Abstract:
The present invention relates to a method and apparatus for streaming media to a plurality of adaptive client devices. In one aspect there is provided a method of providing a media stream over data channel of a best effort transmission network that includes a wireless path to a plurality of client devices. In another aspect there is provided a method of encoding a stream of data into chunks, whereby the chunks are obtained by determining a break point between them that corresponds to a silence point. In another aspect, there is provided a method for creating a library of encoded media for a media stream and linking the library to a plurality of cell phone devices.

Description:
CLAIM OF PRIORITY 
       [0001]    This application claims priority from U.S. Provisional Application No. 60/797,486 titled “An End-To-End System That Delivers Full Version Of Long Form Content To Small Screen Terminals By Combining Text, Image And Streaming Media, And Employing A Client Only Adaptive Bit Rate Adjustment Mechanism Based On Multi-Rate Chunking To Ensure Uninterrupted Streaming In Real-Time In The Face Of Fluctuating Bandwidth Available To The Streaming Session” and filed on May 5, 2006, the contents of which are expressly incorporated by reference herein. 
     
    
     FIELD OF THE INVENTION 
       [0002]    The present invention relates to a method and apparatus for streaming media to a plurality of adaptive client devices. 
       BACKGROUND OF THE INVENTION 
       [0003]    Wireless networks are well known in which many thousands of client wireless devices share the same network bandwidth. Cellular phones are one such example. 
         [0004]    In a typical network, wireless network bandwidth fluctuates for a variety of reasons, including channel sharing between different client wireless devices, as well as changing conditions, environmental or otherwise, between the client wireless devices and a base station with which they communicate, as well as changes between the base station and other [servers] that are accessed for purposes of obtaining content. 
         [0005]    Bandwidth sharing among applications on the same client wireless device, if instituted, is another reason for network bandwidth fluctuations, since bandwidth available to each application can also fluctuate. 
         [0006]    One known method of transmission with a client device is to use constant bit rate streaming. Users experience poor streaming media quality when available bandwidth is lower than the streaming bit rate, as undesired gaps in the stream occur, which thus cause gaps in the audio or other content being experienced. 
         [0007]    In order to overcome the disadvantages of constant bit rate streaming, it is also known to use a streaming server that can adapt its streaming bit rate dynamically. While such dynamic adaptation has advantages, such an approach does not scale well if the streaming server streams a large number of streams, such as tens of thousands. This is because the streaming server needs to fully understand the syntax of the transmitted media bit stream and process the adaptive bit rate request in a sophisticated manner that requires intensive computation processing such as time synchronization, and header seeking within the media bit stream. 
       SUMMARY OF THE INVENTION 
       [0008]    The present invention relates to a method and apparatus for streaming media to a plurality of adaptive client devices. 
         [0009]    In one aspect there is provided a method of providing a media stream over data channel of a best effort transmission network that includes a wireless path to a plurality of client devices. 
         [0010]    In another aspect there is provided a method of encoding a stream of data into chunks, whereby the chunks are obtained by determining a break point between them that corresponds to a silence point. 
         [0011]    In another aspect, there is provided a method for creating a library of encoded media for a media stream and linking the library to a plurality of cell phone devices. 
     
    
     
       BRIEF DESCRIPTION OF THE DRAWINGS 
         [0012]    These and other aspects and features of the present invention will become apparent to those of ordinary skill in the art upon review of the following description of specific embodiments of the invention in conjunction with the accompanying figures, wherein: 
           [0013]      FIGS. 1(   a ), ( b ) and ( c ) together illustrate communications in the wireless network according to the present invention; 
           [0014]      FIG. 2  illustrates a buffer within a client device according to the present invention; 
           [0015]      FIG. 3(   a ) illustrates a representation of an audio sequence, and various break points within the sequence that are used to establish raw chunks according to the present invention and  FIG. 3(   b ) illustrates adaptive encoding for generating from raw chunks encoded chunks for multiple cellphone classes according to the present invention; 
           [0016]      FIG. 4  illustrates a flowchart of an intelligent chunk mechanism for audio according to the present invention. 
           [0017]      FIG. 5  illustrates a flowchart of an adaptive transmission control algorithm according to the present invention; 
           [0018]      FIG. 6  illustrates another flowchart of an adaptive transmission control algorithm according to the present invention; 
       
