to play online to user, and each audio file is divided into due to the difference of online result of broadcast: minimum tonequality, standard tonequality and high definition tonequality.When user triggers the request of the online broadcasting of certain audio file, audio client namely can to server request at this audio file of local cache to play this audio file to user.Simultaneously, audio client is when this audio file of buffer memory, will be calculated this in real time just in the real cache speed of the audio file of buffer memory, the object calculating this real cache speed is in order to the follow-up object by itself and theoretical buffer memory speed compares, thus plays the buffer memory tonequality regulating audio file according to network condition in real time.
Be understandable that, for some audio files, the tonequality starting most to arrange between audio client and server according to one carries out buffer memory, and such as, acquiescence starts according to common tonequality download audio files most.
When audio client starts certain audio file of buffer memory, audio client is in order to calculate the real cache speed of this audio file of inspection, following parameter can be detected: the byte number M' of buffer memory download, the start time of displaying audio file is t1, the real-time time point of monitoring of displaying audio file is t2, can show that the real cache speed of the audio file of monitoring is in real time as follows thus: V'=M'/(t2-t1).
Concrete, in actual applications, step 201 can comprise in implementation process:
Steps A 1: the byte number obtaining the buffer memory of described current audio file under current time, and the broadcasting initial time of described current audio file.
Suppose that the current time monitoring audio file buffer memory is t2, first obtain the byte number M' at t2 moment current audio file buffer memory, and the broadcasting start time point t1 of this current audio file.
Steps A 2: the time difference calculating described current time and described broadcasting initial time.
Then the time difference of t2 and t1 is calculated.
Steps A 3: the result that the byte number of described buffer memory and described time difference are divided by is defined as the real cache speed under described current time.
Finally the byte number M' of buffer memory and time difference (t2-t1) are divided by, the division result obtained is defined as the real cache speed in the t2 moment.
Then step 202 is entered: the size of the theoretical buffer memory speed of more described real cache speed and described current audio file.
Then audio client can the size of the real cache speed that calculates of comparison step 201 and the theoretical buffer memory speed obtained from server end.
In the present embodiment, the theoretical buffer memory speed v of audio file is a reference value, can be calculated by server end.The account form of server end is as follows: the duration of v=current audio file byte number/audio file.
Step 203: the result according to described comparison obtains adaptive tonequality value.
Client adjusts song buffer memory in real time according to comparative result and downloads tonequality, thus coupling buffer memory speed.
In the present embodiment, first network condition can be set and well can ensure that the critical buffer memory speed of audio file smooth playing is 1.5v.In actual applications, when the real cache speed V' of general audio file is less than 1.5v, the broadcasting of audio file may occur that buffer time is long, play slack situation.Therefore setting when real cache speed V' is less than 1.5v is that network condition is not good, and the online broadcasting of current tonequality audio file cannot reach the requirement of smooth playing, needs to consider to reduce tonequality; And real cache speed V' to be network condition when being greater than 1.5v good, the online broadcasting of current tonequality audio file can reach the requirement of smooth playing.
The best buffer memory speed arranging the audio file of high definition tonequality is and then 4v.Generally, when the real cache speed V' of audio file is greater than 4v, the online result of broadcast of audio file reaches best.Wherein, coefficient " 4 " is: in the different tonequality files (i.e. different code check) of same audio frequency, adjacent code check is than the maximum of approximation.Code check typically refers to the average bit rate of digital music or video, simply can think and equal audio file size divided by reproduction time.Reproduction time for the different code checks of same audio file is identical, then code check is than the approximate ratio equaling audio file size.Such as audio frequency has the file of three kinds of tonequality, respectively corresponding 64/160/320kbps, and the ratio of adjacent code check is respectively 2.5/2, gets the maximum 4 of an approximation.
Be understandable that, after the functional relation between Servers installed real cache speed and theoretical buffer memory speed, this functional relation can be synchronized to audio client, for audio client reference when obtaining adaptive tonequality value.
So based on aforesaid functional relation, the tonequality value of described adaptation, when the ratio of real cache speed and theoretical buffer memory speed is less than the first adjustment threshold value, is defined as the first tonequality value regulating threshold value corresponding with first by audio client.Such as, if real cache speed V'<1.5v, namely, the ratio of real cache speed and theoretical buffer memory speed is less than 1.5 (first regulating threshold value can be 1.5 in the present embodiment), so, then the tonequality value of adaptation is adjusted to the first tonequality value, the first tonequality value is 64kbps in the present embodiment.So, when the tonequality value of audio client reduction audio file is to minimum tonequality and 64kbps, by reducing the size of theoretical buffer memory speed v, the real cache speed V' of audio file can be made to be greater than 1.5v, thus make user ensure the fluency play when online this audio file of broadcasting, as V' is still less than 1.5v, then cannot normal play under judging this network condition.
