CN101753422A - Method for improving subjective quality of network audio - Google Patents
Method for improving subjective quality of network audio Download PDFInfo
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- CN101753422A CN101753422A CN200810218156A CN200810218156A CN101753422A CN 101753422 A CN101753422 A CN 101753422A CN 200810218156 A CN200810218156 A CN 200810218156A CN 200810218156 A CN200810218156 A CN 200810218156A CN 101753422 A CN101753422 A CN 101753422A
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Abstract
The invention provides a method for improving the subjective quality of network audio, which comprises the following steps: judging whether to need to enter a pre-buffer mode according to the number of audio data blocks in a buffer area; stopping playing and placing the received audio data into the buffer area until exiting from the pre-buffer mode if needing to enter the pre-buffer mode; judging whether to need to enter an active packet-losing mode according to the number of audio data blocks in a buffer area; and stopping playing and placing the received audio data into the buffer area until exiting from the active packet-losing mode if needing to enter the pre-buffer mode. The method for improving the subjective quality of network audio avoids the reduction of the transmission quality caused by frequent sound break during the network transmission and improves the subjective feeling of an audio receiver on the network audio by adopting pre-buffer and active packet losing modes.
Description
Technical field
The present invention relates to network communication field, relate in particular to a kind of method that improves subjective quality of network audio.
Background technology
At present based on the host-host protocol of ip network is main tcp agreement and udp agreement arranged, the tcp agreement is the reliable transmission that connection is arranged, and the udp agreement is connectionless unreliable transmission.Comparatively speaking, the udp agreement has simple, quick, real-time advantage, so, for the exigent network audio transmission of real-time, the general udp agreement that adopts is transmitted, concrete transmission mode is as follows: transmitting terminal is regularly taked voice data according to certain sample rate, carries out Network Transmission by the udp agreement of ip network, and receiving terminal is play the voice data that receives according to same sample rate.Yet; because the udp agreement itself has the low defective of reliability; loss of data, shake, order entanglement or the like take place through regular meeting in voice data in the process of Network Transmission; caused the network audio degradation; the recipient is very bad to the subjective feeling of the audio frequency that receives, even can't not hear (subjective quality that also can be understood as audio frequency is very low).
Under the low situation of existing network condition and udp protocol network transmission reliability, industry has adopted a lot of methods to improve the subjective quality of audio frequency, the audio frequency that the audio interface debit is heard is more smooth, existing reasonable technical scheme mainly is to have adopted the buffering area with preceding pooling feature to improve the quality of audio frequency, the voice data that receives is put in the buffering area, just begin when data arrive certain threshold value to play, this has improved the desultory problem of audio frequency to a certain extent.But this mode only is to have carried out pre-buffering when beginning to play, in case in playing process buffer empty, just can only wait pending data; And on the other hand, expired in case work as buffering area, will obliterated data.When obliterated data and pending data such as grade, staccato all can occur, and staccato often is accompanied by noise, therefore, the subjective quality of audio frequency still is unsatisfactory.
Summary of the invention
The object of the present invention is to provide a kind of method that improves subjective quality of network audio, be intended to solve the bad defective of recipient's subjective feeling that causes owing to the network audio transmission quality is relatively poor that exists in the prior art.
The present invention realizes like this, a kind of method that improves subjective quality of network audio, it may further comprise the steps a: the number according to buffering area sound intermediate frequency data block judges whether that needs enter initiatively packet loss pattern, enter initiatively packet loss pattern if desired, then stop the voice data that receives is put in the data buffer zone, until withdrawing from initiatively packet loss pattern.
Described method further may further comprise the steps b: the number according to buffering area sound intermediate frequency data block judges whether that needs enter pre-buffer mode, enter pre-buffer mode if desired, then stop to play and the audio data block that receives being put in the buffering area, until withdrawing from pre-buffer mode.