    
    
     DESCRIPTION OF THE PREFERRED EMBODIMENTS 
       [0019]    The present invention provides an end-to-end system that delivers full version of long form content to small screen terminals/client devices in streaming (real-time) fashion and minimize the on-off interruption in the context of fluctuating bandwidth. As described further herein, in a preferred embodiment, the present invention employs an intelligent chunking mechanism that segments a continuous stream of unencoded/uncompressed media data into raw chunks, a multi-rate, multi client encoder that generates a library of streaming media in encoded chunks for an array of different classes of client devices, a client only adaptive adjustment mechanism for various bit rates and compression scheme combinations that is based on multi-client, multi-rate encoding of raw chunks to ensure uninterrupted streaming in real-time in the face of fluctuating bandwidth available to the streaming session and a mechanism to transmit and display on the device screen media such as video, text, flash, or an image, with or without a hyperlink, while audio is being streamed as described herein and played through the device&#39;s audio output device, and particularly when the audio buffer is full or larger than certain threshold, since when audio buffer occupancy is larger than a threshold, it is safer to download other content, without causing the buffer to deplete. 
         [0020]    As described herein, usage of the intelligent chunking mechanism, in combination with the multi-rate, multi client encoder, provides advantages over usage of continuous bit streams that are conventionally used for streamed content These advantages include 1) simplifying the streaming server and the client since each raw chunk (or just “chunk” as used herein) is independently encoded leading to independency between chunks, and more particularly to independency between different files that contain different encoded chunks. As a result, when the streaming server and the client deal with adaptive transmissions, adaptation all is done on the encoded chunk level. There is no need for the streaming server and/or the client to probe into the bit stream and seek for certain header and timing information so as to determine the breaking points in the continuous bitstream at which to switch to another bit stream (encoded at a different bit rate); 2). Another benefit from the chunking approach is the support of multiple compression schemes to accommodate more diverse bit rate change  3 ). Since each encoded chunk is preferably represented as a file, caching of the files in the network save data server bandwidth, whereas a continuous bit stream can not be cached; 4) client device implementation is simplified as the client simply requests the file that corresponds to a chunk encoded at a particular bit rate/compression scheme combination (in contrast to sophisticated streaming protocols such as RTSP). As a result, low-resource cellphones can be supported. 5) natural support for text-based content sources that are converted to audio using a text-to-speech engine as text can be naturally broken into text segments based on sentence stops such as a period sign or a comma sign. Then each text segment is converted to an audio chunk via a text to speech engine; 
         [0021]    In addition to using an intelligent chunk mechanism and a multi-client multi-rate encoding mechanism, usage of a client only approach can greatly simply the streaming server on which content is stored and achieve much better scalability because the server can be stateless. 
         [0022]      FIGS. 1(   a ), ( b ) and ( c ) illustrate the system  100  of the present invention, and the manner of communication between the various elements, particularly the steaming/web server and the client as shown in  FIG. 1(   c ), and that a client device can adaptively retrieve encoded chunks via standard file transferring protocol, e.g., HTTP, greatly simplifying the streaming server. 
         [0023]    The system  100  includes an intelligent chunk mechanism  110 , a multi-rate, multi client encoder  112 , which provides multi-rate, multi-client encoded content, described further hereinafter, to a streaming server  120 . The streaming server  120  serves the content to each of a plurality of client devices  130 -( 1 ,  2 ,  3 , n), preferably wireless client devices, through a transmission network  140 . 
         [0024]    A representative portion of the transmission network  140  is illustrated in  FIG. 1(   b ), and may include a web server data center  142 , load balancers  144  (which load balancers can take the form of either cache servers distributed around the network, or a virtual server having an IP address and port to the client devices, which virtual server is bound to a number of physical servers that provide the redundant or different services), other network elements  146  (such as routers, and cache servers), and a base station  148  that provides for wireless communications with the client devices  130  within its range. It should be understood, however, that while only a few client devices  130  are shown, the labeling from  130 - 1  to  130 - n  is to show the intended scale of the present invention in which very many, in the hundreds and thousands, of different client devices  130  are capable of operating simultaneously, with an appropriately scaled network  140  that will include many different base stations  148  and other network elements. 
