WO2009106015A1

WO2009106015A1 - Dynamic bit rate allocation method, packet-domain streaming media server

Info

Publication number: WO2009106015A1
Application number: PCT/CN2009/070584
Authority: WO
Inventors: 罗泽文; 仇刚
Original assignee: 华为技术有限公司
Priority date: 2008-02-27
Filing date: 2009-02-27
Publication date: 2009-09-03
Also published as: CN101242359A; CN101242359B

Abstract

A dynamic bit rate allocation method and a packet-domain streaming media server, in which the method includes that: when judging that the bit rate needs to be adjusted according to the change of the bandwidth of the user equipment, the bit rate switching is performed; sending the switched bit rate media data.

Description

Dynamic rate allocation method, ^ and domain streaming server

This application claims priority to Chinese Patent Application No. 200810005294.1, entitled "Dynamic Bit Rate Allocation Method, Packet Domain Streaming Media Server", filed on February 27, 2008, the entire contents of which are incorporated by reference. In this application.

Technical field

The present invention relates to the field of communications technologies, and in particular, to a Dynamic Bandwidth Allocation (DBA) method and a packet domain streaming media server.

Background technique

Streaming media is a technology that compresses video and audio information and provides it to users for viewing in the form of media streams. Streaming media services in the field of mobile communications are called mobile streaming media, and mobile streaming media is common. As shown in FIG. 1 , the end-to-end networking structure includes: a User Equipment (UE) located in a packet domain, a Radio Network Controller (RNC) 102, and a General Packet Radio Service Support Node (GGSN). Gateway GPRS Support Node 103; a packet-switched streaming server (PSS) 104 and a portal 105 located in an Internet Protocol (IP) network.

The UE 101 is controlled by the RNC 102, and allocates radio resources through the RNC 102, including the quality of service (QoS) of the wireless connection, accesses the IP network through the GGSN 103, and connects to the server PSS 104 that provides the mobile streaming service through the Internet (Internet), showing the mobile Streaming service server Portall05.

Video on Demand (VOD, Video On Demand) and Live TV (Live TV) are two of the more common mobile streaming services. The PSS 104 pushes the media stream to the client at a specified bit rate, ie UE10K in an ideal situation, UE The receiving and processing capabilities are consistent with the bit rate of the PSS sending streaming media to achieve optimal playback. The receiving capability of the UE is determined by the bandwidth of the UE. Generally, due to the mobility of the UE, the bandwidth of the UE is often fluctuated. Therefore, the quality of the UE may be good or bad when playing the streaming media service. To solve this problem, DBA technology is proposed, which assigns different code rates according to the bandwidth that the client can support.

The more common DBA technology is to use the packet loss rate to determine whether to change the code rate and how to change the code rate.

Summary of the invention The technical problem to be solved by the embodiments of the present invention is to provide a dynamic rate allocation method and a packet domain streaming media server, so that the playback of the mobile streaming media is smoother when switching.

To solve the above technical problem, an aspect of an embodiment of the present invention provides a dynamic rate allocation method, including:

According to the change of the bandwidth of the user equipment, when it is determined that the code rate needs to be adjusted, the code rate is switched; and the media data after the switching code rate is sent.

In another aspect, a packet domain streaming server is provided, including:

A unit for performing rate switching when determining that a code rate needs to be adjusted according to a change in bandwidth of the user equipment;

A unit for transmitting media data after switching the code rate.

It can be seen from the above technical solution that, according to the embodiment of the present invention, when the bandwidth of the user equipment changes, it is determined that the code rate needs to be adjusted, and the code rate is switched, thereby effectively improving the quality of the transmission.

In another aspect, a packet domain streaming server is provided, including:

a round-trip time obtaining unit, configured to obtain a round-trip time of the sender report packet;

a first round-trip time determining unit, configured to determine whether the round-trip time meets a predetermined condition; a first switching unit, configured to perform a code rate switching when the round-trip time satisfies a predetermined condition; and a sending unit, configured to send a switching code rate After the media data.

It can be seen from the above technical solution that, according to the embodiment of the present invention, the round-trip time of the sender report packet is used to determine the switching timing, thereby avoiding the use of the packet loss rate for judging, and when the data is lost, the undercut is performed, resulting in the mosaic. phenomenon.

In another aspect, a packet domain streaming server is provided, including:

a bandwidth determining unit, configured to determine, according to a change in bandwidth of the user equipment, whether the code rate needs to be adjusted;

a second complete frame determining unit, configured to determine whether the current transmission frame is a complete frame;

The second switching unit is configured to: when the code rate needs to be adjusted, and the current transmission frame is a complete frame, perform rate switching, and send the media data after the switching rate.

As can be seen from the above technical solution, since the embodiment of the present invention selects an I frame for switching during handover, quality problems such as unsmooth playback or mosaic caused by switching in a B frame or a P frame are avoided.