Meanwhile, based on aforesaid functional relation, the tonequality value of described adaptation when the ratio of real cache speed and theoretical buffer memory speed is greater than the second adjustment threshold value, can be defined as the second tonequality value regulating threshold value corresponding with described second by audio client; Wherein, described first threshold value is regulated to be less than described second adjustment threshold value.Such as, if real cache speed V'>4v, namely, the ratio of real cache speed and theoretical buffer memory speed is greater than 4 (second regulating threshold value can be 4 in the present embodiment), so, then the tonequality value of adaptation is adjusted to the second tonequality value, the second tonequality value can be 160kbps in the present embodiment.
It should be noted that, tonequality value is in the present embodiment citing signal, those skilled in the art can arrange different the first adjustment threshold value, the first tonequality value, the second adjustment threshold value and the second tonequality value according to the actual requirements, only need first to regulate threshold value to be less than the second adjustment threshold value, and the first tonequality value is less than the second tonequality value.
Then step 204 is entered: according to the tonequality value of this adaptation from the audio file of the corresponding tonequality value of server buffer to this locality.
So, after obtaining adaptive tonequality value in step 203, the likely still current audio file of the current online broadcasting of user, so the tonequality value according to this adaptation continues from server buffer target audio file to this locality by audio client, wherein, target audio file can be: the part audio file of also non-buffer memory in current audio file.
Certainly, also possibility audio client is from the next audio file of current audio file, just continue buffer memory according to the tonequality value of adaptation, so audio client just can continue buffer memory subsequent sound frequency file to this locality according to the tonequality value of this adaptation from server, wherein, subsequent sound frequency file can be: in buffer memory order, be positioned at the next audio file after described current audio file.
Visible, in the present embodiment, can in the process of user cache audio file, calculate the real cache speed of audio file in real time, and comparing by the theoretical buffer memory speed with this audio file, and then regulate tonequality value according to comparative result in the process of caching of the subsequent sound frequency file of this audio file or this audio file, thus the function by regulating tonequality value to realize the buffer memory speed of real-time matching audio file.So just significantly reduce and play phenomenon smooth not online at the situation subaudio frequency file of network condition instability.Also therefore, in the process of the online displaying audio file of user, better experience can be play online for user brings.
example devices
After the method describing exemplary embodiment of the invention, next, with reference to figure 3 pairs of exemplary embodiment of the invention, for audio file buffer memory device, in figure 3, this audio file buffer memory device can comprise:
Computing module 301, is configured for the online playing request of the current audio file in response to current triggering, calculates just in the real cache speed of the current audio file of buffer memory in real time.
Wherein, computing module 301 specifically can comprise:
Obtain submodule, be configured for the byte number obtaining the buffer memory of described current audio file under current time, and the broadcasting initial time of described current audio file; Calculating sub module, is configured for the time difference calculating described current time and described broadcasting initial time; With, determine submodule, be configured for the real cache speed result that the byte number of described buffer memory and described time difference are divided by be defined as under described current time.
Comparison module 302, is configured for the size of the theoretical buffer memory speed of more described real cache speed and described current audio file.
Acquisition module 303, the result be configured for according to described comparison obtains adaptive tonequality value.
Described acquisition module 303 concrete configuration is used for: when the ratio of described real cache speed and theoretical buffer memory speed is less than the first adjustment threshold value, the tonequality value of described adaptation is defined as the first tonequality value regulating threshold value corresponding with described first.
Wherein, acquisition module 303 can also be configured for: when the ratio of described real cache speed and theoretical buffer memory speed is greater than the second adjustment threshold value, the tonequality value of described adaptation is defined as the second tonequality value regulating threshold value corresponding with described second; Wherein, described first threshold value is regulated to be less than described second adjustment threshold value.
Cache module 304, is configured for tonequality value according to this adaptation from the audio file of the corresponding tonequality value of server buffer to this locality.
