A system of distributing video and/or audio information employs digital signal processing to achieve high rates of data compression. The compressed and encoded audio and/or video information is sent over standard telephone, cable or satellite broadcast channels to a receiver specified by a subscriber of the service, preferably in less than real time, for later playback and optional recording on standard audio and/or video tape.

Audio and video transmission and receiving system

A system in which a single VLAN architecture spans multiple VLAN transport protocols and technologies, including a method and system in which multiple different VLANs may be combined in a single enterprise network. Each LAN-switch in the sytem identifies each frame with an identifier, and associates that identifier with particular VLAN identifiers for each type of VLAN technology. When a frame is bridged or routed from a first type of VLAN to a second type of VLAN, the first VLAN encapsulation is removed and the second VLAN encapsulation is added, with appropriate change in the VLAN identifier for the frame or packet. The identifier may also be implicit for the frame, such as when a particular set of sender's MAC addresses are identified with a particular VLAN. Individual VLANs, of whatever architecture, may be added, configured or reconfigured, modified, or deleted, using control tools associated with the multiple VLAN architecture system. Individual ports may be associated with particular VLANs, or may be designated "dynamic" so that frames or packets associated with those ports are associated with particular VLANs in response to source or destination addresses or other information.

Multiple VLAN architecture system

A reliable transport protocol, suitable for routing protocols, that works with unicast transmission and multicast transmission, and an improved routing protocol based thereon. Multicast transmissions can be interspersed with unicast transmissions in situations where some of the receivers have received all of the packets sent to them and others have not. A mechanism to deliver multicast packets quickly to some receivers, even when there are unacknowledged packets pending for other receivers. When a packet is multicast from a sender node to all of its neighbors, the sender puts the packet on a queue for each neighbor and retransmits the packet if an acknowledgement has not been received within a predetermined period of time. If the packet is retransmitted, it is transmitted as a unicast. The invention takes advantage of the fact that the sender already has to maintain state information to determine who has not received its packets. A technique for distributing routing information gathered by a router outside an autonomous system of routers using the same routing protocol. A metric calculated by an identical routing protocol or a different routing protocol may be distributed and used by routers in a different autonomous system. Packets may be labeled with authentication information or with an administrative tag that specifies a method of routing in response to policy considerations.

Reliable transport protocol for internetwork routing

An arrangement (apparatus and method) for detecting jitter caused during transport of digitally-coded information, such as MPEG-encoded data packets, and for correcting time stamp reference values in accordance with the detected jitter. The disclosed arrangement detects program clock reference (PCR) values from an MPEG-encoded transport stream, where each PCR value represents an expected arrival time of a corresponding stream segment. An actual arrival time for the corresponding stream segment is determined in response to detection of the corresponding PCR value and an independent clock signal. The expected arrival time of the stream segment and the actual arrival time are correlated with an accumulation of expected and actual arrival times of previously-received data packet stream segments in order to determine the jitter and the desired expected arrival time of the corresponding stream segment. The jitter is corrected by a combination of adaptive buffering techniques and restamping the PCR value with corrected values coinciding with the actual arrival time of the stream segments. The disclosed arrangement may be implemented in a receiver system, or as part of the network node that minimizes the effects of cell delay variation in an ATM network.

Apparatus and method for correcting jitter in data packets

The invention provides a method and system for switching in networks responsive to message flow patterns. A message “flow” is defined to comprise a set of packets to be transmitted between a particular source and a particular destination. When routers in a network identify a new message flow, they determine the proper processing for packets in that message flow and cache that information for that message flow. Thereafter, when routers in a network identify a packet which is part of that message flow, they process that packet according to the proper processing for packets in that message flow. The proper processing may include a determination of a destination port for routing those packets and a determination of whether access control permits routing those packets to their indicated destination.

Network flow switching and flow data export

A Residual Time Stamp (RTS) technique provides a method and apparatus for recovering the timing signal of a constant bit rate input service signal at the destination node of a synchronous ATM telecommunication network. At the source node, a free-running P-bit counter counts cycles in a common network clock. At the end of every RTS period formed by N service clock cycles, the current count of the P-bit counter, defined as the RTS, is transmitted in the ATM adaptation layer. Since the absolute number of network clock cycles likely to fall within an RTS period will fall within a range determined by N, the frequencies of the network and service clocks, and the tolerance of the service clock, P is chosen so that the 2P possible counts, rather than representing the absolute number of network clock cycles an RTS period, provide sufficient information for unambiguously representing the number of network clock cycles within that predetermined range. At the destination node, a pulse signal is derived in which the periods are determined by the number of network clock cycles represented by the received RTSs. This pulse signal is then multiplied in frequency by N to recover the source node service clock.

