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WO2004039034A1 - Systeme et procede de reduction du temps de tamponnage initial pour une application en continu - Google Patents

Systeme et procede de reduction du temps de tamponnage initial pour une application en continu Download PDF

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Publication number
WO2004039034A1
WO2004039034A1 PCT/IB2003/004675 IB0304675W WO2004039034A1 WO 2004039034 A1 WO2004039034 A1 WO 2004039034A1 IB 0304675 W IB0304675 W IB 0304675W WO 2004039034 A1 WO2004039034 A1 WO 2004039034A1
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WO
WIPO (PCT)
Prior art keywords
bit rate
data
transport
encoding
level
Prior art date
Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
Ceased
Application number
PCT/IB2003/004675
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English (en)
Inventor
Timus Bogdan
Current Assignee (The listed assignees may be inaccurate. Google has not performed a legal analysis and makes no representation or warranty as to the accuracy of the list.)
Telefonaktiebolaget LM Ericsson AB
Original Assignee
Telefonaktiebolaget LM Ericsson AB
Priority date (The priority date is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the date listed.)
Filing date
Publication date
Application filed by Telefonaktiebolaget LM Ericsson AB filed Critical Telefonaktiebolaget LM Ericsson AB
Priority to AU2003274413A priority Critical patent/AU2003274413A1/en
Publication of WO2004039034A1 publication Critical patent/WO2004039034A1/fr
Anticipated expiration legal-status Critical
Ceased legal-status Critical Current

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Classifications

    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L47/00—Traffic control in data switching networks
    • H04L47/10—Flow control; Congestion control
    • H04L47/30—Flow control; Congestion control in combination with information about buffer occupancy at either end or at transit nodes
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/60—Network streaming of media packets
    • H04L65/75—Media network packet handling
    • H04L65/752—Media network packet handling adapting media to network capabilities
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/60—Network streaming of media packets
    • H04L65/75—Media network packet handling
    • H04L65/762—Media network packet handling at the source 
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/80—Responding to QoS

