CN1464685A - Method for processing acoustic frequency flow playback in network terminal buffer - Google Patents
Method for processing acoustic frequency flow playback in network terminal buffer Download PDFInfo
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Abstract
The invention discloses a method for processing audio stream playback in the network terminal buffer zone for solving the problem of voice pausing and jamming in the network communication. The aim of the invention is achieved by performing real time audio (e.g. voice) communication on the packet-switching network (e.g. IP network), arranging a shake buffer zone on the receiving end, after the receiving end receives the audio package, it first performs decoding based on the normal sequence, then places it into the shake buffer zone, when the shake buffer zone is to be filled, lower the sampling rate to the audio data to realize the fast playback of the audio data stream, when the shake buffer zone is to be empty, raise the sampling rate to the audio data to realize the low speed playback of the audio data stream, when the audio data in the shake buffer is within the normal range, playback the audio stream with the original sampling rate.
Description
Technical field
The present invention relates to the method for a kind of processing audio stream, refer in particular to and make conversation processing audio in network terminal buffer continuous, smooth normal play flow back to the method for putting on the packet network.
Background technology
Current, because the fast development of packet network, originally mainly be carried on the speech real time communication on the circuit-switched network, beginning is transferred on the IP packet network with the form of packet-switched speech (VoIP) on a large scale, finally can form the network that audio frequency and data and other Streaming Media merge.There are some different qualities in traditional circuit-switched network and packet network, the communication line that fixed-bandwidth end to end is provided and monopolizes such as circuit-switched network, not problem such as the data-bag lost that can occur existing on the packet network, out of order, delay variation; Because the high efficiency that its transmission bandwidth utilizes, the demand of the network integration, and the flexibility of networking and network management, dilatation etc., determined the necessity of transmitting audio signal on packet network, but problems such as the data-bag lost that exists on the packet network, out of order, delay variation, can have a strong impact on the quality of real-time audio communication, cause non-continuous events such as card sound, staccato.
The influence of losing the voice communication quality of packet is mainly reflected on the packet loss, if packet loss is low such as 1% to 2%, then the hearer can not feel that tangible audio quality descends, but rising along with packet loss, the hearer can feel the intermittent or discontinuous of audio signal, this moment, the hearer may can also understand speaking of the other side when feeling a small amount of intermittent, but a large amount of intermittently will make the hearer not understand the other side what is being said, thereby caused communication to stop.A spot of data-bag lost can promptly remedy the packets of audio data of losing by interpolation by utilizing the redundancy of audio signal, but the lot of data packet loss, in any case the quality of voice communication is descended.
Out of order and delay variation all can cause in essence to be needed continuously, the shake on reproduction time of the packets of audio data of played in order; Because real-time audio communication is a continuous process, each packets of audio data will be pressed the Fixed Time Interval played in order after decoding, so voice data wraps in the temporal shake of arrival and can cause acoustically discontinuous; If shake too greatly, the packet that then has is because the value that not have broadcast too late of coming and being lost.
One of method that solves data package jitter is at receiving terminal a jitter-buffer to be set, when arriving, at first puts into packet this buffering area, when the plays back audio data,, take out packets of audio data in time equably and be sent to the audio playback device broadcast in line with the principle of first in first out; As long as this jitter-buffer is not read sky by audio playback programs, then can guarantee the continuous playback of audio stream.This jitter-buffer is big more, and then shake that can be level and smooth is big more, but excessive jitter-buffer can cause the increasing of voice reproducing time delay, and excessive time delay also is worthless, can cause the difficulty of real-time communication; The big I of jitter-buffer is set according to real needs, also can dynamically adjust according to network condition.But can not effectively guarantee the continuous broadcast of audio stream like this.
Summary of the invention
The purpose of this invention is to provide a kind of speech quality good, and can keep real-time communication a kind of in network terminal buffer processing audio flow back to the method for putting.
The present invention is achieved in that when transmission on the packet network comes packets of audio data to enter jitter-buffer, calculates the distance D between the reading and writing pointer;
In initialization procedure, the scope of the distance D normal value between the reading and writing pointer and the data length of once reading are set from jitter-buffer;
In running, the distance D between the reading and writing pointer just often, audio stream is by the normal speed playback; When the distance D between the reading and writing pointer is greater than or less than normal value, to the data that are greater than or less than normal play length of from jitter-buffer, the reading processing that resamples, make it recover normal playout length by the audio resampling unit.
Above-mentioned jitter-buffer is annular jitter-buffer.
Distance D between the described reading and writing pointer is poor for read pointer side-play amount that enters jitter-buffer and write pointer side-play amount.
When D less than range of normal value following in limited time, the data length of reading from buffer circle adopts to rise sampling processing less than normal play length.
When D greater than the going up in limited time of range of normal value, the data length of reading from buffer circle adopts down-sampled processing greater than normal play length.
When the side-play amount of read pointer during less than the side-play amount of write pointer, the distance D between the reading and writing pointer is side-play amount gained poor of the offset minus read pointer of write pointer.
When the side-play amount of read pointer during greater than the side-play amount of write pointer, the distance D between the reading and writing pointer is that the difference of side-play amount gained of offset minus read pointer of write pointer of annular jitter-buffer is in the value of the length gained that adds annular jitter-buffer.
This method both can realize also can realizing by software coding on implementation by hardware encoding.
After adopting said method, when the distance D between the reading and writing pointer is outlier, by processing that the speech data block that is greater than or less than normal length of reading from jitter-buffer is resampled, thereby quicken or the broadcasting speed that slows down, finally make read pointer, write pointer in the annular jitter-buffer remain certain distance, guaranteed the continuous smooth playing of the real-time audio that transmission comes on packet network stream, make conversation continuous, smooth, reduced non-continuous events such as card sound, staccato.
Description of drawings
The invention will be further described below in conjunction with accompanying drawing and concrete execution mode.
Fig. 1 is a packet network voice communication schematic diagram;
Fig. 2 is the transmission part schematic diagram of audio communication terminal;
Fig. 3 is the receiving unit schematic diagram of audio communication terminal;
Fig. 4 is annular jitter-buffer schematic diagram.
Embodiment
As shown in Figure 1, each terminal is connected to by certain access way among the packet network, and each voice frequency terminal can send or receive packets of audio data to another voice frequency terminal by this packet network; A plurality of voice frequency terminals also can by certain form form one can MPTY conference network.
As shown in Figure 2, audio input signal is at first delivered to the audio collection unit, the conversion of audio signal from the analog to digital signal finished in this audio collection unit, quantizing process just, this processing procedure generally is quantified as audio signal the signed number word signal of 16Bit precision, send into audio coder afterwards and carry out data compression to save the network bandwidth, after the encoded compression of audio signal, send into the packing delivery unit, the audio pack that the packing data unit generally will be used for real-time communication encapsulates by real-time communication protocol (RTP) standard, be encapsulated as User Data Protocol (UDP) bag afterwards again, squeeze into interconnection agreement (IP) bag at last and be sent on the network.Network interface unit generally is physical layer and the array link layer in the network layer, as Ethernet interface chip or modulator-demodulator etc., is sent on the packet network through network interface unit at last through the packets of audio data after the overcompression.
Before packets of audio data arrives the target receiving terminal, be by a series of Network Transmission unit; This may comprise multiple switching equipment and routing device, different routes and network condition can produce different propagation delay times, thereby cause the packets of audio data that transmits by the constant duration order can anisotropically arrive receiving terminal in time, the shake that this just causes packet to receive; The packets of audio data that transmits by the constant duration order might transmit on different routes in addition, such as the packet p1 that order sends, and p2, p3 ... the order at receiving terminal may become p1, p3, and p2 ....
As shown in Figure 3; Packets of audio data is through arriving the network interface unit of receiving terminal after the packet network transmission, network interface unit is reduced into interconnection agreement (IP) packet after the packet of receiving being removed information such as physical layer address; This packet then is admitted to audio stream and receives and the unit of unpacking, and removes interconnection agreement (IP) header packet information, User Data Protocol (UDP) header packet information, real-time communication protocol (RTP) header packet information in this processing unit, is reduced to packets of audio data at last; Packets of audio data after the reduction is admitted to audio pack and separates out of order unit, solves by tactic packet normal time; By normal time tactic packet then be admitted to audio decoding unit, solve the linear code of audio signal; The linear code of the audio signal that solves is write annular jitter-buffer continuously and is temporarily stored; Because the shake of packets of audio data through occurring after the Network Transmission, the write operation that makes voice data flow to annular jitter-buffer is heterogeneous in time, but it is continuously and uniformly that the playback of audio signal requires in time, so the operation of reading voice data and playback from annular jitter-buffer is asynchronous with toward the write operation of annular jitter-buffer the time.
Processing and the replayed section of following detailed description voice data in buffer circle comprises the wobble buffer among Fig. 3, wobble buffer sensing element, wobble buffer read-out control unit, audio resampling unit, voice reproducing unit; Solid line is a data flow among Fig. 3, dotted line position control flows.
As shown in Figure 4, jitter-buffer is actually the continuous memory space that a length is N, the initial address of this memory space is represented with side-play amount 0, the end address is represented with side-play amount N-1, the current address pointer that writes is represented with side-play amount W, the current address pointer of reading represents that with side-play amount R the current distance that writes pointer and read between the pointer is represented with D; Write operation to this jitter-buffer writes voice data for the memory location to current write pointers point, the write pointer side-play amount adds 1 and to the length N complementation of jitter-buffer afterwards, promptly can be expressed as with the C language and whenever write W=(W+1) %N after the data; Read operation to this jitter-buffer adds 1 for read pointer side-play amount after whenever reading a data unit, can be expressed as with the C language equally and whenever read R=(R+1) %N after the data; Such read-write operation can guarantee when the read-write pointer arrives the top of jitter-buffer, and just during side-play amount N-1, read or write meeting next time is turned to the bottom of jitter-buffer automatically, and just side-play amount 0; In fact such operation is equivalent to the top and bottom of this jitter-buffer is joined, and has formed a buffer circle, and the read-write pointer is according to identical direction (clockwise or contrary pointer) rotation; Do not clash in order to guarantee to read and write between the pointer, promptly read pointer will keep following all the time the rear at write pointer, and this is to judge by the distance D between the read-write pointer; If W is greater than R then D=W-R; If W is less than R then D=N+W-R; Thus as long as D can guarantee that greater than 0 read pointer follows the rear at write pointer all the time, the distance D that reads while write between the pointer is sent into the jitter-buffer read-out control unit as control signal, in fact the distance D of reading and writing between the pointer is equivalent to an error controling signal in the closed-loop control system, as the foundation to the jitter-buffer read operation.
The jitter-buffer read-out control unit obtains reading and writing after the distance D between the pointer, the size of the audio data block of reading when deciding each audio playback device needs playback of data according to the size of D; The voice reproducing unit generally is to need the audio data block of certain-length to carry out playback by constant duration (such as 30 milliseconds), if the sample rate of audio signal is that to need length be the audio data block of 240 sampled points for 8000 sampled point per seconds (8k/s) then per 30 milliseconds; If the size of D is in normal scope, read a blocks of data (such as 240 voice datas) of normal length when then jitter-buffer read-out control unit control jitter-buffer sensing element needs data in the voice reproducing unit at every turn, and this moment, the audio resampling unit did not carry out any operation to these data, pellucidly this blocks of data was delivered to the voice reproducing unit and was play.
If the distance D overrun between the read-write pointer, as the excessive or too small adjustment that then needs to carry out necessity, because if D is too small, the randomized jitter that the time because packet arrives then may take place causes this jitter-buffer to be read empty situation appearance frequently, thereby cause W to occur less than the situation of R, the fast situation that read pointer runs in buffer circle than write pointer just occurs, staccato and card sound at this moment will occur; If D is excessive, since above-mentioned identical, the situation appearance that W walks around R one circle may appear, and at this moment staccato and card sound also can appear.
In addition, always occur the situation of data-bag lost on the Network Transmission, the meaning D of statistical average can be therefore As time goes on more and more littler during from length; Also have a factor can influence the variation of D, if the sampling clock frequency of audio frequency transmitting terminal is different from the broadcast clock frequency of audio interface receiving end, following situation will appear, promptly when the sampling clock frequency of audio frequency transmitting terminal during greater than the broadcast clock frequency of audio interface receiving end, D can be therefore As time goes on increasing, when the sampling clock frequency of audio frequency transmitting terminal during less than the broadcast clock frequency of audio interface receiving end, D can be therefore As time goes on more and more littler.