A pre-buffering threshold value is set in described step b, when the number of data block in the data buffer zone is zero, enters pre-buffer mode; If the number of data block is more than or equal to pre-buffering threshold value then withdraw from pre-buffer mode.
Described buffer pool size is 8 data blocks, and each data block size is 1024 bytes, and cushioning threshold value in advance is 2 data blocks.
An initiatively packet loss threshold value is set in described step a, when the number of data block in the data buffer zone equals buffer pool size, enters initiatively packet loss pattern; If the number of data block is smaller or equal to active packet loss threshold value then withdraw from initiatively packet loss pattern.
Described buffer pool size is 8 data blocks, and each data block size is 1024 bytes, and initiatively the packet loss threshold value is 3 data blocks.
After withdrawing from pre-buffer mode among the described step b, begin playing audio-fequency data immediately.
After withdrawing from active packet loss pattern among the described step a, the voice data that receives is deposited in the buffering area.
The present invention also provides a kind of method that improves subjective quality of network audio, and it may further comprise the steps:
Step 1 judges whether that according to the number of buffering area sound intermediate frequency data block needs enter pre-buffer mode, enters pre-buffer mode if desired, then stops to play and the audio data block that receives being put in the buffering area, until withdrawing from pre-buffer mode.
Step 2 judges whether that according to the number of buffering area sound intermediate frequency data block needs enter initiatively packet loss pattern, enters initiatively packet loss pattern if desired, then stops the voice data that receives is put in the data buffer zone, until withdrawing from initiatively packet loss pattern.
A pre-buffering threshold value is set in described step 1, an initiatively packet loss threshold value is set in described step 2, when the number of data block in the data buffer zone is zero, enter pre-buffer mode; If the number of data block is more than or equal to pre-buffering threshold value then withdraw from pre-buffer mode; When the number of data block in the data buffer zone equals buffer pool size, enter initiatively packet loss pattern; If the number of data block is smaller or equal to active packet loss threshold value then withdraw from initiatively packet loss pattern.
The method of raising subjective quality of network audio provided by the invention is by adopting initiatively packet loss and the mode of buffering in advance, and the transmission quality that frequent staccato is brought when having avoided the network audio transmission descends, and has improved the subjective feeling of recipient to network audio.
Description of drawings
Fig. 1 is the flow chart that the present invention improves the method preferred embodiment of subjective quality of network audio;
Fig. 2 is the audio transmission schematic diagram behind the employing pre-buffer mode of the present invention;
Fig. 3 is the audio transmission schematic diagram after the employing active packet loss pattern of the present invention.
Embodiment
In order to make purpose of the present invention, technical scheme and advantage clearer,, the present invention is further elaborated below in conjunction with embodiment.Should be appreciated that specific embodiment described herein only in order to explanation the present invention, and be not used in qualification the present invention.
Compared with prior art, the main feature of the method for raising subjective quality of network audio provided by the invention is to have increased pre-buffer mode and active packet loss pattern, makes that the audio frequency that the recipient heard in real time is smooth relatively understandable.
Described pre-buffer mode is meant when buffer empty, and a pre-buffer mark is set, and enters pre-buffer mode,, the data that receive is put into buffering area that is, wouldn't play.When the voice data of buffering area reaches certain piece and counts, withdraw from pre-buffer mode, just begin the voice data in the play buffer.
Described active packet loss pattern is meant when buffering area is full, an initiatively packet loss sign is set, enter initiatively packet loss pattern, promptly, in the process of playing, the data that receive are initiatively lost, even buffering area has enough spaces can deposit these data, also must initiatively abandon these data, when the empty piece of some appears in buffering area, just begin to receive again and store data, in other words, buffering area is under situation about constantly playing, have only when only being left several voice datas, just withdraw from initiatively packet loss pattern, begin the voice data that receives is put into buffering area.