         [0025]    The client devices  130  include conventional hardware such as transmitter and receiver circuits, a processor, a memory, a user interface, and some type or types of content delivery mechanism, such as a speaker or display unit. The processor, among other functions, will execute a program  132  that provides for the functionality of the present invention as described, which program will reside in an application area of the memory, and allow the downloading of content into a buffer, which content is then ultimately provided from the buffer to the content rendering mechanism (an audio player or a video player) so that it can be rendered and thus presented to the user. The buffer of the client device  130  is functionally illustrated in  FIG. 2  as buffer  200 , and contains as inputs the content data from a network input  210 , and outputs the content data a an output  220 , for further rendering as will be described herein. Monitored from the buffer  200  is the number of encoded chunks of content data that are in the buffer at a given time. The usage of this monitored variable will become apparent from the discussion hereinafter. 
         [0026]    The basic mechanism of providing multi-rate content according to the present invention is described in detail below using the example of audio. Other streaming media formats can be treated in a similar manner, and modifications for different formats are also described further hereinafter. 
         [0027]    With respect to this example, a particular audio source file is segmented into raw chunks (or just “chunks”) using the intelligent chunking mechanism  110  illustrated in  FIG. 1 , the length of each chunk depends on a number of factors: the frequency of wireless bandwidth change, average network bandwidth, intrinsic nature of media (minimum contiguous block of information), buffer size at the client, maximum latency requirement. All these parameters can be optimized mathematically for a particular set of scenarios, although other considerations with respect to chunks, and in particular where to begin and end them, are described further hereinafter. A typical length for each chunk, however, is in the range of 5-10 seconds of content for an average network bandwidth of 10 kbps and a maximum latency of a few seconds. If the network average bandwidth is higher or the latency requirement is lower, the chunk duration can be adjusted proportionally. This also allows, with the implementation of intelligent chunking as described further hereinafter, for the chunks to vary within some range of a preferred chunk length, such as 7 second+/−2 or 3 seconds, during which 4 or 6 second interval for this range, selection therein of a preferred segmentation point can occur. 
         [0028]    The intelligent chunking mechanism  110  is preferably adapted to operate upon different types of content, audio, both audio and video, as well as text. 
         [0029]      FIG. 3  illustrates a representation of an audio sequence within an audio file, and two different embodiments that can be used to establish various break points within the sequence, which break points thus define the chunks according to the present invention. 
         [0030]    The first, shown by break points  310 , are each made so that each chunk has the same period of time. While this has ease of implementation aspects with respect the subsequent encoding of the chunks, as well as with respect to the client device  130 , a disadvantage can be that users may sense a brief pause in the middle of a sentence if the content is audio, for example, as the audio player on the client device  130  switches from the finished encoded chunk to the next encoded chunk. Accordingly an intelligent chunk algorithm is used to segment the streaming content data at appropriate break points, which are not typically purely periodic. 
         [0031]      FIG. 4  illustrates a flowchart of an intelligent chunk algorithm  400  according to the present invention, which is directed to an example using audio. Generally, this algorithm is used to have break point correspond to a point of relative “silence” point to minimize any break effect. In particular, the silence point can be defined in a number of ways, one being the point at which the amplitude is less than a certain threshold (for example, 5% of maximum amplitude). Using this, the algorithm, which is implemented as a computer program, will first, in step  410 , start the data content sequence, and then, in step  420 , skip forward a set amount, typically 7 seconds. In step  430 , a window around this skip forward point is viewed to determine if there exists a silence point therein. A wave representation of the audio file can be used when making this determination. If there exists a silence point within the window, then that point is selected as the break point in step  440 . The sequence then goes step  460  to determine if there is an end to the data stream. If so, then the algorithm ends, but if not, then there is a return back to step  420 . If a silence point does not exist within step  430 , then the lowest amplitude within the window, the skip forward point, or some other point within the window is chosen as the break point in step  420 . Thereafter, step  460  follows. 
         [0032]    For a stream that includes both audio and video, either the audio or video can be used to determine the break point. Preferably the audio is used, and the video break point is made the same. But other methodologies can be used, including using the video scene change. 