Definitions

  • the present invention relates to real-time data communications.
  • the present invention is directed to a system and method for reducing the time-length of an initial buffering phase of a streaming application.
  • a streaming application enables an end-user to enjoy the content (audio, video, or combinations) before the content is completely downloaded.
  • a conversational application such as talking on the phone
  • the coherence of the information is important. Therefore the absolute delay between the transmission moment and the playback moment can be increased in order to preserve the relative time-relation within the information content. This may be necessary if data must be retransmitted, or if the propagation time of the different data packets is not the same, as may happen in a packet-switched network.
  • delay jitter The variation in the propagation time is usually called delay jitter.
  • Playback starts when a sufficient amount of data is contained in the playback buffer. If the transport bit rate decreases below the playback bit rate due to delays in the transport network, the amount of bits in the playback buffer decreases. When the delayed data arrives, the playback buffer is filled again.
  • a problem with downloading a streaming application is that the end-user has to wait for a considerable time while the content is being buffered before the stream can be played back. Unlike a completely downloaded file (which can be immediately utilized through an ftp application), a streaming application is utilized by the end-user (audio, video, or both) before the content is completely downloaded.
  • the target level is selected as a level that provides a predefined amount of playback time (for example, 1 second) if no additional data is delivered.
  • the absolute delay is not important once the playback is started. However, the absolute delay is important in the initial buffering phase since it influences the time elapsed from the moment the end user requests access to a stream until the playback starts. A delay in the range of one to a few seconds is not normally objectionable, but longer delays may be perceived by the end-user as a lack of response from the application, i.e., as a lack of access to a desired service.
  • the time elapsed from the moment the end user requests the stream until the playback starts can be roughly modeled by three components:
  • the first term represents the time needed for the client's request message to reach the streaming server; the second term is the time needed for the data from the server to reach the client; and the last term is the time needed to fill the playback buffer with the initially required amount of data.
  • the first two terms will be ignored in the following description because they do not depend on the application and are negligible when compared to the third term.
  • Minimizing T ln , t is an object of the present invention.
  • the following explanation concentrates on the minimization of T, n ⁇ t by minimizing the initial buffering time since T, n ⁇ t ⁇ Tiering-
  • the buffering time depends on the amount of data to be buffered and on the transport bit rate used in the buffering phase.
  • the optimum size of the playback buffer depends on the delay variation, i.e., on the size of the delay jitter.
  • a straight forward way to define the buffer size is to indicate "how much play-time the buffer should contain, i.e., for how long a time should the application play back if no more data is received from the transport network. This time period may be designated by T p iayback.
  • This design parameter depends on the design of the streaming client (for example, in some RealPlay versions there is a default value that the user may modify).
  • B denotes the bit rate with which the application encodes data. If the encoder is not a constant bit rate encoder, the quantity will vary in time B(t), most often around a target value B.
  • the application may also employ several encoding modes, which means that it may switch between several bit rates ⁇ B ⁇ , B 2 , ... B n ⁇ in case of constant bit rate encoders, or bit rate targets ⁇ B , B 2 , ...B garbage ⁇ in case of variable bit rate encoders.
  • the transport bit rate, Rtransport depends on characteristics of the underlying transport network. When the network is affected by delays or retransmissions due to errors, the transport bit rate varies in time and the equation (3) is valid in some statistical sense.
  • R t r a n s p o r t can be at times very high, as is the case when a large amount of data previously buffered in a routing node is delivered at once.
  • the transport network may have some inherent limitations on this transport bit rate (i.e., R t r a n s port ⁇ Rmax) as is the case when the last link is a slow modem or a radio connection, i.e., when the over provisioning typical to a modern LAN is not available.
  • FIG. 1 is a graph of the number of buffered bits as a function of time when a constant bit rate encoder is utilized in a known buffering procedure.
  • the initial delay (T in i t ) 10 is a function of the bit rate 12 at which the buffer is filled, and the target level (L ta rget) 14 that must be reached before playback begins.
  • the rate at which the buffer is filled is a function of the transport bit rate from the application server to the end-user's buffer.
  • the transport bit rate 12 is shown as 64 kbps.
  • the server could send the data encoded at B with a higher bit rate R t r ans p ort > B. This difference between the transport bit rate and the encoding rate is typically exploited for reducing the initial buffering time.
  • FIG. 2 is an illustrative drawing of a known procedure for reducing the initial time delay, T in i t - In this procedure, assuming that data is available at the streaming server, Tini is reduced by speeding up the transport bit rate.
  • an additional bearer connection 20 is established in addition to the normal connection 22 in order to effectively double the transport bit rate from 64 kbps to 128 kbps. This, of course, reduces the initial delay period by one-half.
  • FIG. 3 is a graph of cumulated data as a function of time, which more clearly illustrates the effect achieved by the procedure of FIG. 2.
  • a constant rate encoder for example 64 kbps
  • 64 kbps is used to illustrate this approach.
  • a known solution for reducing the buffering time is to temporarily provide an additional connection during the buffering period.
  • 3GPP 3 rd Generation Partnership Project
  • a "temporary bearer" is used for sending data at higher bit rates during the start-up phase.
  • this approach has disadvantages because additional connections are not always available.
  • the additional bearer 20 in FIG. 2 might not be provided due to a lack of radio channels.
  • Another disadvantage of the known approach is an increased need for signaling and a rather complicated process of establishing and removing the additional connection. There is no straightforward solution according to the 3GPP standard that avoids the disadvantages raised by the prior art.
  • the present invention trades off quality of the application at the beginning of the session for a reduced initial delay period.
  • the application may be presented initially to the end-user with lower quality, but the delay period is reduced.
  • the delay period is reduced because fewer bits need to be buffered in order to provide the predefined amount of playback time.
  • the buffer continues to fill. Once the buffer reaches a target level that provides the predefined amount of playback time at a normal (higher) quality level, the playback quality can be increased to a normal level.