Above-mentioned pre-buffer mode and active packet loss pattern relate to following main points:
1, select suitable buffer size, block size and piece number: buffering area can not be provided with too greatly, otherwise time-delay is also big, slow in reacting; Buffering area can not be provided with too for a short time, otherwise when network jitter, loss of data increases.The mode that general employing polylith is set of buffering area.Every is the byte number of 2 power.Present embodiment is arranged to 8 with buffering area, and every is set to 1024 bytes.
2, suitable pre-buffering threshold value is set: according to buffer blocks number and delay requirement, select suitable pre-buffering threshold value, too big, time-delay is just big, slow in reacting, and buffering area is easily full, causes the phenomenon of packet loss; Too little, buffering area is empty easily, causes interrupted phenomenon.It is 2 that present embodiment will cushion the threshold value value of establishing in advance, when the voice data in the buffering area when being empty, enters pre-buffer mode, when the voice data in the buffering area withdraws from pre-buffer mode during more than or equal to 2.
3, suitable active packet loss threshold value is set, according to the requirement of buffer blocks number and time-delay, selects suitable numerical value, too big, time-delay is just big, slow in reacting, and buffering area is easily full, causes the phenomenon of packet loss; Too little, buffering area is empty easily, causes interrupted phenomenon.Present embodiment initiatively packet loss threshold value is set to 3, enters initiatively packet loss pattern when buffering area is full, when the voice data in the buffering area during smaller or equal to 3 blocks of data, withdraws from initiatively packet loss pattern.
It is emphasized that pre-buffering threshold value and active packet loss threshold value can be identical, also can be different.
Following according to above-mentioned pre-buffer mode and packet loss pattern initiatively, specify the flow process that data block that the present invention improves the method preferred embodiment of subjective quality of network audio is handled in conjunction with Fig. 1:
After the network audio transmission beginning, in step 10, at first system is carried out initialization and with pre-buffer mark pre_buf zero setting, represent currently not cushion in advance, the current active packet loss that do not carry out is represented in active packet loss sign dismiss zero setting.In step 11, wait for and receive a voice data, receive after the voice data, check that in step 12 data block of buffering area is counted K, judge in step 13 whether K equals zero, if K equals zero, represent that then buffering area is empty, execution in step 131 stops to play, and pre-buffer mark pre_buf put 1, enter pre-buffer mode, execution in step 17 then, return step 11 after the data block that receives is put into buffering area.If judged result is for denying in the step 13, be that K is not equal to zero, then execution in step 14, judge whether pre-buffer mark pre_buf equals 1, promptly whether entered pre-buffer mode, if pre-buffer mark is 1, then expression has entered pre-buffer mode, execution in step 141, whether the data block of judging buffering area counts K more than or equal to pre-buffering threshold value, pre-buffering threshold value is 2 in the present embodiment, if K less than 2, then represents less than pre-buffering threshold value, execution in step 17, data are put into buffering area, if K more than or equal to 2, then represents to have surpassed pre-buffering threshold value, execution in step 142, begin displaying audio file and, withdraw from pre-buffer mode, continue execution in step 17 then pre-buffer mark pre_buf zero setting.
If the judged result of step 14 is for denying, be current not being in the pre-buffer mode, then continue execution in step 15, judge that initiatively whether packet loss sign dismiss is 1 to determine in the current packet loss pattern that whether has the initiative, if the judged result of step 15 is for denying (being in the current packet loss pattern that do not have the initiative), then execution in step 16, judge whether buffering area is full, be that data block is counted K and whether equaled buffer pool size, buffer pool size is 8 blocks of data in the present embodiment, if buffering area less than, then execution in step 7, and the data block that receives is put into buffering area, if buffering area is full, then in step 161, enter initiatively packet loss pattern, the data of receiving are lost and initiatively the packet loss sign put 1, return step 11 then.If the judged result of step 15 is for being, then execution in step 151, judge that whether the data block of buffering area count K smaller or equal to active packet loss threshold value, active packet loss threshold value is 3 in the present embodiment, if K is greater than 3, represent the current initiatively packet loss that needs, then return step 11, and the current voice data that receives is not put in the buffering area, has been lower than initiatively packet loss threshold value if K, represents the data block number in the buffering area smaller or equal to 3, then execution in step 152, withdraw from initiatively packet loss pattern and initiatively packet loss sign dismiss zero setting, enter step 17 then, return step 11 after data are put into buffering area.