Wherein, cache module 304 concrete configuration is used for: continue from server buffer target audio file to this locality according to the tonequality value of this adaptation, wherein, described target audio file is: the part audio file of also non-buffer memory in described current audio file.
Wherein, cache module 304 concrete configuration is used for: continue buffer memory subsequent sound frequency file to this locality according to the tonequality value of this adaptation from server, wherein, described subsequent sound frequency file is: in buffer memory order, be positioned at the next audio file after described current audio file.
In the present embodiment, audio client can in the process of user cache audio file, calculate the real cache speed of audio file in real time, and comparing by the theoretical buffer memory speed with this audio file, and then regulate tonequality value according to comparative result in the process of caching of the subsequent sound frequency file of this audio file or this audio file, thus the function by regulating tonequality value to realize the buffer memory speed of real-time matching audio file.So just significantly reduce and play phenomenon smooth not online at the situation subaudio frequency file of network condition instability.Also therefore, in the process of the online displaying audio file of user, better experience can be play online for user brings.
Although it should be noted that the some devices or sub-device that are referred to audio file buffer memory device in above-detailed, this division is only not enforceable.In fact, according to the embodiment of the present invention, the Characteristic and function of two or more devices above-described can be specialized in one apparatus.Otherwise, the Characteristic and function of an above-described device can Further Division for be specialized by multiple device.
In addition, although describe the operation of the inventive method in the accompanying drawings with particular order, this is not that requirement or hint must perform these operations according to this particular order, or must perform the result that all shown operation could realize expectation.Additionally or alternatively, some step can be omitted, multiple step be merged into a step and perform, and/or a step is decomposed into multiple step and perform.
Although describe spirit of the present invention and principle with reference to some embodiments, but should be appreciated that, the present invention is not limited to disclosed embodiment, can not combine to be benefited to the feature that the division of each side does not mean that in these aspects yet, this division is only the convenience in order to state.The present invention is intended to contain the interior included various amendment of spirit and scope and the equivalent arrangements of claims.
Accompanying drawing explanation
By reference to accompanying drawing reading detailed description hereafter, above-mentioned and other objects of exemplary embodiment of the invention, feature and advantage will become easy to understand.In the accompanying drawings, show some execution modes of the present invention by way of example, and not by way of limitation, wherein:
Fig. 1 schematically shows the application scenarios schematic diagram according to embodiment of the present invention;
Fig. 2 schematically shows the method flow diagram according to embodiment of the present invention;
Fig. 3 schematically shows the equipment frame composition according to embodiment of the present invention;
In the accompanying drawings, identical or corresponding label represents identical or corresponding part.
Embodiment
Below with reference to some illustrative embodiments, principle of the present invention and spirit are described.Should be appreciated that providing these execution modes is only used to enable those skilled in the art understand better and then realize the present invention, and not limit the scope of the invention by any way.On the contrary, provide these execution modes to be to make the disclosure more thorough and complete, and the scope of the present disclosure intactly can be conveyed to those skilled in the art.
One skilled in the art will appreciate that embodiments of the present invention can be implemented as a kind of system, device, equipment, method or computer program.Therefore, the disclosure can be implemented as following form, that is: hardware, completely software (comprising firmware, resident software, microcode etc.) completely, or the form that hardware and software combines.
According to the embodiment of the present invention, a kind of audio buffer method and apparatus is proposed.
In this article, it is to be appreciated that involved term audio client comprises but is not limited only to audio class mobile phone terminal, web terminal, pc terminal etc., the Terminal Service product based on audio content.In addition, any number of elements in accompanying drawing is all unrestricted for example, and any name is all only for distinguishing, and does not have any limitation.
Below with reference to some representative embodiments of the present invention, explaination principle of the present invention and spirit in detail.
Claims (12)
1. an audio file caching method, is applied in client, comprises:
In response to the online playing request of the current audio file of current triggering, calculate just in the real cache speed of the current audio file of buffer memory in real time;
The size of the theoretical buffer memory speed of more described real cache speed and described current audio file;
Result according to described comparison obtains adaptive tonequality value, and according to the tonequality value of this adaptation from the audio file of the corresponding tonequality value of server buffer to this locality.
2. method according to claim 1, the described result according to described comparison obtains adaptive tonequality value, comprising:
When the ratio of described real cache speed and theoretical buffer memory speed is less than the first adjustment threshold value, the tonequality value of described adaptation is defined as the first tonequality value regulating threshold value corresponding with described first.
3. method according to claim 1, the described result according to described comparison obtains adaptive tonequality value, comprising:
When the ratio of described real cache speed and theoretical buffer memory speed is greater than the second adjustment threshold value, the tonequality value of described adaptation is defined as the second tonequality value regulating threshold value corresponding with described second; Wherein, described first threshold value is regulated to be less than described second adjustment threshold value.
4. method according to claim 1, the described tonequality value according to this adaptation continues the audio file of the corresponding tonequality value of buffer memory to this locality from server, comprising:
Continue from server buffer target audio file to this locality according to the tonequality value of this adaptation, wherein, described target audio file is: the part audio file of also non-buffer memory in described current audio file.
5. method according to claim 1, the described tonequality value according to this adaptation continues the audio file of the corresponding tonequality value of buffer memory to this locality from server, comprising:
Continue buffer memory subsequent sound frequency file to this locality according to the tonequality value of this adaptation from server, wherein, described subsequent sound frequency file is: in buffer memory order, be positioned at the next audio file after described current audio file.
6. method according to claim 1, described real-time calculating just in the real cache speed of the current audio file of buffer memory, comprising:
Obtain the byte number of the buffer memory of described current audio file under current time, and the broadcasting initial time of described current audio file;
Calculate the time difference of described current time and described broadcasting initial time;
The result that the byte number of described buffer memory and described time difference are divided by is defined as the real cache speed under described current time.
7. an audio buffer equipment, comprising:
Computing module, is configured for the online playing request of the current audio file in response to current triggering, calculates just in the real cache speed of the current audio file of buffer memory in real time;
Comparison module, is configured for the size of the theoretical buffer memory speed of more described real cache speed and described current audio file;
Acquisition module, the result be configured for according to described comparison obtains adaptive tonequality value;
Cache module, is configured for tonequality value according to this adaptation from the audio file of the corresponding tonequality value of server buffer to this locality.
8. equipment according to claim 7, described acquisition module concrete configuration is used for: when the ratio of described real cache speed and theoretical buffer memory speed is less than the first adjustment threshold value, the tonequality value of described adaptation is defined as the first tonequality value regulating threshold value corresponding with described first.
9. equipment according to claim 8, described acquisition module is also configured for:
When the ratio of described real cache speed and theoretical buffer memory speed is greater than the second adjustment threshold value, the tonequality value of described adaptation is defined as the second tonequality value regulating threshold value corresponding with described second; Wherein, described first threshold value is regulated to be less than described second adjustment threshold value.
10. equipment according to claim 7, described cache module concrete configuration is used for:
Continue from server buffer target audio file to this locality according to the tonequality value of this adaptation, wherein, described target audio file is: the part audio file of also non-buffer memory in described current audio file.
11. equipment according to claim 7, described cache module concrete configuration is used for:
Continue buffer memory subsequent sound frequency file to this locality according to the tonequality value of this adaptation from server, wherein, described subsequent sound frequency file is: in buffer memory order, be positioned at the next audio file after described current audio file.
12. equipment according to claim 7, described computing module comprises:
Obtain submodule, be configured for the byte number obtaining the buffer memory of described current audio file under current time, and the broadcasting initial time of described current audio file;
Calculating sub module, is configured for the time difference calculating described current time and described broadcasting initial time;
Determine submodule, be configured for the real cache speed result that the byte number of described buffer memory and described time difference are divided by be defined as under described current time.
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| CN111552454B (en) * | 2020-04-29 | 2023-10-27 | 广州酷狗计算机科技有限公司 | Audio playing method, device, terminal and storage medium |
| CN114173426A (en) * | 2021-11-30 | 2022-03-11 | 广州番禺巨大汽车音响设备有限公司 | Wireless sound box playing control method, device and system based on wireless audio transmission |
| CN114173426B (en) * | 2021-11-30 | 2023-09-29 | 广州番禺巨大汽车音响设备有限公司 | Wireless sound box playing control method, device and system based on wireless audio transmission |
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Effective date of registration: 20170103 Address after: Hangzhou City, Zhejiang province Zhejiang District 310052 e-businessmen Road No. 599 building 601 room 4 Applicant after: Hangzhou NetEase cloud Music Technology Co., Ltd. Address before: Hangzhou City, Zhejiang province Binjiang District 310052 River Street Network Road No. 599 building 4 layer 7 Applicant before: NetEase (Hangzhou) Network Co., Ltd. |
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