  • the present invention is directed to a method of reducing the time length of an initial buffering phase of a streaming application in a communications network having a streaming server with an encoder, a streaming client with a playback buffer, and a transport network.
  • the method includes the steps of determining a transport bit rate for the transport network; encoding data with the encoder at an encoding bit rate that is lower than the transport bit rate; and transporting the encoded data through the transport network from the streaming server to the playback buffer in the streaming client at the transport bit rate.
  • the method also includes beginning playback of the data when the level of data in the playback buffer reaches a first predefined threshold level; and increasing the encoding bit rate to equal the transport bit rate when the level of data in the playback buffer reaches a second predefined threshold level that is higher than the first predefined threshold level.
  • the present invention is directed to an alternative method of reducing the time length of an initial buffering phase of a streaming application.
  • the method includes the steps of determining a transport bit rate in the transport network between the streaming server and the playback buffer in the streaming client; encoding data with the encoder during an initial buffering period at a first encoding bit rate that is lower than the determined transport bit rate; and transporting the encoded data through the transport network from the streaming server to the playback buffer in the streaming client at the transport bit rate.
  • the method also includes beginning playback of the data at a first quality level determined by the first encoding bit rate when the level of data in the playback buffer reaches a first predefined threshold level; increasing the first encoding bit rate to a second encoding bit rate when the level of data in the playback buffer reaches a second predefined threshold level that is higher than the first predefined threshold level; and playing back the data at a second quality level determined by the second encoding bit rate when the level of data in the playback buffer reaches the second predefined threshold level.
  • the present invention is directed to a method of reducing the time length of an initial buffering phase of a streaming application in a communications network having a streaming server with a variable rate encoder, a streaming client with a playback buffer, and a transport network.
  • the method includes the steps of determining a transport bit rate in the transport network for transporting a plurality of frames of data between the streaming server and the playback buffer in the streaming client; encoding data with the variable rate encoder during an initial buffering period at a first average encoding bit rate that is lower than the determined transport bit rate; and transporting the encoded data through the transport network from the streaming server to the playback buffer in the streaming client at the transport bit rate.
  • the method also includes beginning playback of the data at a first quality level determined by the first average encoding bit rate when the level of data in the playback buffer reaches a predefined threshold level; and varying the instantaneous encoding bit rate for each data frame to make the plurality of data frames more equal in size.
  • the present invention is directed to a system in a communications network for reducing the time length of an initial buffering phase of a streaming application.
  • the system first includes a streaming server comprising a multi-rate encoder that encodes data at an encoding rate selected from a plurality of encoding rates; a switching mechanism for selecting the encoding rate; and server logic that controls the switching mechanism.
  • the system includes a streaming client comprising a playback buffer that stores data received from the server; and client logic that reports buffer parameters to the server.
  • the system includes a transport network for transporting data at a transport bit rate from the server to the client.
  • the multi- rate encoder initially encodes the data at an encoding bit rate that is lower than the transport bit rate.
  • the playback buffer begins playing back the data at a first quality level determined by the first encoding bit rate when the level of data in the playback buffer reaches a first predefined threshold level.
  • the client logic reports to the server logic when the level of data in the playback buffer reaches a second predefined threshold level that is higher than the first predefined threshold level.
  • the server logic controls the switching mechanism to increase the first encoding bit rate to a second encoding bit rate when the level of data in the playback buffer reaches the second predefined threshold level.
  • the playback buffer plays back the data at a second quality level determined by the second encoding bit rate after the level of data in the playback buffer reaches the second predefined threshold level.
  • FIG. 1 is a graph of the number of buffered bits as a function of time when a constant bit rate encoder is utilized in a known buffering procedure;
  • FIG. 2 (Prior Art) is an illustrative drawing of a known procedure for reducing the initial time delay
  • FIG. 3 (Prior Art) is a graph of cumulated data as a function of time, which more clearly illustrates the effect achieved by the procedure of FIG. 2;
  • FIG. 4 is an illustrative drawing of a procedure for reducing the initial time delay in accordance with the teachings of the present invention
  • FIG. 5 is a graph of the number of buffered bits as a function of time when utilizing a constant bit rate encoder with the procedure of FIG. 4;
  • FIG. 6 is a simplified block diagram of the preferred embodiment of the system of the present invention.
  • FIG. 4 is an illustrative drawing of a procedure for reducing the initial time delay in accordance with the teachings of the present invention.
  • the present invention trades the quality of the stream at the beginning of the session for a reduced buffering time. If the total experienced satisfaction of the end user is seen as a function of the annoyance for waiting for the session to start and the quality of the session once it started, then the proposed solution can be used to increase the total satisfaction by trading off the two terms against each other.
  • the present invention reduces the user's annoyance by reducing the buffering time, at the expense of temporarily reducing the quality of the stream by temporarily reducing the encoding rate.
  • the initial waiting time is reduced from a reference waiting time 40 (the waiting time that would be experienced if the data was encoded at the transport rate) to a new waiting time 42.
  • a reference waiting time 40 the waiting time that would be experienced if the data was encoded at the transport rate
  • a new waiting time 42 the application playback starts.
  • the application playback starts, there is a period of decreased quality 44 due to the lower encoding rate.
  • the playback buffer level reaches the target level, the encoding rate is increased to equal the transport rate, and the streaming quality is consequently increased to a normal level 46.
  • the invention may be utilized for both a constant bit rate encoder and a variable bit rate encoder. In both cases, it is assumed that the target (average) encoding bit rate can be changed.
  • the invention also protects the receiver buffer fullness from overflow and underflow using known techniques. However, the invention may also be applied to cases in which the receiver buffer does not have a practical upper bound (for example when the client has a hard disk).
  • the invention utilizes a multi-rate encoder of a known type.
  • a multi-rate encoder is an encoder that is able to generate encoded data with different average bit rates.
  • the high bit rate generally leads to a higher perceived quality, as perceived by the end-user.
  • a lower quality will be perceived when the encoder uses a lower bit rate mode.
  • the lower quality in this instance may not be as low as that experienced by the end-user when the transmission is affected by errors, or when some of the data is lost in the network. If there are long delays, or data is lost in the network, the quality perceived by the end-user may be dramatically degraded.
  • the lower quality level proposed by the present invention is generally better than the low quality that results from transmission errors and ⁇ or losses, even when a higher bit rate mode is utilized.
  • the multi-rate encoder also provides a way to gracefully degrade the quality of the stream when network conditions dictate.
  • an encoding mode with a high bit rate can be used.
  • the encoder can be switched to a lower bit rate that is fitted to the reduced transport bit rate. By doing so, the end user quality is gracefully degraded in a controlled manner.
  • the present invention may be utilized with both constant rate encoders and variable rate encoders.
  • the main difference between a constant rate encoder and a variable rate encoder is the amount of bits used to encode each application frame.
  • a constant rate encoder uses the same amount of bits for encoding each frame regardless of the content of the frame.
  • a variable rate encoder may use different amounts of bits for each frame, depending on the content of the frame.
  • the multi-rate encoder discussed above may be implemented as a constant rate encoder or a variable rate encoder. If the multi-rate encoder is a constant rate encoder, it is capable of switching between at least two constant, but different, encoding rates. If the multi-rate encoder is a variable rate encoder, it is capable of determining average encoding rates and switching between at least two different average encoding rates.
  • FIG. 5 is a graph of the number of buffered bits as a function of time when utilizing a constant bit rate encoder with the procedure of FIG. 4.
  • Data is buffered for Tinit i.e. until L bits have been buffered. Tini t in this case is equal to the new waiting time 42 in FIG. 4.
  • the period of decreased quality 44 comprises the period of time in which frames 1-4 are played back. Thereafter, frame 5 and subsequent frames are played back with normal (higher) quality 46.
  • the transport bit rate is assumed in this example to be constant and therefore the buffer fullness varies between two fixed limits 50 and 51. In practice, the transport bit rate will have variations, for example due to RLC retransmissions, which means that the buffer fullness will also have variations.
  • the input parameter T p ⁇ ay back should be selected according to the receiver buffer size.
  • the above pseudo code assumes that the encoder can encode data at any rate B 2 , and therefore it uses T in i t as an input parameter. If this is not the case, and a limited number of encoding rates ⁇ B 2 ⁇ , e[1 ⁇ /] are available, Tini t is selected from a set of N-1 values. It should also be noted that the shorter the input Tinit, the lower the initial quality level due to the lower value of B 2 .
  • FIG. 6 is a simplified block diagram of the preferred embodiment of the system of the present invention.
  • the preferred embodiment includes a streaming server 100, a transport network 200, and a steaming client 300.
  • the streaming server includes a multi-rate encoder 101 , so that it can selectively encode the data utilizing one of two or more encoding rates (for example, 32 kbps or 64 kbps).
  • the encoding rate may be selected, for example, with the help of a switch 102 if a constant bit rate encoder is used.
  • the transport network provides a transport channel 400 whose transport bit rate is fitted to the highest encoding rate of the streaming server.
  • the streaming client 300 has a playback buffer 301 characterized by a playback time. The amount of data to be buffered in the playback buffer depends on the playback time and on the encoding rate.
  • the streaming client also includes logic 302, which controls the playback buffer and provides access to the buffer by a decoder (not shown).
  • the logic also uses a control channel 500 to inform a logic device 103 in the server about parameters such as transport channel bit rate R, initial (target) buffering time Tj n j t , the time period for lower quality, and other parameters controlling playback buffer fullness.
  • the information about the buffering time is apriori known by the server so that the control channel 500 is not needed.
  • the logic device 103 in the streaming server 100 also triggers the switch 102 to switch between different encoding modes in order to minimize the initial buffering rate.
  • the logic device 103 selects the lower encoding mode (rate). After the time period for lower quality, the logic device switches the mode to the higher encoding rate.
  • the exact implementation of the logic device 103 depends of the characteristics of the system. In the case of a constant rate encoder with a simple switch, the implementation is a straightforward algorithm for switching from one constant rate to another. The algorithm has only two input parameters: Tini t controlling the waiting time, and T p ⁇ a y a ck controlling both the buffer fullness level and the time played back at the lower encoding rate.
  • a variable rate encoder may be utilized to provide for more efficient transmission of data. For the same amount of transmitted data, the quality perceived by the end user is better when a variable rate encoder is used than when a constant rate encoder is used.
  • the use of a variable bit-rate encoder on transport channels with limited (upper bounded) bit rates may cause problems in the client, because use of the variable bit-rate encoder may lead to an overfilled or underfilled condition in the receiver buffer.
  • the receiver buffer may overfill when the client decoder extracts fewer bits from the receiver buffer than the amount of data entering the receiver buffer.
  • the receiver buffer may underfill when the decoder attempts to consume more bits than are available in the receiver buffer.
  • This embodiment of the present invention utilizes a procedure similar to the constant bit rate encoder case to achieve similar results for the variable bit rate encoder case. However, instead of switching between two constant encoding rates (and handling frames of constant sizes), this embodiment switches between two average encoding bit rates. Since a variable bit rate encoder is being utilized, the frame sizes and encoding rates may vary. Because of this, if the initial buffering time is reduced, several adverse consequences are possible. A first possible consequence would be that a number of frames with relatively few bits of data are received, and the target buffer fullness decreases, making the client application more sensitive to delay jitter. Additionally, receiving frames with relatively few bits of data may cause the receiver buffer to run the risk of underflowing.
  • the invention may avoid this risk by varying the instantaneous encoding bit rate for each frame so that the frame sizes become more equal, which is done at the expense of slightly decreased quality over the entire video window.