         [0033]    For a pure video stream without an audio component, a scene-change point, where there is a significant change in the background is used, and can be detected by looking at the difference between two consecutive video frames. One detection mechanism is to use a difference threshold, and if the difference is larger than that threshold, that is referred to as a scene change that is appropriate to use as a break point. 
         [0034]    Another type of content is live audio. Chunking takes place in a manner that is the same as for a large pre-recorded audio file as described above, except that the live content is chunked in real-time. 
         [0035]    Another type of content is a pre-stored large text file. Such a text file is preferably first chunked based upon text breaks, including but not limited to the period sign, commas, as well as more sophisticated divisions (such as not causing a break between a subject and verb that are adjacent to each other). Once chunked in text form, the intelligent chunker will convert each text chunk to an audio chunk using a text-to-speech engine (not shown). 
         [0036]    Real-time text, such as an RSS feed, is handled in the same way as a large pre-stored text file as described above, except that the real-time text is chunked in real time. 
         [0037]    The multi-client multi-rate encoder  112  inputs the chunks that have been obtained by the intelligent chunker  110  described above, and for each of the different type of client devices  130 , taking into account the specifications the client type, the type of compression scheme being supported on the client device  130 , and the type of wireless network that the particular type of client device  130  operates upon, encode each chunk at a plurality of different bit rates/compression scheme combination, with each bite rate/compression scheme combination corresponding to a so-called track. This is shown also in  FIG. 3(   b ) to illustrate adaptive encoding for generating encoded chunks for an example of multiple cellphone classes. For audio streams, coding will vary, from a few kbps, all the way to several hundreds of kbps (for audio), and it depends on a specific compression scheme chosen. For example, if AMR is used, there are only 8 combinations, all using the same compression scheme, at 4.75 kbps, 5.15 kbps, 5.9 kbps, 6.7 kbps, 7.4 kbps, 7.95 kbps, 10.20 kbps, 12.20 kbps, and wideband AMR, there are 9 bit rates with this compression scheme, and if MP3 or AAC is used, still other bit rates can be used with these compression schemes. Thus, for example, certain cellphones will have the capability to use a set of combinations of tracks that have different compression algorithms and bit rates, while the set of combinations used by other cellphones may only rely on the same compression scheme with different bit rates to differentiate. 
         [0038]    The choice of the tracks by the multi-client multi-rate encoder  112 , which correspond to one of the bit rates and a corresponding compression scheme as described above, depends on minimum type of client device, acceptable media quality, preferred/target media quality, network conditions, the compression algorithm chosen, and reasonable differences between two adjacent bit rates. For example, when network bandwidth can fluctuate from a few kbps to 100 kbps, 20 tracks can be generated for a single chunk: the first 8 tracks compressed with AMR and with the rates of 4.75 kbps, 5.15 kbps, 5.9 kbps, 6.7 kbps, 7.4 kbps, 7.95 kbps, 10.20 kbps, 12.20 kbps; the next tracks compressed with W-AMR with the rates of 14.25 kbps, 15.85 kbps, 19.85 kbps, 23.85 kbps, and the next two tracks using AACPlus with the bit rates of 32 kbps and 48 kpbs, and the remaining tracks encoded with MP3 at bit rates between 40 kbps and 100 kbps. 
         [0039]    Since each client device  130  can be of one of many different types as mentioned above, with each type having a different capability: cpu power, memory, compression scheme supported, how fast the client can switch from one player (that plays one encoded chunk) to the next player (that plays the next encoded chunk). The present invention uses these various different cellphone parameters to determine a cellphone profile table (shown in  FIG. 3(   b ) of different types for each different chunk, and then for each different types, the track/bit rate, compression, and the average chunk duration that should be used. For example: cellphone A support only AMR, and cellphone B supports only AAC. The multi-client multi rate encoder  112  generates many sets of audio output, including one encoded with AMR (to be used by A), and another with AAC (to be used by B). 
         [0040]    Also, in use, a separate hint track with bit rate/compression information for all tracks (of different bit rate/compression combinations) is also generated and supplied to the client device  130  at the beginning of a session that tells the client device  130  the different existing tracks that exist for each encoded chunk, and the approximate starttime and endtime for each encoded chunk, based on an chunk index scheme. For example, the hint track will notify the client device that the media content in issue has 12 tracks, and the target combination (of bit rate/compression) for each track. In addition, the hint track can also be used through the download of the media content, so that information on each chunk can be obtained—such as chunk # 2  has 8 tracks (bit rate/compression scheme), starts from 7.5 second and has a length of 8.2 seconds. 
         [0041]    With respect to the hint track, the hint track has the following fields, as shown in Table I below: 
         [0000]    
       
