| M. Garrett and M. Vetterli. Joint Source#Channel Coding of Statistically Multiplexed Real-Time Services on Packet Networks. IEEE#ACM Transactions on Networking, 1#1#:71#81, February 1993. |
....applications. 2.1 Data partitioning and layered coding Scalable and joint source channel coding important ways for providing real time transport. Very early evidence of combined source channel coding technique exists in [24] which discusses its usage for images. More recently, works presented in [7,18,31] discussed the usage of such techniques for real time services. Particular emphasis is given to scalable coding for VBR video over ATM networks in [8,11] Data loss is a major problem for transmitting video over wireless networks. Bit error rates of around 10 Gamma3 to 10 Gamma2 may arise ....
M.W. Garrett and M. Vetterli. Joint source/channel coding of statistically multiplexed real-time services on packet networks. IEEE/ACM Transactions on Networking, 1(1):71--80, February 1993.
....the fact that voice and video are inherently variable rate sources [7, 33] The potential gain in efficiency is significant. Studies show 20 30 of the time in a voice conversation consists of silence [7] and the peak to mean ratio of variable rate video is typically in the range of 1.5 to 4. 5 [20]. Accommodation of continuous media streams in packet switched networks represents a tremendous technical challenge. Distribution of continuous media streams is sensitive to network delays and variations in those delays, and these real time requirements contrast with those for the applications ....
....are given preference in the network. Specifically, in networks such as ATM networks, the channel carrying the most significant components can be given higher priority in the network when congestion forces packet discard. In the context of ATM networks, a number of techniques have been proposed [20, 33, 55, 63] to split signals into separate channels and use priority channels in controlling quality degradation due to network loss. For example, one study on voice [58] has investigated a least significant bit LSB dropping scheme. Each set of B samples is divided into two segments with the six significant ....
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M.W. Garrett and M. Vetterli. Joint Source/Channel Coding of Statistically Multiplexed Real-Time Services on Packet Networks. IEEE/ACM Transactions on Networking, 1(1):71--81, February 1993.
....of i. This choice was dictated by our experience with audioconferences over the MBone. 17 Our algorithm is a very unsophisticated version of a joint source channel coding algorithm. The use of such algorithms has been advocated in both the networking and the signal processing communities (e.g. [12]) and we are investigating better designed and more eOEcient algorithms. However, the algorithm above is useful because it is very simple to implement and it is computationally cheap. Furthermore, it serves as a baseline to evaluate future algorithms. Figure 11 shows measurements obtained ....
M.W. Garrett, M. Vetterli, iJoint source/channel coding of statistically multiplexed real time services on packet networks, ACM/IEEE Trans. Networking, vol. 1, no. 1, pp. 71-80, Feb. 1993.
....and channel coding means wasting resources, because the channel coder cannot exploit the fact that not all of the stream data is of equal importance. For most lossy video coding algorithms, a combined source and channel coding offers a higher performance than a separated approach [7] 12] [18], 37] Secondly, these traditional video coding methods do not offer the elasticity to adapt to frequent changes of channel conditions. For interactive video transmissions over networks with wireless links, not only a certain minimum bandwidth of correctly transmitted data has to be provided, ....
M. W. Garrett, M. Vetterli, Joint Source/Channel Coding of Statistically Multiplexed Real-time Services on Packet Networks, IEEE/ACM Transactions on Networking, Vol. 1, No. 1, pp. 71-80, February 1993
....(such as that available with WFQ routers and the Packet Pair technique) it is more difficult to decide when to join a new group. In any case, the use of (overlapping or non overlapping) layered coding schemes allows the development and evaluation of efficient joint source channel coding schemes [15]. The goal then is to make sure that the more important bands are received without error or loss at the destinations. One way to do this in the current single class best effort service Internet is to use FEC schemes to provide unequal error protection proportional to the importance of each band ....
