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Method for processing acoustic frequency flow playback in network terminal buffer

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Publication number
CN1464685A
CN1464685A CN 02115372 CN02115372A CN1464685A CN 1464685 A CN1464685 A CN 1464685A CN 02115372 CN02115372 CN 02115372 CN 02115372 A CN02115372 A CN 02115372A CN 1464685 A CN1464685 A CN 1464685A
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audio
buffer
playback
shake
data
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CN 02115372
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Chinese (zh)
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余泊
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优创科技(深圳)有限公司
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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 ofthe 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

一种在网络终端缓冲区中处理音频流回放的方法 A method of processing an audio stream playback terminal in a network buffer

技术领域 FIELD

本发明涉及一种处理音频流的方法,尤其是指在包交换网络上使通话连续、流畅正常播放的在网络终端缓冲区中处理音频流回放的方法。 The present invention relates to a method for processing an audio stream, and particularly to a method of processing an audio stream playback terminal in a network buffer so that the call on a packet switched network of continuous, smooth normal playback.

背景技术 Background technique

当前,由于包交换网络的快速发展,原来主要承载在电路交换网络上的话音实时通信,开始大规模以包交换语音(VoIP)的形式转移到IP包交换网络上,最终会形成音频与数据及其它流媒体融合的网络。 Current real time voice communication, since the rapid development of packet-switched networks, originally carried on the main circuit switching network, in the form of mass transfer start packet switched voice (VoIP) to the IP packet-switched network, audio and data will eventually be formed and other network streaming media convergence. 传统的电路交换网络与包交换网络存在一些不同特性,比如电路交换网络提供端到端的固定带宽并且独占的通信线路,不会出现包交换网络上存在的数据包丢失、乱序、时延抖动等问题;由于其传输带宽利用的高效率,网络融合的需求,以及组网与网络管理、扩容等的灵活性,决定了在包交换网络上传输音频信号的必要性,但是包交换网络上存在的数据包丢失、乱序、时延抖动等问题,会严重影响实时音频通信的质量,造成卡音、断音等不连续现象。 Traditional circuit switched networks and packet switched networks have some different characteristics, such as a circuit-switched network and provide end to end fixed bandwidth exclusive communication line, will not be present on the packet switched network packet loss, disorder, delay jitter, etc. problem; transmission bandwidth utilization due to its high efficiency, the demand for network convergence, and networking and network management, expansion and so flexibility, determines the necessity of transmission of audio signals in a packet switched network, the packet exists on the network but switched packet loss, out of order, delay jitter and other issues, will seriously affect the quality of real-time audio communication, resulting in sound card, staccato, etc. discontinuity.

数据包的丢失对音频通信质量的影响主要体现在丢包率上,如果丢包率较低比如1%到2%,则听者不会感觉到明显的音频质量下降,但是随着丢包率的上升,听者会感觉到音频信号的断断续续或不连续,此时听者在感觉到少量断断续续时可能还能听懂对方的说话,但是大量的断续就会使听者听不懂对方在说什么了,从而造成通信终止。 Effect of lost packets on the audio quality communication mainly in the packet loss rate, packet loss rate is low, such as if the 1% to 2%, the listener does not feel significantly decreased audio quality, but with the loss ratio the rise, the listener will feel the audio signal is intermittent or discontinuous, this time in the listener feel a small amount of intermittent may be able to understand each other speak, but it will make a lot of intermittent listener did not understand each other what to say, resulting in termination of communication. 少量的数据包丢失可以通过利用音频信号的冗余性,即通过插值对丢失的音频数据包进行弥补,但是大量的数据包丢失,无论如何都会使音频通信的质量下降。 A small amount of packet loss by using the redundancy of the audio signal, i.e., to compensate for lost audio packets through interpolation, but a large amount of packet loss, in any case the quality of the audio communication is reduced.