         
               
             
           
               
                 TABLE I 
               
               
                   
               
             
             
               
                 hint track format: 
               
               
                 Number of tracks = 16 
               
               
                 1 = 4750 amr 
               
               
                   (means the first track&#39;s bit rate is 4.75 kbps, the compression 
               
               
                   scheme is amr) 
               
               
                 2 = 5900 amr 
               
               
                 ... 
               
               
                 16 = 48000 aacplus 
               
               
                 Number of encoded chunk = 24 
               
               
                 1 = 7340 text_url image_url video_url 
               
               
                   (7340 means the duration of the first encoded chunk is 7340 milli 
               
               
                   seconds, each encoded chunk&#39;s start and end time can be easily 
               
               
                   derived from this information. Also the encoded chunk&#39;s size is 
               
               
                   encoded chunk duration * current track bit rate text_url 
               
               
                   refers to the url for the text to be displayed when this audio is being 
               
               
                   played, the same for image and video, and they can be null. That 
               
               
                   content will be downloaded when the buffer level is high enough 
               
               
                 2 = 5670 text_url image_url video_url 
               
               
                 ... 
               
               
                 24 = 4765 text_url image_url video_url 
               
               
                   
               
             
          
         
       
     
         [0042]    The client application program  132  on the mobile handset device  130  monitors the downloading speed of recent encoded chunks (in the stream buffer  134  associated with the device  130  and the application  132 ). The downloading speed is averaged over a specified period of time, and can have a widely vary range depending on the wireless network capabilities, in the range of a few kbps to a few mbps. Actions are triggered based on buffer overflow or underflow status, which actions are first generally described below, with a more detailed discussion provided thereafter. 
         [0043]    The client program  132  is the preferable way in which to monitor the network as the method described herein is network agnostic: via monitoring the buffer  200  at the client device  130  side, at the application level, the present invention can tell how good or bad the channel is. It is noted that since each encoded chunk can be represented as a file on the server side, and the streaming server  120  can tell the client device  130  the file size before the transmission (such as in a HTTP protocol, there is a content-length field in the http header, and there is no need for the client device  130  to know the size of each encoded chunk a priori, because the system preferably operates on a need to know basis), this assists in allowing the client device  130  to initiate the request for the appropriate track, based upon the application level monitoring of the buffer  200 . Nonetheless, other ways to monitor the network channel situation on the client device  130  side can be used. For example, one can monitor the signal-to-noise level at the physical layer. Another way is to monitor the packet loss at the logical layer. Another way is to monitor delay and loss at the IP layer. However, all of these schemes are isolated from the application, and as such aren&#39;t the preferred monitoring method. 