M.W. Garrett, M. Vetterli, "Joint source/channel coding of statistically multiplexed real time services on packet networks, ACM/IEEE Trans. Networking, vol. 1, no. 1, pp. 71-80, Feb. 1993.
....provide slightly better audio quality than other TCP like mechanisms we experimented with. Our algorithm is a very unsophisticated version of a joint source channel coding algorithm. The use of such algorithms has been advocated in both the net working and the signal processing communities (e.g. [12]) and we are investigating better designed and more eOEcient algorithms. However, the algorithm above is useful because it is very simple to imple ment and it is computationally cheap 4 . Furthermore, 44 The combinations in Table 4 involve ADM coded redun dant information precisely so as to ....
M.W. Garrett, M. Vetterli, iJoint source/channel coding of statistically multiplexed real time services on packet networks, ACM/IEEE Trans. Networking, vol. 1, no. 1, pp. 71-80, Feb. 1993.
....The priorities are used by network switches to selectively discard low priority packets which carry information that is less crucial in reconstructing the voice signal. Channel coding techniques have been shown to provide a graceful degradation of playback quality in a variety of loss scenarios [12, 21, 26]. For PCM encoded voice, packet loss rates of over 5 on the channel carrying the least significant information have been reported as tolerable [22] A drawback to channel coding, however, is that the network is required to support selective dropping of packets during periods of congestion, a ....
M.W. Garrett and M. Vetterli. Joint Source/Channel Coding of Statistically Multiplexed Real-Time Services on Packet Networks. IEEE/ACM Transactions on Networking, 1(1):71-- 81, February 1993.
....information with a guaranteed performance (in particular guaranteed maximal loss rate) service, or that these packets be iprotectedj against loss. This can be done using a variety of so called unequal error protecting schemes, which have been the subject of much research eoeort recently (e.g. [16, 10]. Our goal was not to develop or experiment such schemes, but instead to experiment with rate control schemes. Thus, we have focused on simple coders with low cpu requirements and no need for unequal error protection schemes. We describe next a simple layered encoding scheme in which all layers ....
M.W. Garrett, M. Vetterli, iJoint source/channel coding of statistically multiplexed real time services on packet networksj, ACM/IEEE Trans. Networking, vol. 1, no. 1, pp. 71-80, Feb. 1993.
.... design of transport protocols was rst de ned in Clark and Tennenhouse s Application Level Framing (ALF) protocol architecture [5] whereas the idea that coders for this application need to be designed based on Joint Source Channel Coding (JSCC) principles was rst proposed by Garrett and Vetterli [11]. McCanne in his thesis developed the idea that ALF and JSCC are in fact parallel concepts [20] In recent independent work [26] Puri et al. put together the error resilience techniques of [21] with the transport protocol of [17] That work di ers from ours however in that, although the protocol ....
M. W. Garrett and M. Vetterli. Joint Source/Channel Coding of Statistically Multiplexed Real-Time Services over Packet Networks. IEEE/ACM Trans. Networking, 1(1):71-80, 1993.
....and problems associated with TCP friendly schemes, and discussed earlier, remain. Specific mechanisms based on the above scheme are described in [38, 37, 34, 41] In any case, the use of layered coding schemes allows the development and evaluation of efficient joint source channel coding schemes [11, 21]. The goal then is to make sure that the more important bands are received without error or loss at the destinations. One way to do this in the current single class best effort service Internet is to use FEC schemes to provide unequal error protection proportional to the importance of each band ....
M.W. Garrett, M. Vetterli, "Joint source/channel coding of statistically multiplexed real time services on packet networks, ACM/IEEE Trans. Networking, vol. 1, no. 1, pp. 71-80, Feb. 1993.
....in order to encode and decode the redundant information with sufficient speed. Finally, FEC does not guarantee that corrupted or lost packets can always be recovered. Channel coding refers to a class of approaches that separate the voice signal into multiple data streams with different priorities [11, 26, 27]. These priorities are then used to tag voice packets so that during periods of congestion the network is more likely to discard low priority packets which carry information that is less crucial in reconstructing the original signal. By controlling the cells that the network discards, these ....