DRAWINGS 1 is a schematic diagram of an end-to-end networking structure of a mobile streaming media in the prior art; a schematic diagram of an input embodiment; a wide variation curve, a PSS packet rate change curve, an RNC buffer data change curve, and a UE buffer data change curve;

4 is a structural diagram of an embodiment of a packet domain streaming media server according to an embodiment of the present invention;

FIG. 5 is a structural diagram of another embodiment of a packet domain streaming server according to an embodiment of the present invention.

detailed description

The embodiment of the invention provides a dynamic rate allocation method and a packet domain streaming media server, which can effectively improve the playback quality of the mobile streaming media service.

The packet loss rate is used as the basis for adjusting the code rate. When the packet loss has occurred, the handover is performed. The handover is lagging behind, and the playback quality is often poor and has been adjusted for a while. Especially for the case where the code rate needs to be adjusted downward, that is, the code rate is cut, the code rate is lowered when the packet loss rate reaches a certain level, indicating that the bandwidth of the UE side is lower than the code rate and the packet loss occurs for a period of time. The media server only noticed that the bit rate was cut. The feeling for the user was that the picture returned to normal after a period of mosaic, and the experience brought to the user was very poor.

The dynamic rate allocation method provided by the present invention includes:

Can effectively improve the quality of transmission.

Currently, streaming media generally uses Real Time Streaming Protocol (RTSP) I Real Time Transport Protocol (RTP)/RTP Control Protocol (RTCP) protocol cluster to control/transmit streaming media data. Among them, RTSP is used to establish and control the time synchronization stream of continuous media. RTSP is a text protocol and is similar to the Hypertext Transfer Protocol (HTTP). The main difference is that RTSP is a standard streaming protocol and usually uses an independent transport protocol such as RTP to transmit media data. RTP is used to provide time information and to synchronize audio/video streams. RTP does not process resource reservations and does not guarantee the quality of service for real-time services. The main function of RTCP is to provide feedback on the transmission of data. Sending end After the user clicks on the RTP data packet, the RTCP sender report (SR, Sender Report) packet is sent at regular intervals to perform statistics on the sender data transmission. The receiving end periodically sends a message receiver (RR, Receiver Report) packet to the sender.

When sending an SR packet, the PSS adds the timestamp of the transmission time to record the time when the SR packet is sent. When the UE receives the SR packet, it does not immediately return the RR packet. Usually, it waits for a period of time. After processing, the RR packet is sent again. When the UE sends the RR packet, the UE records the time (LSR, Last SR) of the last received SR packet sent from the PSS in the RR packet, that is, the SR recorded in the SR packet. The packet transmission time also records in the RR how long the UE waits for the SR packet to be sent after receiving the SR packet, that is, the delay of receiving the SR packet to the transmission time (DLSR, delay since last SR). The time that the SR packet is sent from the PSS to the UE, plus the time that the UE returns the RR packet to the PSS, may be called the round-trip time of the SR packet (RTT, Round-Trip Time), and the RTT does not include the UE receiving the SR packet. Waiting time, therefore, assuming that the current time that the PSS receives this RR packet is A, the RTT is calculated as: RTT = A - LSR - DLSR.

In the embodiment of the present invention, after receiving the RR packet, the PSS reads the LSR and the DLSR, calculates the RTT, and determines whether the code rate needs to be switched through the RTT.

Both LSR and DLSR can be read directly from the corresponding fields of the standard RTCP RR packet. The first embodiment of the dynamic rate allocation method provided by the embodiment of the present invention can be divided into two steps: an RTCP analysis phase and a bandwidth adjustment.

The reported RTCP packet is analyzed in the RTCP analysis phase to calculate the RTT.

The calculation formula of RTT in i time period can be expressed as: RTTi = Ai - LSRi - DLSRi; i indicates that the i-pack RR packet is received.

In the bandwidth state adjustment phase, according to the RTT size, select an appropriate rate adjustment strategy.

Set an RTT range according to the actual situation of the network. When the RTT size is within the range, it belongs to the normal state of the network. You can adjust it without adjustment. When the RTT size is outside the range, you need to adjust the code rate. Assume that the maximum value of the RTT range is RTTmax and the minimum value is RTTmin. The method of adjusting the code rate under different conditions is as follows:

1. When RTTi > RTTmax and RTTi > RTTi-1, it indicates that the SR packet has serious delay and becomes more and more serious. At this time, the media stream is adjusted to the low code rate, that is, the undercut is performed. Among them, n times RTTi > RTTmax means that RTT exceeds the upper limit for consecutive n time periods, and the delay is more serious. To perform the undercut, RTTi > RTTi-1 indicates that the RTT of this time period is greater than the RTT of the previous time period. The reason why this condition needs to be increased is because when the RTTi > RTTmax is continuously n times, the RTT is adjusted to the low bit rate media stream. It can't return to less than RTTmax in a short time. At this time, there will usually be a short period of time. RTTi is still greater than or equal to RTTmax. If RTTi > RTTi-1 is not added, the system may continue to perform unnecessary switching. Therefore, after judging the continuous n times RTTi > RTTmax, it is necessary to determine whether RTTi is greater than RTTi-1. If the undercut operation adjusted to the low bit rate media stream has started, then RTTi will be less than RTTi-1, and no undercut is required. If the result is that RTTi is greater than RTTi-1, it indicates that the SR packet has a serious delay and is getting more and more serious. It needs to be adjusted to the low-rate media stream.