Synchronous residual time stamp for timing recovery in a broadband network

A sub-band coder for video signals is disclosed, which is easily realizable in VLSI due to its simple structure. The coder uses sets of two-tap finite impulse response (FIR) filters having equal-magnitude coefficients to decompose the video signal at multiple stages. In the preferred embodiment the sub-band coder has an asymmetrical structure. With the arrangement, the sub-band decomposition effects what is equivalent to a multiple block-size Hadamard transform of the video signal in which coefficients are transmitted from various block-sized transforms of larger blocks of pixel data.

Multiple block-size transform video coding using an asymmetric sub-band structure

A compatible high-definition television (HDTV) coding system is described in this paper. This scheme allows the HDTV signal to be transmitted at a channel rate of 135 Mbps while an extended-quality television (EQTV) signal is encoded into a 45 Mbps subchannel. Three major coding strategies are used to achieve this goal: first, a hierarchical one stage pyramid coding structure is adopted to separate the HDTV signal into two paths; second, a DPCM scheme with run-length coding and a novel modified quantization scheme is used to compress the first path into a 45 Mbps stream for EQTV; and third, a simple PCM scheme is used to compress the second residual path into a 90 Mbps stream. At the receiver, the first path alone is used to reconstruct the EQTV signal, or the two paths are combined to generate the HDTV signal. This scheme has the advantages of relatively low hardware complexity since the minimum number of channels are used (for a complementary approach) and, since it is implementable with concurrent processing, no scan conversion buffers are required. Also, the proposed pyramid structure incorporates a feedback loop for the coding of the complementary high-definition residual signal. The system design details are investigated and computer simulation results are presented. Both the HDTV and EQTV reconstructed picture quality were found to be very good.

A Hierarchical HDTV Coding System Using A DPCM-PCM Approach

The effect of different finite impulse response analysis/synthesis filter pairs on sub-band coding performance is investigated based on spatial domain considerations. Deterministic as well as real video signals are used for rate and distortion evaluations of various filter pairs. A sub-band coding scheme suitable for advanced television (ATV) transmission in Broadband ISDN is then described based on the above simulations. The scheme uses the shortest kernel (2-tap) quadrature mirror filter (QMF) pairs for signal decomposition and reconstruction. The short span of the filter not only avoids quantization noise spread but also allows simple implementation in the high rate, high quality application considered here. Through successive QMF operations, the signal is decomposed into six bands to achieve low noise coding at the desired rate. The resultant coding scheme is identical to a Hadamard transform (HT) with non-constant block sizes. In addition to excellent coding performance, the hardware architecture is also much simpler than conventional transform coding or sub-band coding schemes.

Sub-Band Coding For ATV Signals Based On Spatial Domain Considerations

An intrafield subband high-definition TV (HDTV) system has been shown to be capable of achieving high-quality coding at 130 Mbps, roughly the net payload of the proposed CCITT H4 access channel. In order to attain the maximum achievable quality within the available bit rate for a variety of video scenes, the system requires intelligent rate control. The intelligent controller regulates the coder bit rate by properly selecting prefilters and quantizers for various bands. The control decision is derived from the occupancy of the transmit buffer and the bit rate history. Using rate statistics based on test pictures, a rate model for all coding modes is established which helps in choosing the proper coding mode that leads to the desired bit rate. Programmability and flexibility to accommodate future improvement of the control are assured by implementing the rate control with a commercially available microcontroller. Computer simulations of the coding system with the rate control indicate that consistent high quality can be achieved. >

Paul E. Fleischer

Papers

Synchronous residual time stamp for timing recovery in a broadband network

Multiple block-size transform video coding using an asymmetric sub-band structure

A Hierarchical HDTV Coding System Using A DPCM-PCM Approach

Sub-Band Coding For ATV Signals Based On Spatial Domain Considerations

An intelligent rate control for an intrafield subband HDTV coder