         [0044]    If the downloading speed is lower than the content rendering rate (the audio decoder has to take an encoded chunk out of the buffer after it finishes playing the previous encoded chunk, hence the rendering rate has to do with the duration of each encoded chunk, not the track characteristics of each encoded chunk), it will lead to decrease in buffer occupancy. When the buffer occupancy is lower than Bl, the client program  132  initiates a switch to a track of lower resolution (bit rate/compression scheme) (commensurate to measured network speed) by requesting the server to send a different set of files (encoded chunks) that are encoded at that lower resolution track. The tradeoff is lower streaming media quality, but this quality is preferred to streams with a substantial number of lost packets. B l  is decided by the maximum latency required. The larger the latency can be, the larger B l  can be, and the less likely underflow of buffer  200  will happen. 
         [0045]    On the other hand, if network speed is higher than current content rendering rate, and the buffer level is higher than Bt, the client program  132  can initiate an upshift to higher resolution track to increase/restore the streaming media quality. The reason why downshifting use Bl and upshifting use Bt is because we want to be conservative: when the buffer level is higher than Bl, if we upshift right away and if it happens that the bandwidth drop again to a very low resolution track, the buffer may deplete very soon. Hence we want to build up the buffer level to a higher value of Bt to play safe. 
         [0046]    The downloading protocol can be of any, but in particular well suited for HTTP, in which case the streaming server can be standard stateless web server without any modification. The client simply uses the HTTP protocol to request a file, which corresponds to a chunk encoded at a particular track. 
         [0047]    Among other benefits, two distinct benefits stand out: First, eliminated is the requirement to support complicated streaming protocols, such as (RTSP), on the device  132 , which complicated protocols are typically currently available only on high end cellphones. Second, eliminated is the requirement for expensive and complicated and non-scalable streaming servers, as the present invention requires only conventional stateless web/wap servers to serve streaming audio content 
         [0048]    Streaming server  120  illustrated in  FIG. 1  stores all the audio content, which can be precoded or encoded in real time. The transmission network  140 , which may include a web server data center  142 , will deliver the encoded chunks of audio content to devices  130 , such as cellphones or other mobile devices, typically based on file name of the content. A load balancer  144  can be optionally placed in front of web server data center  142  for better response time. 
         [0049]    The program  132  of the streaming client device  130  that requests encoded chunks will now be described with reference to the flowcharts of  FIGS. 5 and 6 , each of these being alternative implementations. It should be understood, however, that there are also other manners in which the buffer or other characteristics can be monitored in order to adjust the bit rate/compression scheme, and still fall within the scope of the present invention. 
         [0050]    The following annotations are used for both the flowcharts of  FIGS. 5 and 6 :
       R a =Network bit rate   R t =media track bit rate   B t =target number of encoded chunks in the buffer (which corresponds to a target amount of time of available content for same sized encoded chunks, and roughly corresponds to the target amount of time for intelligent raw chunks as described herein)   B l =buffer low indicator, also the minimum buffer for start playing when R n &lt;R t      T c =encoded chunk length in seconds, e.g. 5 sec   T d =encoded chunk download time in seconds=T c *R t /R n          
 