n is to prevent the value set by the normal jitter of the network. It can be configured according to the actual situation. It can be one time or multiple times. When the value of n is greater than "1", it will not occur when RTTi > RTTmax Immediately adjust to low bit rate media streams to prevent the effects of normal jitter on the network.

2. When RTTi < RTTmin is continuous for m times, it indicates that the network is in a smooth state for a long time. At this time, the media stream is adjusted to the high code rate, that is, the upper cut is performed. Where m is the value set to prevent the normal jitter of the network, and can be configured according to the actual situation, either once or multiple times; when the value of m is greater than "1", it will not occur when RTT RTTmin occurs. Immediately adjust to low bit rate media streams to prevent the effects of normal jitter on the network.

3. When RTTmin <RTTi < RTTmax, it indicates that the network is in a normal state. At this time, the current code rate state is maintained. When the value of η is greater than "1", the number of RTTi > RTTmax is less than n times, which can be considered as a network. The normal jitter can also maintain the current bit rate state; when the value of m is greater than "1", the number of RTTi < RTTmin is less than n times, which can also be considered as the normal jitter of the network, and can also maintain the current bit rate state.

In actual use, the value of n needs to consider the buffer size of the RNC and the UE. After the continuous RTTi > RTTmax, the RNC and the UE buffer can save all the data that cannot be sent, usually continuous. The time taken for n times RTTi > RTTmax can be 2 or 3 RTCP packet transmission/reception cycles.

Further, since the data of the mobile streaming media generally includes three data frames: an I frame, a P frame, and a B frame. The I frame is a frame completely encoded by the entire picture, and is also called an internal frame, a complete frame, and a key frame. I frames can be decoded independently. The I frame can also provide a reference for the decoding of the associated P frame and B frame. A P frame is a frame that encodes a difference from the previous frame. Only the difference data with the previous frame is recorded, and the display cannot be decoded independently. It must be decoded before the I frame it refers to.

A B frame is a frame that encodes a difference from a previous frame and a subsequent I frame. The B frame cannot be decoded independently, and must be decoded when the I frame it refers to is received in advance.

If the B frame is being transmitted while switching, the subsequent I frame referenced by the B frame will not be received. When the B frame is played, quality problems such as unsmooth playback or mosaic will occur.

If the P frame is being transmitted during handover, the P frame that is encoded after the handover is used for encoding, and the P frame after the handover is different from the previous I frame, and may not be able to refer to the previous frame. An I frame decodes the complete image, and at the same time there will be quality problems such as unsmooth playback or mosaic.

The prior art only considers switching when the packet loss rate reaches a certain level, and it is very common to switch between B frames or P frames. Therefore, quality problems such as unsmooth playback or mosaic due to switching often occur.

Therefore, in the first embodiment of the dynamic rate allocation method provided by the embodiment of the present invention, a technical solution for switching in an I frame is given. After determining that the frame to be transmitted is an I frame after determining that the handover is required according to the RTT size, Switching, since the I frame is a completely encoded frame of the entire picture, it can be decoded and displayed independently. Do not refer to the data before and after, so there is no problem that the complete image cannot be produced, and the switching between B frames or P frames is avoided. Play quality problems such as smooth or mosaic.

The two forms of switching are mentioned above: upper and lower cuts. The bit rate is cut upwards, and the high-rate media stream is adjusted to indicate that the UE bandwidth is much larger than the current code rate and the upper code rate is required. At this time, as long as the I frame is switched, quality problems such as unsmooth playback or mosaic can be avoided. The code rate is cut down, and the adjustment to the low bit rate media stream indicates that the UE bandwidth cannot meet the current code rate requirement, resulting in a down code rate, which is complicated. In order to avoid possible quality problems such as unsmooth playback or mosaic, the present invention The code rate undercut method given in Embodiment 1 of the dynamic rate allocation method provided by the embodiment is as follows:

In the radio access network (RAN, Radio Access Network), the RNC sets a buffer for each UE, and the buffer size is preferably a program that can store the maximum buffer data that may appear in the RNC, in each case. Established when the UE establishes a connection relationship with the RNC. The maximum buffer data that may appear in the RNC actually relates to the transmission period of the RTCP packet, and the above mentioned n, that is, the maximum data that the RNC may buffer after consecutive n times of RTTi > RTTmax. In the case of a smooth network, The time it takes to transmit a packet is considered to be very short. Under normal circumstances, it will not exceed the second level, but in the case of a network failure, the packet will be stuffed into the buffer. In order to prevent network jitter or other slightly unstable factors from causing unsmooth playback or degradation of playback quality, a buffer is also required on the UE to buffer media data, and the size of the buffer also needs to be able to store the UE. The maximum buffered data that may appear in practice. At the same time, when the PSS switches from a high code rate to a low code rate, the SR packet has a serious delay and becomes more and more serious. The buffer of the RNC has accumulated a large amount of data, and the bandwidth of the UE has dropped to a short time. The above accumulated data is transmitted to the UE. If the UE does not have a buffer or the buffer is relatively small, it will definitely cause the UE to play unsmoothly or even mosaic due to the inability to obtain the media data to be played in time.

Now, an example is shown to switch the PSS from a high code rate to a low code rate, that is, the data transmission condition when performing the undercut, and a schematic diagram of the data transmission embodiment when performing the undercut is shown in FIG. 2:

In Fig. 2, the vertical axis of the coordinate indicates the size of the buffer. The unit is Kbit. The PSS has just switched from 384Kbit to 128Kbit. The media rate of the RNC is 128Kbit. The receiving rate is

128Kbit/S, the received media data is buffered in the RNC buffer. At this time, the media data rate of the UE buffer buffer is still 384Kbit, and the rate of transmitting media data to the player is still

At 384 Kbit/s, the UE bandwidth has dropped to 150 Kbit/s, and the rate of data transmission between the RNC buffer and the UE buffer is 150 Kbit/s.

At this time, the RNC buffers 3840Kbit of data, that is, buffers the data rate of 384Kbit for 10 seconds. This time is the time for the PSS to detect the RTT abnormality and the next code rate, that is, after n consecutive RTTi > RTTmax Cached data. The UE buffers 1536Kbit data, that is, the buffer rate is 4 seconds.

384Kbit of data, since this data is 384Kbit code rate data, so the playback speed is still

384Kbit/S.

At this time, the bandwidth of the UE side has been reduced to 150 Kbit/s, and the code rate of the PSS is cut to 128 Kbit/s. After 4 seconds, when the UE plays the first buffered 1536Kbit data, it buffers 150 Kbit χ 4 = 600Kbit data, that is, the RNC transmits 600Kbit data to the UE, but the RNC also buffers 128 Kbit X 4 = 512Kbit of data; and so on, after about 600/(384 - 150) = 1.65 seconds, the UE can play the data of the UE buffer. At this time, the data rate of the RNC buffer buffer is still 384Kbit. Since the UE buffer is already empty, the data received by the UE at a rate of 384Kbit transmitted at a code rate of 150K can only be played out immediately, so there will be some time. 384Kbit code rate The data is transmitted at a rate of 150K, although it can be guaranteed not to lose packets, but the playback is certainly not smooth. PSS adjusts the bit rate down to reduce the time for this to happen. In order to completely solve this problem, you need

The UE buffer and RNC buffer size are configured to a ratio according to the bit rate level that the PSS can support and the bandwidth level supported by the RAN network.

For example, the PSS is switched from a high code rate to a low code rate, that is, when the undercut is performed, the UE bandwidth change curve, the PSS packet rate change curve, the RNC buffer data change curve, and the UE buffer data change curve are shown in FIG. :

Curve 1 is a curve indicating the change of UE bandwidth, and curve 2 is a curve indicating the change of PSS packet rate. The coordinate system is located, the horizontal axis is the time axis, the unit is seconds, and the vertical axis is the bandwidth/code rate, and the unit is Kbit/S. UEBWup indicates a relatively high bandwidth of the UE before the bandwidth is dropped, UEBWup may also be referred to as a high bandwidth of the UE; PSSBWup indicates a code rate corresponding to the PSS and the UEBWup, that is, a higher code rate matching the high bandwidth of the UE. PSSBWup may also be referred to as a high code rate; UEBWlow indicates a relatively low bandwidth of the UE after the bandwidth is dropped, and UEBWlow may also be referred to as a low bandwidth of the UE;

PSSBWlow indicates the bit rate corresponding to PSS and UEBWlow, that is, the lower bit rate that matches the low bandwidth of the UE. PSSBWlow can also be called low bit rate.

Curve 3 is a curve showing the RNC buffer change, its coordinate system, the horizontal axis is the time axis, the unit is seconds, the vertical axis is the RNC cache data amount; RNCBUFmax represents the maximum buffer capacity of the RNC, that is, the RNC buffer Maximum capacity; RNCRBUFmax represents the actual maximum buffered data of the RNC, that is, the maximum amount of data buffered by the RNC buffer.

The curve 456 is a curve indicating the UE buffer change; the UEBUFmax indicates the maximum buffer capacity of the UE, that is, the maximum capacity of the UE buffer, and the UERBUFmax indicates the actual maximum buffer data of the UE, that is, the UE buffer buffer actually. The maximum amount of data.