Suppose that above-mentioned combined cause allows D to change in greater than the scope of d1 less than d2, that is to say when d1<D<d2 that the jitter-buffer sensing element is each under the control of control corresponding unit reads the audio data block that length is L from jitter-buffer; When D<d1, represent that this jitter-buffer will be read sky probably, when control unit should be controlled the jitter-buffer sensing element and needed the next frame data in the voice reproducing unit this moment, read the frame data of length, suppose that its length is L1 (L1<L) less than normal length L; This moment, the audio resampling unit just was that the audio data block of L1 becomes the audio data block that length is L by the interpolation arithmetic that meets the audio frequency apperceive characteristic with length, and there is the audio data playback of full-length assurance voice reproducing unit; Be equivalent to like this slow down playback speed of this frame voice data, if the difference of L and L1 be not too greatly (such as ((L-L1)/L)<1%) then subjectivity acoustically do not have appreciable variation; The playback speed of voice data owing to slowed down, can predict D therefore can become big more, (d1<D<d2) just can carry out the playback of audio frequency by normal speed in the time of in D gets back to normal scope more.
When D>d2, represent that this jitter-buffer will be write completely probably, when control unit should be controlled the jitter-buffer sensing element and needed the next frame data in the voice reproducing unit this moment, read the frame data of length, suppose that its length is L2 (L2>L) greater than normal length L; This moment, the audio resampling unit just was that the audio data block of L2 becomes the audio data block that length is L by meeting the extract operation of audio frequency apperceive characteristic with length, and there is the audio data playback of full-length assurance voice reproducing unit; Be equivalent to quicken the playback speed of this frame voice data like this, equally if the difference of L and L2 be not too greatly (such as ((L2-L)/L)<1%) then subjectivity acoustically do not have appreciable variation; Owing to accelerated the playback speed of voice data, can predict D therefore can become more little, (d1<D<d2) just can carry out the playback of audio frequency by normal speed in the time of in D gets back to normal scope more.
If the jitter range of packet arrival time is too big or packet loss is too big, then might cause buffer circle to be read sky or write full, can do abnormality processing this moment, when this buffering area is read sky, the previous frame speech data of can resetting, if buffering area still be a sky during back to back read operation next time, then play mute signal; If it is full that buffer circle is write, then wash out all played data not in this buffering area automatically, and restart normally normal read write operation this buffering area.
This method both can realize also can realizing by software coding on implementation by hardware encoding.
Claims (8)
1, a kind of in network terminal buffer processing audio flow back to the method for putting, it is characterized in that: when transmission on the packet network comes packets of audio data to enter jitter-buffer, calculate the distance D between the reading and writing pointer;
In initialization procedure, the scope of the distance D normal value between the reading and writing pointer and the data length of once reading are set from jitter-buffer;
In running, the distance D between the reading and writing pointer just often, audio stream is by the normal speed playback; When the distance D between the reading and writing pointer is greater than or less than normal value, to the data that are greater than or less than normal play length of from jitter-buffer, the reading processing that resamples, make it recover normal playout length by the audio resampling unit.
2, according to claim 1 a kind of in network terminal buffer processing audio flow back to the method for putting, it is characterized in that: described jitter-buffer is annular jitter-buffer.
3, according to claim 1 and 2 a kind of in network terminal buffer processing audio flow back to the method for putting, it is characterized in that: the distance D between the described reading and writing pointer is poor for read pointer side-play amount that enters jitter-buffer and write pointer side-play amount.
4, according to claim 1 and 2 a kind of in network terminal buffer processing audio flow back to the method for putting, it is characterized in that: when D prescribes a time limit less than the following of range of normal value, the data length of reading from buffer circle adopts to rise sampling processing less than normal play length.
5, according to claim 1 and 2 a kind of in network terminal buffer processing audio flow back to the method for putting, it is characterized in that: when D going up in limited time greater than range of normal value, the data length of reading from buffer circle adopts down-sampled processing greater than normal play length.
6, according to claim 3 a kind of in network terminal buffer processing audio flow back to the method for putting, it is characterized in that: when the side-play amount of read pointer during less than the side-play amount of write pointer, the distance D between the reading and writing pointer is side-play amount gained poor of the offset minus read pointer of write pointer.
7, according to claim 3 a kind of in network terminal buffer processing audio flow back to the method for putting, it is characterized in that: when the side-play amount of read pointer during greater than the side-play amount of write pointer, the distance D between the reading and writing pointer is that the difference of side-play amount gained of offset minus read pointer of write pointer of annular jitter-buffer is in the value of the length gained that adds annular jitter-buffer.
8, according to claim 1 and 2 a kind of in network terminal buffer processing audio flow back to the method for putting, it is characterized in that: this method both can realize also can realizing by software coding on implementation by hardware encoding.
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