M.W. Garrett and M. Vetterli. Joint Source/Channel Coding of Statistically Multiplexed Real-Time Services on Packet Networks. IEEE/ACM Transactions on Networking, 1(1):71-- 81, February 1993.
....Mosaic. The sampling rate (8000 Hz) and the sample size (8 bits) are consistent with default Sun workstation settings. We do not attempt to cite the massive literature on various aspects of the problem that our work relates to. Some notable examples in the area of voice are [YH93, BM 88, HL86, GV86, SL89, DS89, Goo90] Most pieces of work partition the buffer into various zones. When the buffer level is in a certain zone, a specific codebook is used by the sources. Typically, at lower buffer levels codebooks that cause low distortion are used. As buffer levels increase, lower rate codebooks ....
Mark W. Garrett and Martin Vetterli. Joint source/channel coding of statistically multiplexed real time services on packet networks. IEEE/ACM Transactions on Networking, 4(6):856--868, September 1986.
....of redundancy. The Priority Encoding Transmission (PET) scheme [1] 9] allows the user to set different levels of error protection for different portions of the MPEG stream, but unlike FLIIT provides no explicit mechanism for allocating these levels. Layered transmission schemes such as [7][11] incorporate similar ideas, but require networks which treat packets differently according to their priorities. Our FLIIT scheme can function on any network supporting a simple datagram protocol. Similar techniques of joint source channel coding for continuous bitstreams have been developed in ....
M. W. Garrett and M. Vetterli. Joint source/channel coding of statistically multiplexed real-time services on packet networks. IEEE Transactions on Networking, 1(1):71--80, Feb. 1993.
....The priorities are used by network switches to selectively discard low priority packets which carry information that is less crucial in reconstructing the voice signal. Channel coding techniques have been shown to provide a graceful degradation of playback quality in a variety of loss scenarios [10, 20, 23]. For PCM encoded voice, packet loss rates of over 5 on the channel carrying the least significant information have been reported as tolerable when using small (32 byte) packets [21] A drawback to channel coding, however, is that the network is required to support selective discarding of packets ....
M. W. Garrett and M. Vetterli. Joint Source/Channel Coding of Statistically Multiplexed Real-Time Services on Packet Networks. IEEE/ACM Transactions on Networking, 1(1):71-- 81, February 1993.
....packet dropping in the network [5] Karlsson and Vetterli [100, 101] later described the integration of packet video into a general network architecture. They were the first to suggest the use of layered sourcecoding for video combined with prioritized packet discard to carry out loss recovery. In [63, 64], Garrett and Vetterli demonstrated the benefits of using network feedback to control the coding process. They were the first to describe their approach as a form of joint source and channel coding (JSCC) borrowing this terminology from traditional communication theory. In their scheme, the ....
....concept appeared long ago in the context of communications theory. Joint source channel coding states that one can often achieve better system performance by combining the design of compression and error control coding rather than treating these sub problems independently. Garrett and Vetterli [63, 64] first applied this terminology to packet networks by viewing the packet transport mechanism as the channel coding algorithm. In his thesis, Garrett argues that real time traffic is better served by jointly designing the compression algorithm with the transmission system [66] We believe that the ....
Mark W. Garrett and Martin Vetterli. Joint source/channel coding of statistically multiplexed real-time services on packet networks. IEEE/ACM Transactions on Networking, 1(1):71--80, February 1993.
....due to network packet loss, the better the quality of the network service. The sensitivity of the video stream to lost packets depends on the encoding used. Proposed mechanisms for the recovery of packet loss in delay sensitive streams include forward error correction [1, 6] channel coding [5], or retransmission [3] Synchronization of the video source with its multiple receivers is needed in order to ensure continuous playback of the video frames at the receiving endsystems. Factors contributing to the loss of synchronization between the video producer and consumer include ....