The time point T0 indicates the time when the UE bandwidth drops from the high bandwidth to the low bandwidth, and the time point T1 indicates

The PSS detects that the bandwidth change of the UE actively switches the code rate from the high code rate to the low code rate. The time point T2' indicates the time when the RNC transmits the high code rate data, and the time point Τ2 indicates that the UE plays the high code rate data. At the same time, the buffer is also cleared. The time point T3 indicates the time when the buffer of the RNC is cleared. At this point, the end-to-end network is officially restored to normal.

As shown in FIG. 3, from time T0 to time T3, that is, from the time when the UE changes bandwidth to when the PSS adjusts the code rate in time and stabilizes, RNCRBUFmax<RNCBUFmax and UERBUFmax < UEBUFmax, therefore, data packets are not lost, and data that cannot be sent or broadcast in time is stored in the buffer area. In the case of complete data, as long as the I frame is switched, the quality such as mosaic can be avoided. problem.

During the TO~T1 time period, the PSS detects the change of the UE bandwidth and actively adjusts the code rate during this time period;

In the period of Τ1 ~ Τ2': RNC receives the low bit rate data at the speed of PSSBWlow, and transmits the high bit rate data at the speed of UEBWlow. Because UEBWlow > PSSBWlow, the buffer data of RNC is gradually reduced, and it is not high until T2' time. The rate rate output is completed;

During the period from T1 to T2: The UE plays the high bit rate data at the speed of PSSBWup, and receives the high bit rate data at the speed of UEBWlow. Because PSSBWup > UEBWlow, the buffered data of the UE is gradually reduced until the T2 time, the UE buffer is reduced. Is 0.

When T2 > T2', it indicates that the RNC has transmitted the high bit rate data before the UE buffer has not been cleared. After the T2' time point, the UE receives the new low bit rate data at the speed of UEBWlow. After the PSSBWup speed is played, the buffered code rate is the old data of PSSBWup, and the new low bit rate data is played at the speed of PSSBWlow. Because PSSBWlow < UEBWlow, the data buffered by the UE buffer is gradually increased after T2' until T3. After the time point, the RNC clears the buffered data, and how much data is output by the RNC. At this point, the network is unobstructed and the UE buffer is stable. The curve of the UE buffer change after the T2' time point of the UE is shown as curve 5 in the figure. In the above process, for the UE broadcaster, its data stream is always smooth, so there is no quality problem in playback.

When T2 = T2', it indicates that the RNC has finished transmitting the high bit rate data, and the UE just clears the buffer. After that, the UE receives the new low bit rate data at the speed of UEBWlow, and plays the new low code at the speed of PSSBWlow. The rate data, the curve of the UE buffer change curve is shown as curve 6 in the figure. In the above process, for the UE player, its data stream is always smooth, so there is no quality problem in playback.

When T2 < T2', it indicates that the RNC clears the buffer before the RNC transmits the high bit rate data, so during the period Τ2 ~ Τ2', the high bit rate data is transmitted to the UE by the low bandwidth, and the UE The buffer is empty, and the received high bit rate data can only be played at a low bit rate. In this case, the playback is certainly not smooth, and the user experience is that the play has a pause.

Note that the time (DT, Detect Time) required by the PSS to detect the UE bandwidth transmission abnormality is DT. DT can be expressed as: DT = T1-T0. It can be seen from Figure 3 that the size of DT will affect RNCRBUFmax. The larger DT is, the larger the RNCRBUFmax is. The R packet of R packets is assumed to be P, and the number of consecutive RTTis is > When RTTmax is detected, the PSS detects that the UE bandwidth transmission abnormality needs to be cut. The relationship between DT, n, and P can be expressed as:

DT = n XP ( 1 ) During TO to T1, the RNC buffer receives the high bit rate data at the PSSBWup speed and outputs the data to the UE at the UEBWlow speed. The difference between the two is multiplied by the time interval DT, which is RNCRBUFmax. The relationship can be expressed. For:

RNCRBUFmax = DT x (PSSBWup - UEBWlow) ( 2 ) The data before T2 is the PSSBWup data of the bit rate, so the playback speed of the UE player is also PSSBWup, but at this time, the transmission speed of the UE has dropped to UEBWlow, so padding The speed of the UE buffer is UEBWlow. Therefore, the current UE actual buffer is divided by the difference between the above two speeds, that is, the time required for the UE to clear the buffer. The time required for the UE to clear the buffer is CTUE, and the CTUE can be expressed as :

CTUE = T2 - TO = UERBUFmax / (PSSBWup - UEBWlow) (3) The same time required for RNC to clear the code rate for PSSBWup data is CUTRNC, CUTRNC can be expressed as:

CUTRNC = T2' - T1 = RNCRBUFmax I ( UEBWlow - PSSBWlow ) ( 4 ) According to the above analysis, 播放2 > Τ2' can ensure that there is no quality problem, ie CTUE > CUTRNC, according to equation (3), formula (4) ), Available:

UERBUFmax > RNCRBUFmax x (PSSBWup - UEBWlow ) / ( UEBWlow - PSSBWlow ) ( 5 )

In practical applications, the code rate of each level, such as PSSBWup, PSSBWlow, etc., is relatively fixed, and the receiving bandwidth of the UE, such as UEBWup and UEBWlow, is uncontrollable. Therefore, the dynamic code provided by the embodiment of the present invention is The first embodiment of the rate allocation method further provides the following solutions:

The sizes of UERBUFmax and RNCRBUFmax are controlled so as to satisfy the relationship of equation (5), and generally the larger the UERBUFmax or the smaller the RNCRBUFmax, the better.

According to the formula (2), the RNCRBUFmax size and the time required for the PSS to detect the UE bandwidth change DT related, PSS is detected according to the RR packet received by RTCP, so DT is related to the period P and n of sending RTCP packets. By reducing the value of n, the DT is made smaller to achieve RNCRBUFmax. The purpose of reduction.

In the first embodiment of the dynamic rate allocation method provided by the embodiment of the present invention, the timing of the handover is determined according to the round trip time RTT of the SR packet, and how to switch, and when the undercut is needed, the packet loss does not wait. Switching again, avoiding quality problems such as mosaics and pauses caused by packet loss.

Further, the I frame is selected for switching at the time of switching, thereby avoiding quality problems such as unsmooth playback or mosaic caused by switching in the B frame or the P frame.

Further, in order to avoid that the UE buffer area is emptied before the RNC transmits the high code rate data, and the received high code rate data is played at a low code rate, the playback is not smooth, and the maximum buffered data by controlling the RNC is given. The UE actually buffers the largest amount of data to avoid the UE buffer area being emptied before the RNC sends the high code rate data.

Therefore, the second embodiment of the dynamic code rate allocation method provided by the embodiment of the present invention can be judged in the second embodiment of the dynamic code rate allocation method provided by the embodiment of the present invention. When the bandwidth of the UE changes, when I need to switch, select the I frame to switch:

First, according to the change of the bandwidth of the user equipment, it is determined whether the code rate needs to be adjusted;

When it is judged that the code rate needs to be adjusted, it is judged whether the frame to be transmitted is an I frame, and when the current transmission frame is a complete frame, the code rate is switched. According to the change of the bandwidth of the user equipment, it is determined that the code rate needs to be adjusted, and the bandwidth change may be determined according to the round-trip time, that is, the round-trip time of the sender report packet is obtained; when the round-trip time meets the predetermined condition, it is determined that the code rate needs to be adjusted. In this case, the detailed agreement is not repeated here.

It is also possible to determine the bandwidth change according to the packet loss rate, that is, to obtain the packet loss rate; when the packet loss rate satisfies the predetermined condition, it is determined that the code rate needs to be adjusted.

The detailed method of performing rate adjustment by determining the bandwidth change by the packet loss rate is as follows:

The streaming media server analyzes the reported RTCP packets and calculates the packet loss rate.

i time period packet loss rate algorithm: Ri=f_lost/(Seq_numberi-Seq_numberi-l); where f_lost is the total number of RTP drops between adjacent RTCP RR fields; Seq_numberi is the i-th RTCP packet Report the received RTP packet sequence number; Seq_numberi-1 is the RTP packet sequence number reported by the i-1th RTCP packet;

According to the degree of packet loss:

When Ri>=Rmax for n consecutive times, n is a predetermined value, which is a serious packet loss, and is selected to adjust to the low code stream; when 0<Ri<=Rmax, or occasionally Ri=0, or occasionally Ri>=Rmax, it is mild. Lose the packet and maintain the current bit rate status;

When n times for Ri = 0, m is a predetermined value, which belongs to long time without packet loss, and chooses to adjust to high code stream. In the second embodiment of the dynamic bit rate allocation method provided by the embodiment of the present invention, an I frame is selected for switching during handover, which avoids quality problems such as unsmooth playback or mosaic caused by switching in a B frame or a P frame.

It will be understood by those skilled in the art that all or part of the steps of implementing the foregoing embodiments may be performed by a program to instruct related hardware, and the program may be stored in a computer readable storage medium. , including the following steps:

A dynamic rate allocation method includes:

Get the round trip time of the sender report package;

Determining whether the round trip time meets a predetermined condition;

Performing code rate switching when the round trip time satisfies a predetermined condition;

Send the media data after switching the bit rate.

A dynamic rate allocation method includes:

According to the change of the bandwidth of the user equipment, when it is determined that the code rate needs to be adjusted, when the current transmission frame is a complete frame, the code rate is switched, and the number of media after the switching code rate is transmitted.

The above-mentioned storage medium may be a read only memory, a magnetic disk or an optical disk or the like.