         [0057]      FIG. 5  illustrates one embodiment of the invention program  132 . The buffer  200  is filled to B l  in step  510  quickly before rendering using the rendering engine (an audio player or a video player). 
         [0058]    Then, preferably before each subsequent encoded chunk is downloaded, or after some number of encoded chunks are downloaded, the track (with the combination of bit rate and compression type for each track) is decided in step  520  so that the client device  130  can inform the server which encoded chunk to send. The track is determined in the following manner: first, the buffer level/occupancy is viewed. If the buffer level is less than Bl, then the lowest track is used in step  530  until the buffer level is larger than Bl. 
         [0059]    When B&gt;Bl, the current network bandwidth Rn is first estimated in step  540 , based on the network bit rate measured for the previous encoded chunk (encoded chunk size divided by the time it took to download the previous encoded chunk). Then the target track is decided for this current encoded chunk to be downloaded in step  550 . The algorithm is: if B&gt;Bt, then the target track is set in step  560  to a closest track of the estimated network bandwidth, which may be a track that has a different combination that has a better quality (with greater bit rate/different compression scheme). If Bl&lt;B&lt;Bt, then the target track is set in step  570  to one level lower than the estimated network bandwidth. Steps  540 - 570  are then repeated until the end of the media stream, preferably for each encoded chunk, and as shown by the arrows back to step  540 . 
         [0060]    With respect to the flowchart of  FIG. 6 , in the initial start step  600 , the buffer  200  is filled to B t  as quickly as possible. 
         [0061]    Preferably after the encoded chunk is downloaded (or after some interval or some other measure) in step  610  there is checked whether R n &gt;R t . If no, step  620  follows. If yes (ideally R n =2*R t ), then step  630  follows. 
         [0062]    In step  620 , since R n &lt;R t  the program within the mobile device initiates a request for a track at a lower bit rate/compression scheme R t   low  right away, based on the measured R n , so that an R t  is chosen that is lower than R n . Once encoded chunks at this track combination are being received, then step  630  follow (to ensure that buffer  200  will gradually grow to B t ). 
         [0063]    In step  630 , the content within the buffer  200  is begun to get serially read out in sequence for rendering, while downloading of additional encoded chunks into the buffer  200  continues as fast as possible. 
         [0064]    After B t  is reached, step  640  follows, and the download is slowed to a normal rate. Every T c , an encoded chunk is downloaded, so that content is being downloaded into buffer  200  at the same rate it is being output from buffer  200 . 
         [0065]    At each encoded chunk downloaded, preferably R n  is calculated in step  650  and a determination of the network conditions is made, based upon the low threshold B l , as will now be described. 
         [0066]    If conditions are normal, then step  640  repeats. If R n &lt;R t , the amount of data content in the buffer  200  will decrease, and a lower track resolution is likely needed. It is noted, however, that temporary fluctuation are permitted, so that the overall system will not change based on a temporary fluctuation. In order to determine that a fluctuation is significant, however, the present invention tracks some other measure, preferably a low threshold B l , which corresponds to a low threshold amount of content data in the buffer  200 . 
         [0067]    If B l  is reached, step  660  follows and the program  132  at the client device  130  initiates a request for a track of a lower bit rate/compression scheme, R t   low &lt;R n , to ensure that the buffer  200  will gradually grow to B t . Alternatively, if R t &lt;R n , instead of B l  being reached, a track of higher resolution can be requested. 
         [0068]    After step  660 , the buffer  200  should grow from B l  to B t  There are, however two possibilities. Both are based on the detected R n  as shown by step  670 . 
         [0069]    In the first, the buffer  200  grows, and R t &lt;R n . While normally, as discussed above, this would indicate that a change to a higher resolution track, in this instance, a wait period of one encoded chunk occurs, as shown by step  680 , before changing to a higher resolution track to avoid overreaction, since the increase in R n  can be temporary, just like the temporary decreases as noted above. 
         [0070]    In the second possibility, it is determined that R n  still decreases below R t , which would indicate that a change to a lower resolution track. Similarly as describe above, however, in step  690 , a wait period occurs so that a change to a further lower resolution track is not made until B l  is again reached, in order to avoid a continued oscillation of track changes. 