M. Garrett and M. Vetterli. Joint Source/Channel Coding of Statistically Multiplexed Real-Time Services on Packet Networks. IEEE/ACM Transactions on Networking, 1(1):71--81, February 1993.
....section is well suited for experimentation with new signal compression schemes. This is especially important since video compression algorithms have been traditionally designed for constant bit rate channels and new approaches are required for transmission over heterogeneous packet networks [18, 41] like the Internet. In its current form, the Internet is a relatively harsh environment for compressed video signals. Packet loss rates are often significant and loss patterns are bursty [4] Compression schemes that rely on low bit error rate or on channel coding techniques to effectively reduce ....
GARRETT, M. W., AND VETTERLI, M. Joint source/channel coding of statistically multiplexed real-time services on packet networks. IEEE/ACM Transactions on Networking 1, 1 (Feb. 1993), 71--80.
....The priorities are used by network switches to selectively discard low priority packets which carry information that is less crucial in reconstructing the voice signal. Channel coding techniques have been shown to provide a graceful degradation of playback quality in a variety of loss scenarios [10, 20, 23]. For PCM encoded voice, packet loss rates of over 5 on the channel carrying the least significant information have been reported as tolerable when using small (32 byte) packets [21] A drawback to channel coding, however, is that the network is required to support selective discarding of packets ....
M.W. Garrett and M. Vetterli. Joint Source/Channel Coding of Statistically Multiplexed Real-Time Services on Packet Networks. IEEE/ACM Transactions on Networking, 1(1):71-- 81, February 1993.
....of redundancy. The Priority Encoding Transmission (PET) scheme [1] 13] allows the user to set different levels of error protection for different portions of the MPEG stream, but unlike this paper provides no explicit mechanism for allocating these levels. Layered transmission schemes such as [8][15] incorporate similar ideas, but require networks which treat packets differently according to their priorities. Our allocation scheme can be used with any network supporting a simple datagram protocol. Our contribution is to provide a simple, low complexity mechanism for obtaining an optimized ....
M. W. Garrett and M. Vetterli. Joint source/channel coding of statistically multiplexed real-time services on packet networks. IEEE Transactions on Networking, 1(1):71--80, Feb. 1993.
....then obtained by modulating the size of the compressed representation. The scheme is depicted in Fig. 4 where R 3 R 2 R 1 . This is in contrast to traditional packetization of layered video where the inter dependency and varying importance of the packets often require transport prioritization [11, 9]. Each component contains coefficient blocks which are independently compressed using progressive quantization followed by hierarchical block coding [8] described in more details in Section 2.2. When subband coefficients are represented using signed magnitude format, progressive quantization ....
M.Garrett and M.Vetterli. Joint Source/Channel Coding of Statistically Multiplexed RealTime Services on Packet Networks. IEEE/ACM Trans. on Networking, Vol. 1, No. 1, pp 71-80, February 1993.
....encoder to improve the overall system performance. This last observation is yet another indicator of something we have argued about extensively in previous work: the fact that the problem of providing real time services over packet networks naturally calls for joint source channel coding methods [3, 8, 14]. Indeed, even disregarding practical issues related to the complexity of encoders and decoders (as argued above) real time communication over packet networks is a prime example of a situation in which the assumptions of the Source Channel Separation Theorem [1, Ch. 8] are not satis ed: that ....
M. W. Garrett and M. Vetterli. Joint Source/Channel Coding of Statistically Multiplexed Real-Time Services over Packet Networks. IEEE/ACM Trans. Networking, 1(1):71-80, 1993.
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M. Garrett and M. Vetterli. Joint Source#Channel Coding of Statistically Multiplexed Real-Time Services on Packet Networks. IEEE#ACM Transactions on Networking, 1#1#:71#81, February 1993.
No context found.
M. W. Garrett and M. Vetterli. Joint source/channel coding of statistically multiplexed real time services on packet networks. IEEE/ACM Trans. on Networking, 1(1):71--80, February 1993.
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