By the foregoing description, the method that the present invention improves subjective quality of network audio can be summarized as following step:
Step 1 judges whether that according to the number of buffering area sound intermediate frequency data block needs enter pre-buffer mode, enters pre-buffer mode if desired, then stops to play and the audio data block that receives being put in the buffering area, until withdrawing from pre-buffer mode;
Step 2 judges whether that according to the number of buffering area sound intermediate frequency data block needs enter initiatively packet loss pattern, enters initiatively packet loss pattern if desired, then stops the voice data that receives is put in the data buffer zone, until withdrawing from initiatively packet loss pattern.
Certainly, step 1 and step 2 are not have inevitable sequencing.Step 1 can realize by pre-buffering threshold value is set, and is zero to be buffering area when being sky when the number of data block in the data buffer zone, enters pre-buffer mode; If the number of data block is greater than pre-buffering threshold value then withdraw from pre-buffer mode, the beginning playing audio-fequency data.Same, step 2 also can realize by the packet loss threshold value is set initiatively, is buffering area when full when the number of data block in the data buffer zone equals the buffer blocks number, then enters initiatively packet loss pattern; If the number of data block smaller or equal to active packet loss threshold value then withdraw from initiatively packet loss pattern, begins to receive data.
Below illustrate resulting effect after the method that adopts the present invention to improve subjective quality of network audio.In conjunction with referring to shown in Fig. 2, Fig. 2 has disclosed the improvement of adopting the network audio transmission quality that brings behind the pre-buffer mode of the present invention.Do not have pre-buffer mode of the present invention if cushion before having only, then when beginning, cushion 2 blocks of data, and begin to play at 128ms, 3 blocks of data are arranged, tentation data piece 3,4,5 lose, and finish buffer empty at 320ms, staccato 1 waits pending data, at 385ms multicast data piece 6, finish at 385+64=449ms, staccato 2 waits pending data, at 450ms multicast data piece 7, finish staccato 3 at 450+64=514ms, at 515ms multicast data piece 8, finish at 515+64=579ms, staccato 4 is at 580ms multicast data piece 9, finish at 580+64=644ms, staccato 5 at 650 multicast data pieces 10, finishes at 650+64=714ms, staccato 6, at 740ms multicast data piece 11, finish staccato 7 at 740+64=804ms, at 836ms multicast data piece 12, finish at 836+64=900ms, staccato 8 is at 930ms multicast data piece 13, finish at 930+64=994ms, staccato 9 at 1025ms multicast data piece 14, finishes at 1025+64=1089ms, staccato 10, at 1090ms multicast data piece 15, before 1090+64=1154ms finished, data block 16 entered, then broadcast piece 16, finish at 1090+2*64=1218, staccato 11 is at 1254 multicast data pieces 18.11 place's staccato and noises are during this period of time arranged, and sound effect is poor.
And under pre-buffer mode of the present invention, begin to play at 128ms, three blocks of data are arranged, finish same tentation data piece 3 at 320ms, 4,5 have lost, buffer empty, staccato 1 enters pre-buffering 1, data block 6,7,8 begin to play at 515ms, before 515+3*64=707ms finishes, data block 9,10 enter, and before 707+2*64=835ms finished, data block 11 entered, before 835+64=899ms finishes, data block 12 enters, and before 899+64=963ms finished, data block 13 entered, before 963+64=1027ms finishes, data block 14 enters, and before 1027+64=1091ms finished, data block 15 entered, before 1091+64=1155ms finishes, data block 16 enters, and finishes staccato 2 at 1155+64=1219ms, data block 17,18 enter pre-buffering 2, have only two place's staccatos during this period of time.