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  • Engineering & Computer Science (AREA)
  • Multimedia (AREA)
  • Computer Networks & Wireless Communication (AREA)
  • Signal Processing (AREA)
  • Two-Way Televisions, Distribution Of Moving Picture Or The Like (AREA)

Abstract

La présente invention concerne un procédé et un système qu'on utilise dans un réseau de communications pour réduire la durée d'une phase de tamponnement initial dans une application en continu. Un serveur en continu envoie des données à un débit binaire de transport, à un tampon de reproduction d'un client. Un codeur multidébit code initialement les données à un premier débit binaire de codage qui est inférieur au débit binaire de transport. Le tampon de reproduction commence à reproduire les données avec un niveau de qualité inférieur déterminé par le premier débit binaire de codage lorsque le niveau du tampon de reproduction atteint un premier niveau seuil. Lorsque le niveau du tampon atteint un deuxième niveau seuil supérieur, le débit binaire de codage est augmenté et passe à un deuxième débit binaire de codage qui est de préférence égal au débit binaire de transport. Au bout du compte, le tampon de reproduction reproduit les données avec un niveau de qualité supérieur déterminé par le deuxième débit binaire de codage après que le niveau du tampon ait atteint le deuxième niveau seuil.
PCT/IB2003/004675 2002-10-24 2003-10-22 Systeme et procede de reduction du temps de tamponnage initial pour une application en continu Ceased WO2004039034A1 (fr)

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AU2003274413A AU2003274413A1 (en) 2002-10-24 2003-10-22 System and method for reducing initial buffering time for a streaming application

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US42091102P 2002-10-24 2002-10-24
US60/420,911 2002-10-24

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EP2536075A1 (fr) * 2011-06-14 2012-12-19 France Telecom Procédé de demande d'accès par un terminal à un contenu numérique apte à être téléchargé depuis un réseau.
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