乱序与时延抖动本质上都会造成需要连续、顺序播放的音频数据包的在播放时间上的抖动;由于实时音频通信是一个连续的过程,每个音频数据包要在解码之后按固定时间间隔顺序播放,因此音频数据包在到来时间上的抖动会造成听觉上的不连续;如果抖动太大,则有的数据包由于来的太晚已没有播放的价值而被丢掉。 Scrambled and are essentially caused by the delay jitter required continuous dither audio packet sequence played on a playback time; as real-time audio communication is a continuous process, for each audio packet by a fixed time interval after decoding the order of play, so the audio packet jitter in the arrival time of the hearing caused by discontinuities; if jitter is too large, then some packets due to the value of late has not been playing to lose.

解决数据包抖动的方法之一是在接收端设置一个抖动缓冲区,在数据包到来时首先放入该缓冲区,在回放音频数据时,本着先进先出的原则,在时间上均匀地取出音频数据包并送往音频回放设备播出;只要该抖动缓冲区不被音频回放程序读空,则可以保证音频流的连续回放。 One solution to packet jitter at the receiving end is provided a jitter buffer, when the incoming data packet is first placed into the buffer, playback of audio data, the FIFO principle, uniformly removed in time audio packets and sent to broadcast the audio playback device; as long as the jitter buffer is not empty the program reads the audio playback, is guaranteed a continuous playback of audio streams. 该抖动缓冲区越大,则可平滑的抖动越大,但是过大的抖动缓冲区会造成音频回放时延的加大,过大的时延也是不可取的,会造成实时通讯的困难;抖动缓冲区的大小可根据具体需求来设定,也可以根据网络状况动态调整。 The larger the jitter buffer, the greater the jitter can be smooth, but too much can cause increased jitter buffer delay of audio playback, excessive delay is not desirable, it can cause difficulties in real-time communication; jitter the size of the buffer can be set according to specific needs, you may be dynamically adjusted based on network conditions. 但是这样并不能有效保证音频流的连续播放。 But this does not effectively ensure continuous playback audio stream.

发明内容 SUMMARY

本发明的目的是提供一种通话质量好、并能维持实时通讯的一种在网络终端缓冲区中处理音频流回放的方法。 Object of the present invention is to provide a good call quality, and to maintain real-time communication method of processing an audio stream playback terminal in the network buffer.

本发明是这样实现的:当包交换网络上传输来音频数据包进入抖动缓冲区时,计算读、写指针之间的距离D;在初始化过程中,设置读、写指针之间的距离D正常值的范围和一次从抖动缓冲区中读出的数据长度;在运行过程中,当读、写指针之间的距离D正常时,音频流按正常速度回放;当读、写指针之间的距离D大于或小于正常值时,通过音频重采样单元对从抖动缓冲区中读出的大于或小于正常播放长度的数据进行重采样处理,使其恢复正常的播放长度。 The present invention is achieved: When the transmitted audio packets into the jitter buffer on a packet switched network, computing a read, the distance D between the write pointer; during initialization, set to read, write pointer between the normal distance D range of values ​​and the length of the primary data read out from the jitter buffer; in operation, when the distance D between normal read and write pointer, the audio stream playback at normal speed; when the distance between the read and write pointer when D is greater than or less than normal, the data for re-sampling process to larger or smaller than the length of the normal playback is read out from the jitter buffer by the audio resampling unit, to restore the normal length of play.

上述的抖动缓冲区为环形抖动缓冲区。 The above-described jitter buffer is annular jitter buffer.

所述的读、写指针之间的距离D是为进入抖动缓冲区的读指针偏移量与写指针偏移量的差。 The read and write pointers are the distance D between the difference between pointer offset is the offset into the read pointer and the write jitter buffer.

当D小于正常值范围的下限时,从环形缓冲区读出的数据长度小于正常播放长度,采用升采样处理。 When D is smaller than the lower limit of the normal range, the length of data read from the ring buffer length less than the normal playback, using up-sampling process.

当D大于正常值范围的上限时,从环形缓冲区读出的数据长度大于正常播放长度,采用降采样处理。 When D is greater than the upper limit of the normal range, the length of data read from the ring buffer length is greater than the normal playback, using downsampling process.

当读指针的偏移量小于写指针的偏移量时,读、写指针之间的距离D为写指针的偏移量减读指针的偏移量所得的差。 When the read pointer offset is less than the offset of the write pointer, read, write pointer for the distance D between the write pointer minus read difference resulting offset offset pointer.

当读指针的偏移量大于写指针的偏移量时,读、写指针之间的距离D为环形抖动缓冲区的写指针的偏移量减读指针的偏移量所得的差在加上环形抖动缓冲区的长度所得的值。 When the difference resulting offset read pointer offset is greater than the write pointer offset, read, write, the distance D between the annular jitter buffer pointer write pointer minus offset read pointer plus annular jitter buffer length value obtained.

该方法在实现方式上既可以通过硬件编码实现也可以通过软件编码实现。 The method may also be implemented either by software implemented on the encoder implementation by hardware encoding.

采用上述方法后,当对读、写指针之间的距离D是非正常值时,通过对从抖动缓冲区中读出的大于或小于正常长度的语音数据块进行重采样处理,从而加速或减慢播放速度,最终使环形抖动缓冲区中的读指针、写指针始终保持一定的距离,保证了包交换网络上传输来的实时音频流的连续流畅播放,使通话连续、流畅,减少了卡音、断音等不连续现象。 With the method described above, when the read, write the distance D between the non-normal pointer, by re-sampling process on the voice data block is greater or less than the normal length read out from the jitter buffer to speed up or slow down playback speed, and finally to an annular jitter buffer read pointer, the write pointer always maintain a certain distance, to ensure continuous smooth playback of the real-time audio stream to packet switched transmission on the network, a call-continuous, smooth, reduces the sound card, staccato and other discontinuity.

附图说明 BRIEF DESCRIPTION

下面结合附图和具体的实施方式对本发明作进一步详述。 The present invention will be described in further detail in conjunction with the accompanying drawings and specific embodiments.

图1是包交换网络音频通讯示意图;图2是音频通信终端的发送部分示意图; FIG 1 is a schematic diagram for audio packet switched network; FIG. 2 is a partial schematic diagram of the transmission of the audio communication terminal;

图3是音频通信终端的接收部分示意图;图4是环形抖动缓冲区示意图。 FIG 3 is a schematic view of the receiving part of the audio communication terminal; FIG. 4 is a schematic view of an annular jitter buffer.

具体实施方式 detailed description

如图1所示,每个终端通过某种接入方式连接到包交换网络之中,每个音频终端可以通过该包交换网络向另一个音频终端发送或接收音频数据包;多个音频终端也可以通过某种形式组成一个可以多方通话的会议网络。 As shown, each terminal 1 is connected in some way into the packet-switched access network, each audio terminal may send or receive audio data packet to another terminal through which an audio packet-switched network; a plurality of audio terminals also may be composed of a multi-party conference calls through the network in some form.

如图2所示,音频输入信号首先送到音频采集单元,该音频采集单元完成音频信号从模拟到数字信号的转换,也就是量化过程,该处理过程一般将音频信号量化为16Bit精度的有符号数字信号,之后送入音频编码器进行数据压缩以节省网络带宽,音频信号经编码压缩之后,送入打包传送单元,数据打包单元一般将用于实时通讯的音频包按实时通信协议(RTP)标准进行封装,之后再封装为用户数据协议(UDP)包,最后打入互连协议(IP)包传送到网络上。 2, the first audio signal input to the audio capture unit, the audio signal acquisition unit to perform an audio signal from the analog to digital conversion, i.e. the quantization process, the process is generally the audio signal quantization precision signed 16Bit a digital signal, then fed to an audio encoder data compressed to save network bandwidth, the encoded audio signal is then compressed, packed into a transmission unit of data packing typically used for real-time communication unit audio pack according to real-time communication protocol (RTP) standard encapsulated, after repackaging the user data protocol (UDP) packets, and finally into the interconnection protocol (IP) packets to the network. 网络接口单元一般是网络层中的物理层及数组链路层,如以太网接口芯片或调制解调器等,经过压缩后的音频数据包最后经过网络接口单元传送到包交换网络上。 The network interface unit is typically the network layer and the physical layer link layer array, and the like such as a modem or Ethernet interface chip, through the compressed audio data packet and finally through the network interface unit to the packet-switched network.

在音频数据包到达目标接收终端之前,要通过一系列的网络传输单元;这可能包括多种交换设备和路由设备,不同的路由及网络状况会产生不同的传输时延,从而造成按等时间间隔顺序传送的音频数据包会在时间上非均匀地到达接收端,这就造成数据包接收的抖动;另外按等时间间隔顺序传送的音频数据包有可能在不同的路由上传送,比如顺序发送的数据包p1,p2,p3,...在接收端的顺序可能变成p1,p3,p2,...。 Before reaching the target audio packet reception terminal, to the network through a series of transmission units; this may include a variety of switching devices and routing device routes and different network conditions will produce different transmission delay, thereby causing at equal intervals audio packet transmission order may arrive at non-uniform time on the receiving side, which causes the received data packet jitter; additional sequence at equal intervals transmit audio data packets may be transmitted on different routes, such as the order of transmission of packet p1, p2, p3, ... in the order of the receiving side may become p1, p3, p2, ....

如图3所示;音频数据包经过包交换网络传输之后到达接收端的网络接口单元,网络接口单元将收到的数据包拆去物理层地址等信息后还原成互连协议(IP)数据包;该数据包接着被送入音频流接收及拆包单元,在该处理单元中拆去互连协议(IP)包头信息、用户数据协议(UDP)包头信息、实时通信协议(RTP)包头信息,最后还原为音频数据包;还原后的音频数据包被送入音频包解乱序单元,解出按正常时间顺序排列的数据包;按正常时间顺序排列的数据包接着被送入音频解码单元,解出音频信号的线性码;解出的音频信号的线性码被连续写入环形抖动缓冲区暂时存储;由于音频数据包经过网络传输之后出现的抖动,使音频数据流往环形抖动缓冲区的写入操作在时间上是非均匀的,但音频信号的回放在时间上要求是连续且均匀的,所以从环形抖动缓冲区读出音频 3; audio packets after transmission through the packet-switched network is reduced to reach the interconnection protocol (IP) data packet after receiving information of the network side interface unit, a network interface unit Remove packets received physical layer address; the packet is then sent to the audio stream receiving unit and unpacking, remove the interconnect protocol (IP) header information, user Datagram protocol (UDP) header, real-time protocol (RTP) header information included in the processing unit, and finally restore audio packet; the restored audio data packet is sent scrambled audio depacketizer unit, extract the data packet normal chronological; normal packets chronological order is then sent to an audio decoding unit, Solutions a linear code the audio signal; linear code solve the audio signal is continuously written into the ring buffer for temporarily storing jitter; audio packet jitter due to the transmission network after occurrence of the audio data stream is written to the ring buffer jitter non-uniform operation over time, but the playback of the audio signal in time requirement is continuous and uniform, so that from the ring read the audio jitter buffer 数据并回放的操作与往环形抖动缓冲区的写入操作时非同步的。 Data playback operation and asynchronous operation and to write the annular jitter buffer.

如下详细描述音频数据在环形缓冲区中的处理及回放过程,包括图3中的抖动缓冲器,抖动缓冲器读出单元,抖动缓冲器读出控制单元,音频重采样单元,音频回放单元;图3中实线为数据流,虚线位控制流。 Processing the following detailed description and playback of the audio data in the ring buffer, the jitter buffer including FIG. 3, the jitter buffer read unit, the jitter buffer read control unit, an audio resampling unit, an audio playback unit; FIG. 3 by the solid lines are data flows, dashed line control bit stream.

如图4所示,抖动缓冲区实际上是一个长度为N的连续存储空间,该存储空间的起始地址用偏移量0表示,结束地址用偏移量N-1表示,当前的写入地址指针用偏移量W表示,当前的读出地址指针用偏移量R表示,当前的写入指针与读出指针之间的距离用D表示;对该抖动缓冲区的写入操作为对当前写指针指向的存储位置写入音频数据,之后写指针偏移量加1并对抖动缓冲区的长度N取余运算,即用C语言可以表示为每写入一个数据后 W=(W+1)%N;对该抖动缓冲区的读出操作为每读出一个数据单元后读指针偏移量加1,同样用C语言可以表示为每读出一个数据后R=(R+1)%N;这样的读写操作可以保证在读写指针到达抖动缓冲区的顶部时,也就是偏移量N-1时,下一次读或写操作会自动翻转到抖动缓冲区的底部,也就是偏移量0;这样的操作实际上相当于将该抖动缓冲区的 4, the jitter buffer is actually a length of N consecutive memory space, the starting address of the memory space, an offset of 0 indicates the end address, an offset N-1, said current write offset address pointer is represented by W, the current read address pointer represented by the offset R, the distance between the current write pointer and the read pointer is represented by D; writing operation of the jitter buffer is current write pointer to write the audio data memory location, then the write pointer is incremented by 1 and the offset length jitter buffer operation modulo N, i.e., can be expressed as a C language after every write data W = (W + 1)% N; jitter buffer the readout operation after the readout of each data unit read pointer plus an offset, the same may be represented by C language after each read data R = (R + 1) % N; such read and write operations can be guaranteed when the read pointer reaches the top of the jitter buffer, i.e., the next read or write operation 1 N-offset will automatically flip to the bottom of the jitter buffer, i.e. offset 0; this operation is actually equivalent to the jitter buffer 顶部与底部相接,组成了一个环形缓冲区,读写指针按着相同的方向(顺时针或逆指针)旋转;为了保证读写指针之间不发生冲突,即读指针要保持始终跟随在写指针的后方,这是通过读写指针之间的距离D来判断的;如果W大于R则D=WR;如果W小于R则D=N+WR;由此只要D大于0则可以保证读指针始终跟随在写指针的后方,同时读写指针之间的距离D作为控制信号送入抖动缓冲区读出控制单元,实际上读写指针之间的距离D相当于一个闭环控制系统中的误差控制信号,作为对抖动缓冲区读出操作的依据。 Top and bottom phase, consisting of a ring buffer read pointer by pressing the same rotational direction (clockwise or counterclockwise pointer); in order to ensure read and write pointers do not conflict, i.e. the read pointer to remain always follow the write rear pointer, which is the distance D between the read and write pointers to determine; if R is larger than W D = WR; if R is less than W D = N + WR; whereby D is greater than 0 as long as it is guaranteed that the read pointer follows the write pointer is always behind, while the distance D between the read and write pointer jitter buffer as a control signal fed to the control unit reads out, in fact, the distance D between the read and write pointers corresponding to a closed loop error control system in the control signal, the jitter buffer as a basis for the read operation.

抖动缓冲区读出控制单元得到读写指针之间的距离D之后,根据D的大小来决定每次音频回放设备需要回放数据时读出的音频数据块的大小;音频回放单元一般是按等时间间隔(比如30毫秒)需要一定长度的音频数据块进行回放,如果音频信号的采样率为8000个采样点每秒(8k/s)则每30毫秒需要长度为240个采样点的音频数据块;若D的大小在正常的范围内,则抖动缓冲区读出控制单元控制抖动缓冲区读出单元每次在音频回放单元需要数据时读出正常长度的一块数据(比如240个音频数据),而此时音频重采样单元对此数据不进行任何操作,透明地将该块数据送到音频回放单元播放。 After the jitter buffer readout control unit to obtain the distance D between the read and write pointers, the size of D is determined according to the size of each audio data block read out the audio playback device when playback data required; the audio playback unit is generally at equal time audio blocks interval (such as 30 milliseconds) required for playback of a certain length, the required length of 30 ms per block of audio data 240 sampling points of the audio signal if the sampling rate is 8000 samples per second (8k / s); when the size D is within a normal range, the jitter buffer control means controls the readout jitter buffer readout unit reads out each piece of data (such as audio data 240) when the normal length of the audio playback unit needs data, and resampling the audio unit case no operation is performed on this data, the block data transparently to the audio playback unit playing.

如果读写指针之间的距离D超过正常范围,如过大或者过小则需要进行必要的调整,因为如果D过小,则可能发生由于数据包到来时间的随机抖动造成该抖动缓冲区被不时读空的情况出现,从而造成W小于R的情况出现,也就是出现读指针比写指针在环形缓冲区中跑的快的情况,这时就会出现断音与卡音;如果D过大,由于上述相同的原因,可能出现W绕过R一圈的情况出现,这时也会出现断音与卡音。 If the distance D between the read and write pointers above the normal range, too large or too small as is required to make the necessary adjustments, because if D is too small, it may occur due to random packet arrival time jitter caused by the jitter buffer is often empty case of a read occurs, resulting in occurrence of R W is smaller than the case, which is faster than the case where the read pointer is the write pointer in the ring buffer run occurred, then there will staccato tone of the card; if D is too large, For the same reason as described above, may occur where W R bypass lap occurs, then the card will appear staccato tone.

另外,网络传输上总会出现数据包丢失的情况,从长时统计平均的意义上来讲D会因此随着时间的推移越来越小;还有一个因素会影响D的变化,如果音频发送端的采样时钟频率不同于音频接收端的播放时钟频率,就会出现如下情况,即当音频发送端的采样时钟频率大于音频接收端的播放时钟频率时,D会因此随着时间的推移越来越大,当音频发送端的采样时钟频率小于音频接收端的播放时钟频率时,D会因此随着时间的推移越来越小。 In addition, there will always appear on the network transmission of packet loss situation, statistics on the average length of time from the sense D will therefore getting smaller and smaller as time goes on; there are factors that can affect a change in D, if the audio sender sampling clock frequency different from the clock frequency of the audio player receiving end, there will be a case, i.e., when the sampling frequency of the audio clock transmitting side is greater than the audio broadcast receiving terminal clock frequency, D will thus increasing over time when the audio sampling clock frequency of the transmission end is less than the audio broadcast receiving terminal clock frequency, D will thus become smaller and smaller over time.

假设上述综合原因允许D在大于d1小于d2的范围内变化,也就是说当d1<D<d2时,抖动缓冲区读出单元在相应的控制单元的控制下每次从抖动缓冲区中读出长度为L的音频数据块;当D<d1时,表示该抖动缓冲区很可能将要被读空,此时控制单元应控制抖动缓冲区读出单元在音频回放单元需要下一帧数据时,读出长度小于正常长度L的一帧数据,假定其长度为L1(L1<L);此时音频重采样单元就将长度为L1的音频数据块通过符合音频感知特性的插值运算变为长度为L的音频数据块,保证音频回放单元有标准长度的音频数据回放;这样相当于减慢了该帧音频数据的回放速度,只要L与L1的差值不是太大(比如((L-L1)/L)<1%)则主观听觉上不会有可感知的变化;由于减慢了音频数据的回放速度,可以预知D会因此越变越大,当D回到正常的范围内时(d1<D<d2)就可以按正常的 Assuming that the above multiple factors vary within the scope allowed D is less than d1 is greater than d2, that when d1 <D <d2, each jitter buffer reading unit reads out from the jitter buffer under control of a respective control unit length L block of audio data; when D <d1, represents the jitter buffer is likely to be read empty, then the control unit should control the jitter buffer reading unit when the audio data playback unit next frame needed, read a length less than the length L of a normal data, assuming a length of L1 (L1 <L); resampling the audio unit case will be a length of L1 block of audio data via the audio in line with the interpolation operation becomes a perceptual characteristic length L audio data blocks, audio playback unit to ensure that the audio data playback of standard length; this is equivalent to slow down the playback speed of the audio data of the frame, as long as the difference between L and L1 is not too large (for example, ((L-L1) / L) <1%) then there will be no appreciable change in subjective hearing; by slowing down the playback speed of the audio data, can predict D will therefore becoming larger and larger, when D back into the normal range (d1 < D <d2) can be according to the normal 度进行音频的回放了。 Of the audio playback.

当D>d2时,表示该抖动缓冲区很可能将要被写满,此时控制单元应控制抖动缓冲区读出单元在音频回放单元需要下一帧数据时,读出长度大于正常长度L的一帧数据,假定其长度为L2(L2>L);此时音频重采样单元就将长度为L2的音频数据块通过符合音频感知特性抽取运算变为长度为L的音频数据块,保证音频回放单元有标准长度的音频数据回放;这样相当于加速了该帧音频数据的回放速度,同样只要L与L2的差值不是太大(比如((L2-L)/L)<1%)则主观听觉上不会有可感知的变化;由于加快了音频数据的回放速度,可以预知D会因此越变越小,当D回到正常的范围内时(d1<D<d2)就可以按正常的速度进行音频的回放了。 When D> d2, represents the jitter buffer is likely to be filled, then the control unit should control the jitter buffer reading unit when the audio data playback unit next frame needed, read length greater than the length L of a normal frame data, assuming a length of L2 (L2> L); resampling the audio unit case will length L2 block of audio data by audio perception characteristic extraction operator compliance becomes a length L block of audio data, the audio playback unit to ensure audio data playback of standard length; this is equivalent to the accelerated playback speed of the frame of audio data, the same as long as the difference between L and L2 are not too large (for example, ((L2-L) / L) <1%) of the subjective listening on no appreciable change; as accelerated playback speed of the audio data, you can predict D will therefore get smaller when the D back into the normal range (d1 <D <d2) can be normal speed the audio playback.

如果数据包到来时间的抖动范围太大或者丢包率太大,则有可能造成环形缓冲区被读空或者写满,此时可以做异常处理,在该缓冲区被读空时,可以重放上一帧语音数据,若紧接着的下一次读操作时缓冲区仍为空,则播放静音信号;如果环形缓冲区被写满,则自动冲掉该缓冲区中的所有未播放数据,并重新开始正常的对该缓冲区的正常读写操作。 If the packet arrival time jitter range is too large or the packet loss rate is too large, it may cause the ring buffer is full or empty is read, exception handling can be done at this time, when the read buffer is empty, can be reproduced on a voice data, if the subsequent read operation when the buffer is still empty, the player mute signal; if the ring buffer is filled, the automatic play data does not wash away all the buffer, and re- beginning the normal operation of the normal read and write buffer.

该方法在实现方式上既可以通过硬件编码实现也可以通过软件编码实现。 The method may also be implemented either by software implemented on the encoder implementation by hardware encoding.

Claims (8)

1.一种在网络终端缓冲区中处理音频流回放的方法,其特征在于:当包交换网络上传输来音频数据包进入抖动缓冲区时,计算读、写指针之间的距离D;在初始化过程中,设置读、写指针之间的距离D正常值的范围和一次从抖动缓冲区中读出的数据长度;在运行过程中,当读、写指针之间的距离D正常时,音频流按正常速度回放;当读、写指针之间的距离D大于或小于正常值时,通过音频重采样单元对从抖动缓冲区中读出的大于或小于正常播放长度的数据进行重采样处理,使其恢复正常的播放长度。 A method of processing an audio stream playback buffer in the network terminal, wherein: when the transmitted audio packets into the jitter buffer on a packet switched network, computing a read, the distance D between the write pointer; initialization process, a read, write the normal range of the distance D and the length of a data read out from the jitter buffer between the pointer; during operation, when the distance D between normal read and write pointer, the audio stream normal-speed playback; when the distance D between the read and the write pointer is greater than or less than normal, the data for re-sampling process to larger or smaller than the length of the normal playback is read out from the jitter buffer by the audio resampling means that the restore normal length of play.
2.根据权利要求1所述的一种在网络终端缓冲区中处理音频流回放的方法,其特征在于:所述的抖动缓冲区为环形抖动缓冲区。 The method for processing one of said one audio stream playback buffer in a network terminal as claimed in claim, wherein: said annular jitter buffer jitter buffer.
3.根据权利要求1或2所述的一种在网络终端缓冲区中处理音频流回放的方法,其特征在于:所述的读、写指针之间的距离D是为进入抖动缓冲区的读指针偏移量与写指针偏移量的差。 3. According to one of claim 1 or claim 2 method for processing the audio stream playback buffer in the network terminal, wherein: said read, write distance D between the pointer is read into the jitter buffer offset difference between the write pointer and the pointer shift amount.
4.根据权利要求1或2所述的一种在网络终端缓冲区中处理音频流回放的方法,其特征在于:当D小于正常值范围的下限时,从环形缓冲区读出的数据长度小于正常播放长度,采用升采样处理。 4 according to one of claim 1 or claim 2 method for processing the audio stream playback buffer in the network terminal, wherein: when D is smaller than the lower limit of the normal range, the length of data read from the ring buffer is less than normal play length, using up-sampling process.
5.根据权利要求1或2所述的一种在网络终端缓冲区中处理音频流回放的方法,其特征在于:当D大于正常值范围的上限时,从环形缓冲区读出的数据长度大于正常播放长度,采用降采样处理。 5 according to one of claim 1 or claim 2 method for processing audio streams in the network terminal playback buffer, characterized in that: when D is greater than the upper limit of the normal range, the length of data read from the ring buffer is greater than normal play length, using downsampling process.
6.根据权利要求3所述的一种在网络终端缓冲区中处理音频流回放的方法,其特征在于:当读指针的偏移量小于写指针的偏移量时,读、写指针之间的距离D为写指针的偏移量减读指针的偏移量所得的差。 6. The method of claim 3 audio stream playback processing terminal in a network buffer claim, wherein: when the read pointer offset is less than the offset of the write pointer, read and write pointers between the difference between the distance D is the write pointer minus offset of the read pointer offset obtained.
7.根据权利要求3所述的一种在网络终端缓冲区中处理音频流回放的方法,其特征在于:当读指针的偏移量大于写指针的偏移量时,读、写指针之间的距离D为环形抖动缓冲区的写指针的偏移量减读指针的偏移量所得的差在加上环形抖动缓冲区的长度所得的值。 7. The method of claim 3 audio stream playback processing terminal in a network buffer claim, wherein: when the read pointer offset is greater than the write pointer offset, the read, write pointer between the annular distance D jitter buffer write pointer minus the difference between the obtained offset read pointer offset plus the jitter value obtained by an annular length of the buffer.
8.根据权利要求1或2所述的一种在网络终端缓冲区中处理音频流回放的方法,其特征在于:该方法在实现方式上既可以通过硬件编码实现也可以通过软件编码实现。 8. According to one of claim 1 or claim 2 playback method for processing an audio stream, characterized in that the network terminal buffer: This method may be realized by software may be implemented by hardware coding encoded on the implementation.
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