         [0071]    With respect to parameter setting for B t  and B l , the difference B l  between B t  is actually the window in which we observe the network bit rate fluctuation. B l  is set so that with the roundtrip delay during downshifting to a lower resolution track, the buffer will not be depleted: B l &gt;(R t −R n )*T roundtrip . 
         [0072]    B t −B l  is determined by the statistical behavior of track change frequency and range. If the track changes very often, in order of T c , the window should be large to accommodate such frequent change. If the track changes dramatically (e.g., from 20 kbps to 2 kbps), the buffer  200  can deplete quickly, in this case B t  should also be made larger. It should be understood, however, that it&#39;s not the case that the larger the B t , the better, as this will waste network bandwidth (and users will potentially have to pay more) if the user abandons the transmission, or do a backward rewind to peruse content again. Therefore, B t  is preferably decided by how often the track change and how dramatically it changes. 
         [0073]    In the above-described embodiment, if the buffer  200  undesirably depletes, the rendering/play is stopped, waiting for the next encoded chunk to arrive. In normal operation, the buffer  200  needs to be filled to B t  as described above before rendering will being, which also assists in ensuring that the rendering will last if the network condition become bad again. The drawback with such an implementation, however, is that such buffering requires a longer time, leading to bad user experience. In order to minimize such breaks in streaming, in another aspect, current network bandwidth is reviewed, and if the current network bandwidth is higher than the lowest encoding track and the transmission of content data encoded chunks is at that lowest bit, then rending is started right away, even if the level of the buffer  200  is not yet at the normal level B t . 
         [0074]    It should be apparent that other algorithms can be used to determine which track to use for a particular encoded chunk. 
         [0075]    Given the above description, an application level multicast support and unicast caching support feature will now be discussed. 
         [0076]    In one particular implementation of the system shown in  FIG. 1 , each client device  130  will use an HTTP protocol to retrieve files from the streaming server  120  or the data center  142 , as described above. Since content data is divided into encoded chunks, each of which is essentially a file with a URL, the combination of client device  130  initiated adaptation and the use of an HTTP protocol lead to a benefit that media encoded chunks (files) can be automatically buffered at Internet web cache servers within the network  140 . This can lead to significant bandwidth saving and server CPU savings, as the content data will not always need to be extracted from the streaming server  120  or the data center  142 . Thus, if many users are requesting the same content (if at the same time, called multicast, otherwise called unicast) (for example when user device  130 -A is retrieving encoded chunks from a concert show), the encoded chunks it retrieves are cached at a nearby cache server. When another user device  130 -B&#39;s then attempts to retrieve the same content with the same set of encoded chunks, many of the files are served directly from the cache server, eliminating the need to go back to the streaming server  120 , thus reducing server load and server bandwidth. Not all of the files, will be served directly from the cache server, however, since the particular track that was used for user device  130 -A may not be the same track used by the device  130 -B for the same encoded chunk of content data. It is also noted that while timing information is not needed in order for the application on the client device  130  to determine what next encoded audio chunk is needed, timing information is preferably obtained, as shown above by the information provided in the hint track, in order to allow for other media content, such as an image or video, to be rendered at some specific time or duration during the streaming of the audio content. 
         [0077]    Although the present invention has been particularly described with reference to embodiments thereof, it should be readily apparent to those of ordinary skill in the art that various changes, modifications and substitutes are intended within the form and details thereof, without departing from the spirit and scope of the invention. Accordingly, it will be appreciated that in numerous instances some features of the invention will be employed without a corresponding use of other features. Further, those skilled in the art will understand that variations can be made in the number and arrangement of components illustrated in the above figures. It is intended that the scope of the appended claims include such changes and modifications.