Obviously, after adopting pre-buffer mode of the present invention, when entering pre-buffering, staccato 1 is arranged, and the time is longer, the 3rd with time difference of first, the time of Network Transmission is uncertain, this time also is uncertain, but has avoided the staccato 2,3,4,5,6,7,8,9,10 of back to become 9 skies into once empty, and 9 staccatos are a staccato; Staccato 2 is equivalent to staccato 11 of the prior art, enters pre-buffering 2, and the network jitter of avoiding the back to occur has improved the subjective quality of the resulting audio frequency of audio interface debit effectively, has improved audio interface debit's the sense of hearing and has experienced.
Shown in Fig. 3, Fig. 3 has disclosed the improvement of adopting the audio transmission quality of being brought after the active packet loss pattern of the present invention: if there is not active packet loss pattern of the present invention, then each buffering area completely just begins packet loss, when data block 0,1,2 when entering, begin to play at 8ms, before first of 8+64=72ms finishes, data block 3,4,5,6,7 enter, when data block to 8 arrives, buffering area is full, and obliterated data piece 8 produces staccato 1, when data block 9 arrives, one of buffer empty can be put into, before second of 8+2*64=136ms finishes, during data block 10, buffering area is full, loses staccato 2, when data block 11 arrives, can enter, before the 3rd of 8+3*64=200ms finished, data block 12 was lost, staccato 3, data block 13 can enter, and before the 4th of 8+4*64=264ms finished, data block 14 was lost, staccato 4, data block 15 can enter, and before the 5th of 8+5*64=328ms finished, data block 16 was lost, staccato 5, data block 17 can enter, and before the 6th of 8+6*64=392ms finished, data block 18 was lost, staccato 6, data block 19 can enter, and before the 7th of 8+7*64=456ms finished, data block 20 was lost, staccato 7, data block 21 can enter, and before the 8th of 8+8*64=520ms finished, data block 22 was lost, staccato 8, data block 23 can enter.Staccato is arranged eight times during this period of time.
And if adopted active packet loss pattern of the present invention, when data block 8 arrives, enter initiatively packet loss pattern, before 8+5*64=328ms the 5th data block is play, data block 8,9,10,11,12,13,14,15,16 all lose, and staccato 1 is when data block 17 arrives, buffering area has only 3 blocks of data, withdraws from packet loss mechanism, data block 17,18,19,20,21 can enter, and data block 22 is when 510ms comes, data block 6,7 all finishes, and can enter, and data block 23 also can enter.The staccato 1 that has only occurred a long period 320ms during this period of time.
Obviously, said process is when buffering area is full, enter initiatively packet loss pattern, all data blocks that arrive during 320ms are after this all lost, withdraw from packet loss mechanism when buffering area has only 3 blocks of data behind the 320ms, make buffering area be in optimum state, can tolerate more packet loss and bigger network jitter, adopt and once to lose 9 blocks of data, avoid losing 8 staccatos that data are brought next 8 times.And often be accompanied by noise during staccato, and it is discontinuous in a minute that staccato can make, and is difficult to understand the other side's speech, and the present invention has reduced the staccato number of times significantly, has improved the audio frequency subjective quality effectively.
The above only is preferred embodiment of the present invention, not in order to restriction the present invention.All any modifications of being done within the spirit and principles in the present invention, be equal to and replace and improvement etc., all should be included within protection scope of the present invention.
Claims (10)
1. method that improves subjective quality of network audio is characterized in that may further comprise the steps a:
Number according to buffering area sound intermediate frequency data block judges whether that needs enter initiatively packet loss pattern, enters initiatively packet loss pattern if desired, then stops the voice data that receives is put in the data buffer zone, until withdrawing from initiatively packet loss pattern.
2. the method for raising subjective quality of network audio according to claim 1 is characterized in that described method further may further comprise the steps b:
Number according to buffering area sound intermediate frequency data block judges whether that needs enter pre-buffer mode, enters pre-buffer mode if desired, then stops to play and the audio data block that receives being put in the buffering area, until withdrawing from pre-buffer mode.
3. the method for raising subjective quality of network audio according to claim 2 is characterized in that: a pre-buffering threshold value is set in described step b, when the number of data block in the data buffer zone is zero, enters pre-buffer mode; If the number of data block is more than or equal to pre-buffering threshold value then withdraw from pre-buffer mode.
4. the method for raising subjective quality of network audio according to claim 3 is characterized in that: described buffer pool size is 8 data blocks, and each data block size is 1024 bytes, and cushioning threshold value in advance is 2 data blocks.
5. the method for raising subjective quality of network audio according to claim 1 is characterized in that: an initiatively packet loss threshold value is set in described step a, when the number of data block in the data buffer zone equals buffer pool size, enters initiatively packet loss pattern; If the number of data block is smaller or equal to active packet loss threshold value then withdraw from initiatively packet loss pattern.
6. the method for raising subjective quality of network audio according to claim 5 is characterized in that: described buffer pool size is 8 data blocks, and each data block size is 1024 bytes, and initiatively the packet loss threshold value is 3 data blocks.
7. the method for raising subjective quality of network audio according to claim 2 is characterized in that: after withdrawing from pre-buffer mode among the described step b, begin playing audio-fequency data immediately.
8. the method for raising subjective quality of network audio according to claim 1 is characterized in that: after withdrawing from active packet loss pattern among the described step a, the voice data that receives is deposited in the buffering area.
9. method that improves subjective quality of network audio is characterized in that may further comprise the steps:
Step 1 judges whether that according to the number of buffering area sound intermediate frequency data block needs enter pre-buffer mode, enters pre-buffer mode if desired, then stops to play and the audio data block that receives being put in the buffering area, until withdrawing from pre-buffer mode.
Step 2 judges whether that according to the number of buffering area sound intermediate frequency data block needs enter initiatively packet loss pattern, enters initiatively packet loss pattern if desired, then stops the voice data that receives is put in the data buffer zone, until withdrawing from initiatively packet loss pattern.
10. the method for raising subjective quality of network audio according to claim 9, it is characterized in that: a pre-buffering threshold value is set in described step 1, an initiatively packet loss threshold value is set in described step 2, when the number of data block in the data buffer zone is zero, enter pre-buffer mode; If the number of data block is more than or equal to pre-buffering threshold value then withdraw from pre-buffer mode; When the number of data block in the data buffer zone equals buffer pool size, enter initiatively packet loss pattern; If the number of data block is smaller or equal to active packet loss threshold value then withdraw from initiatively packet loss pattern.
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| CN200810218156A CN101753422A (en) | 2008-12-12 | 2008-12-12 | Method for improving subjective quality of network audio |
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Cited By (2)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN104113777A (en) * | 2014-08-01 | 2014-10-22 | 广州金山网络科技有限公司 | Audio stream decoding method and device |
| CN109378019A (en) * | 2018-10-31 | 2019-02-22 | 成都市极米科技有限公司 | Audio data read method and processing system |
-
2008
- 2008-12-12 CN CN200810218156A patent/CN101753422A/en active Pending
Cited By (4)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN104113777A (en) * | 2014-08-01 | 2014-10-22 | 广州金山网络科技有限公司 | Audio stream decoding method and device |
| WO2016015670A1 (en) * | 2014-08-01 | 2016-02-04 | 广州金山网络科技有限公司 | Audio stream decoding method and device |
| CN104113777B (en) * | 2014-08-01 | 2018-06-05 | 广州猎豹网络科技有限公司 | A kind of stream decoding method and device |
| CN109378019A (en) * | 2018-10-31 | 2019-02-22 | 成都市极米科技有限公司 | Audio data read method and processing system |
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Application publication date: 20100623 |