The structure of the packet domain streaming server according to the embodiment of the present invention is as shown in FIG. 4, and includes: a round-trip time obtaining unit 410, configured to acquire a round-trip time of a sender report packet;

The first round-trip time determining unit 420 is configured to determine whether the round-trip time meets a predetermined condition. The first switching unit 430 is configured to perform code rate switching when the round-trip time meets a predetermined condition, and the sending unit 440 is configured to send Switch the media data after the bit rate.

The first complete frame determining unit 450 is configured to determine whether the current transmission frame is a complete frame. When the current transmission frame is a complete frame, the first switching unit 440 is further controlled to perform rate switching. The round trip time obtaining unit 410 includes:

The sending time obtaining unit 411 is configured to acquire a sending time of the sender report packet carried in the receiver report packet.

The delay obtaining unit 412 is configured to obtain a delay of the last received sender report packet carried in the receiver report packet to the sending time;

The processing unit 413 is configured to obtain a round trip time of the sender report packet according to a sending time of the sender report packet carried in the receiver report packet, and a delay of receiving the sender report packet to the sending time.

The first round trip time judging unit 420 includes:

The upper cut determining unit 421 is configured to determine whether the round trip time is continuously less than or equal to a preset minimum round trip time minimum number of times;

The undercut determination unit 422 is configured to determine whether the round trip time is continuously greater than or equal to the preset maximum round trip time maximum number of times, and whether the round trip time of the current time period is greater than the round trip time of the previous time period;

The first switching unit 430 includes:

The upper cut unit 431 is configured to adjust to the high code rate media stream when the round trip time is continuously less than or equal to the preset minimum round trip time minimum number of times.

The undercut unit 432 is configured to adjust the round trip time to be greater than or equal to the preset round trip time maximum value by a predetermined number of times, and adjust the low round rate media stream when the round trip time of the current time period is greater than the round trip time of the previous time period .

The radio network controller embodiment provided by the embodiment of the present invention includes:

The bandwidth judging unit judges whether the bandwidth between the radio network controller and the user equipment decreases; and the buffer unit uses buffer buffer data preset for each user equipment when the judgment result is decreased.

The structure of the packet domain streaming server according to the embodiment of the present invention is as shown in FIG. 5, and includes: a bandwidth determining unit 510, configured to determine, according to a bandwidth change of the user equipment, whether the code rate needs to be adjusted;

a second complete frame determining unit 520, configured to determine whether the current transmission frame is a complete frame;

The second switching unit 530 is configured to: when the code rate needs to be adjusted, and the current transmission frame is a complete frame, The code rate is switched, and the media data after the switching code rate is transmitted.

The bandwidth determining unit 510 includes:

The second round-trip time determining unit 511 is configured to obtain a round-trip time of the sender report packet; when the round-trip time meets the predetermined condition, determine that the code rate needs to be adjusted;

Or, the packet loss rate determining unit 512 is configured to obtain a packet loss rate. When the packet loss rate meets a predetermined condition, it is determined that the code rate needs to be adjusted.

The packet domain streaming server embodiment provided by the embodiment of the present invention includes:

A unit for transmitting media data after switching the code rate.

The unit for performing rate switching according to the change of the bandwidth of the user equipment and determining that the code rate needs to be adjusted is further used to:

Get the round trip time of the sender report package;

Determining whether the round trip time meets a predetermined condition;

When the round-trip time satisfies a predetermined condition, rate switching is performed.

The unit is configured to: when the bandwidth of the user equipment changes, and determine that the code rate needs to be adjusted, the unit for performing rate switching is further configured to:

When the current transmission frame is a complete frame, rate switching is performed. .

For the specific usage of the packet domain streaming server embodiment and the radio network controller embodiment provided by the embodiment of the present invention, reference may be made to the foregoing description of the dynamic rate allocation method embodiment provided by the embodiment of the present invention, which is not repeated here. .

Through the description of the above embodiments, those skilled in the art can clearly understand that the present invention can be implemented by means of software plus a necessary hardware platform, and of course, all can be implemented by hardware. Based on such understanding, all or part of the technical solution of the present invention contributing to the background art may be embodied in the form of a software product, which may be stored in a storage medium such as a ROM/RAM, a magnetic disk, an optical disk, or the like. A number of instructions are included to cause a computer device (which may be a personal computer, server, or network device, etc.) to perform the methods described in various embodiments of the present invention or portions of the embodiments.

The dynamic bit rate allocation method and the packet domain streaming media server provided by the embodiments of the present invention may The timing of the handover is determined according to the round trip time RTT of the SR packet, and how to switch, or I frame is selected for switching at the time of switching, thereby improving the quality of the playback.

The foregoing description of the foregoing embodiments of the dynamic rate allocation method and the packet domain streaming server provided by the present invention is only for helping to understand the method and core idea of the present invention. Meanwhile, for those skilled in the art, The present invention is not limited by the scope of the present invention.

Claims

Rights request

A dynamic rate allocation method, comprising:

The dynamic rate allocation method according to claim 1, wherein, when the bandwidth of the user equipment changes, and it is determined that the code rate needs to be adjusted, performing rate switching includes: obtaining a round trip time of the sender report packet ;

Determining whether the round trip time meets a predetermined condition;

The dynamic bit rate allocation method according to claim 2, wherein the round-trip time for obtaining the sender packet comprises:

The round trip time of the sender report packet is obtained according to the transmission time of the sender report packet carried in the receiver report packet and the delay of receiving the sender report packet to the transmission time.

The dynamic rate allocation method according to claim 2, further comprising: determining that the current transmission frame is a complete frame after determining that the round-trip time satisfies a predetermined condition, and performing rate switching.

The dynamic rate allocation method according to claim 2, 3 or 4, wherein, when the determining that the round-trip time satisfies a predetermined condition, performing rate switching includes:

When it is determined that the round trip time is continuously less than or equal to the preset round trip time minimum number of times reaches a predetermined number of times, the media stream is adjusted to a high code rate.

The dynamic rate allocation method according to claim 2, 3 or 4, wherein, when the determining that the round trip time satisfies a predetermined condition, performing rate switching includes:

And determining that the round trip time is continuously greater than or equal to the preset round trip time maximum number of times reaches a predetermined number of times, and when the round trip time of the current time period is greater than the round trip time of the previous time period, adjusting to the low code rate media stream.

The dynamic rate allocation method according to claim 4, wherein the method further comprises: After the bandwidth between the radio network controller and the user equipment drops, the radio network controller uses the buffer buffer data preset for each user equipment.

8. The dynamic bit rate allocation method according to claim 1, wherein performing rate switching comprises:

When the current transmission frame is a complete frame, rate switching is performed.

The dynamic rate allocation method according to claim 8, wherein the determining the need to adjust the code rate according to the change of the bandwidth of the user equipment comprises:

Obtaining a round trip time of the sender report packet; determining that the code rate needs to be adjusted when the round trip time satisfies the predetermined condition;

Or, the packet loss rate is obtained; when the packet loss rate satisfies the predetermined condition, it is determined that the code rate needs to be adjusted.

A packet domain streaming media server, comprising:

The packet domain streaming server according to claim 10, wherein the round trip time acquiring unit comprises:

a sending time obtaining unit, configured to acquire a sending time of the sender report packet carried in the receiver report packet;

a delay acquiring unit, configured to obtain a delay of a last receiving sender report packet carried in the receiver report packet to a sending time;

And a processing unit, configured to acquire a return time of the sender report packet according to a sending time of the sender report packet carried in the receiver report packet, and a delay of receiving the sender report packet to the sending time.

The packet domain streaming server according to claim 10, further comprising: a first complete frame determining unit, configured to determine whether the current transmission frame is a complete frame, and when the current transmission frame is a complete frame, Controlling the first switching unit to perform rate switching.

The packet domain streaming server according to claim 10, 11 or 12, wherein the first round trip time determining unit comprises:

a top-cut determining unit, configured to determine whether the number of times that the round-trip time is continuously less than or equal to a preset minimum round-trip time reaches a predetermined number of times;

The first switching unit includes:

And an upper cutting unit, configured to adjust to the high code rate media stream when the round trip time is continuously less than or equal to a preset minimum number of round trip times.

The packet domain streaming media server according to claim 10, 11 or 12, wherein the first round trip time determining unit comprises:

The undercut determining unit is configured to determine whether the round trip time is continuously greater than or equal to a preset number of times of the maximum round trip time, and whether the round trip time of the current time period is greater than the return time of the previous time period;

The first switching unit includes:

And an undercut unit, configured to adjust the round trip time to be greater than or equal to a preset round trip time maximum number of times, and adjust the round trip time to a low bit rate media stream when the round trip time of the current time period is greater than the round trip time of the previous time period.

15. A packet domain streaming media server, comprising:

The packet domain streaming server according to claim 15, wherein the bandwidth determining unit comprises:

a second round-trip time determining unit, configured to obtain a round-trip time of the sender report packet; when the round-trip time meets the predetermined condition, determine that the code rate needs to be adjusted;

Or, a packet loss rate determining unit is configured to obtain a packet loss rate; when the packet loss rate meets a predetermined condition, Determine the need to adjust the bit rate.

17. A packet domain streaming media server, comprising:

A unit for transmitting media data after switching the code rate.

The packet domain streaming server according to claim 17, wherein the means for performing rate switching when the bandwidth of the user equipment changes and determining that the code rate needs to be adjusted is further used for:

Get the round trip time of the sender report package;

Determining whether the round trip time meets a predetermined condition;

The packet-domain streaming server according to claim 17, wherein the means for performing rate switching when the bandwidth of the user equipment changes and determining that the code rate needs to be adjusted is further used to: