MXPA01003544A - High speed video transmission over telephone lines - Google Patents

Abstract

A system for sending and receiving audio/video data from a first location (2, 8-10) to a second location over existing telephone lines (18), includes device (6, 12) for converting audio/video data into a stream of packets, device for reconverting the stream of packets into audio/video data, and device (20) for bypassing telephone company voice switches (24), thereby allowing the packets to travel directly from the local telephone lines (18) to the long distant telephone network (28).

Description

TRANSMISSION OF HIGH-SPEED VIDEO ON TELEPHONE LINES BACKGROUND OF THE INVENTION FIELD OF THE INVENTION This invention relates to the transmission of data over communication links and, in particular, to the transmission of live, full-motion, totally dual video over existing telephone lines.

DESCRIPTION OF THE RELATED TECHNIQUE The use of the telephone system to transmit video as well as voice communications is well known, and has been practiced for several decades. However, existing systems are typically very expensive and only allow the transmission of images, particularly those that include motion, with notable delays, poor transmission quality or some combination of these problems. Full motion, and in particular, full-motion interactive video, requires the provision of a very significant amount of data in a relatively uninterrupted stream.The ability to achieve this on standard twisted copper conductor pairs has not yet been achieved. One of the most recent developments in the search to provide improved video transmission has been the use of dedicated Integrated Services Digital Network (ISDN) lines for data transmission, although this has resulted in a much improved transmission quality. , the expense of the ISDN lines is still a formidable obstacle to for its wide use The use of ISDN systems for video transmission has been described in, for example, Dagdeviren et al., U.S. No. 5,371,534, Aramaki et al., Patent of E.U.A. No. 5,751,339, and Sabni, patent of E.U.A. No. 5,184,345. An alternative for the transmission of video data over telephone lines is the use of the Internet or other computer networks. The Internet or other computer networks do not have the disadvantage of the cost of using ISDN lines. However, the Internet is based on the grouping of data that will be transmitted on it to small packages of data components called packages. These packages, in general, are of unequal length and contain information to indicate where they start and where they end, as well as the source and destination information. Packages from different sources travel on the Internet as a whole and, thus, must be recognized at any intermediate switching point and at their final destination points for recombination with other appropriately associated packets if successful transmission is to occur. The methodology of packages through their nature leads to potential delays in transmission and processing, and to a degradation in the quality of the transmission. In addition, the ever-increasing number of Internet users has combined the delay in transmission. Delays and degradation in quality, although generally non-critical in voice communications, can not be tolerated in applications that require high-speed data transmission, such as a fully dual, live, fully moving video transmission, in particular if quality video broadcasting or near broadcasting quality is required. However, to achieve a quality visual transmission totally in motion, and to do this in a relatively inexpensive way, a number of important challenges are presented. Many of these challenges could be overcome if a method, apparatus and system were developed, which can have the maximum advantage of the vast existing network of twisted copper conductor pairs, which carries most of the voice telephone communication in the present. The present invention presents a solution to the problem, hitherto unresolved, of the live transmission supply, totally in motion, totally double quality video broadcasting or almost broadcasting on existing telephone networks. The benefits of the extensive implementation of this technology are innumerable. Among its benefits are the vast expansion of use, both commercial and recreational, of the video phone and the creation of a new era of telephone video and interactive television communications. Potential uses extend to almost every field of endeavor, including, for example, the entertainment industry, the financial services industry, the field of corporate communications, hotel and travel services, applications in government agencies and public services, services doctors, educational services, and endless applications for the consumer. In the field of entertainment, for example, television viewers can participate visually and interactively not only with the host of a program, but also with other viewers. All-new television broadcasting programming formats will be made possible. New television programs can create chat rooms or video conversation and request the immediate reaction of the viewer to the new while it occurs. New organizations can jointly link their vast affiliate network "to cover news as it happens using their combined resources." Programs that sell products or services can be improved through the ability of viewers to interact through video as well as through voice with the merchant and other customers The non-expensive video conferencing, particularly given the secure nature of wireline landline telephone communications, is more compatible with the needs of many industries, such as the financial services industry, which in In the past, I had justifiable concerns regarding the transmission of information about the Internet, and the practice of the present business and financial community conference call that reports to the investment community can be greatly improved by providing video as well as Interactivity among participants. The hotel and travel industry, the ability to transmit video and hotel reception improve the services available to a business traveler who can maintain a constant interactive point-to-point contact with their home, office and other business associates. In general, private industry will have, if it chooses, the easy availability of creating its own relatively inexpensive closed circuit television network, without the concomitant costs of the ISDN network and T1 lines. Interactive communication between government agencies also has obvious important advantages, particularly in emergency situations, and the advantages in the medical field of visual communication accessible list and interactivity for the diagnosis and verification of patients, and the proliferation of services and knowledge among the providers of health care, patients and hospitals, is also self-evident. Also, the advantages in the field of education are important, ranging from video teaching and "wide area" classrooms to a budget reduction as information is offered from a central location with students trained to maintain a personal, visual interaction with the instructor. However, the most obvious beneficiaries of the present invention are individuals who, with the ability of a means to capture video, can, through the use of the present invention, employ standard telephone lines to communicate with another person or a group of others. people, in a quality video broadcasting or almost broadcasting. The above examples are not intended to be exhaustive and many other benefits derived from the present invention will be obvious to those skilled in the art.

COMPENDIUM OF THE INVENTION The present invention comprises means for capturing associated video and audio images, packaging them, transmitting them as a moderate continuous stream of video and audio data over existing telephone networks, decoupling the data carried by the packets, and reassembling the data as image images. video with associated audio at the location of the desired receiver. In its preferred form, the present invention also allows a combination of the images from various locations, so that not only is there interactivity between an individual transmitter and receiver, but also with multiple parts. To achieve high-speed transmission of video and audio data between source (s) and destination (s) and between switching points, preferably at speeds of approximately 2 megabits per second (mbits / sec), the present invention is particularly adapted to the interconnection with Digital Subscriber Line (DSL) technology, recently developed by the telecommunications industry for existing telephone networks. In its simplest form, a user has in its location a means to capture video images and the associated audio to convert them to a digital signal, and a means to convert a received digital signal to a presentation of video images and the associated audio . These means are generally well known in the art and may include devices such as a microphone, a camera, a video / audio encoder / decoder, a monitor, and an announcer. In accordance with the present invention, the audio signal transmitted or received is transmitted to or received from local telephone lines in packets of predetermined length, preferably equal. Each packet is generally encoded with the following information: (1) information indicating the start and end of the packet, (2) information indicating the packet length, (3) information indicating the algorithm used to encode the audio / video data, and (4) the same encoded audio / video data. In this way, even before the audio / video data leave the location of the transmitter, they have been transformed into packets that will be transmitted over ordinary communication lines (which in most cases can be regular lines of conductive pairs). twisted copper) to the central switching office maintained by the telephone company, and the audio / video data remain in package form until they reach the receiver's location. When the transmitted packets are received at the central office, the data passes through a derivation unit, designed in accordance with the teachings of the present invention, which recognizes the nature of the signal that is being presented. If the signal is only voice, then the transmission enters the telephone system to be processed in the usual manner of a voice transmission. However, if it is a signal carrying the impression of the present invention, as it is applied at the point of origin, then the signal is segregated and the order of data reception is continuously maintained so that they can be transmitted in a current relatively undivided, moderate of related packages. When that signal is received at the central office, which serves the receiver, it is identified by another referral unit. The packet stream is directed through the bypass unit to the receiver who has the apparatus designed in accordance with the present invention to capture the data and convert the packets to an audio / video signal, which in turn is presented on a video device with projected companion audio. Since the transmission and reception of packets is based on a first-in / first-out protocol, since the packets are pulled to be transmitted or received, they are immediately replaced with the following packets required to be transmitted or received. As a consequence, the sequence is maintained in a relatively uninterrupted manner. Although in reality it is preferred to transmit and receive all the data without any error, the system of the present invention does not need to transmit and receive 100% of the data, since an acceptable video signal, in fact of very high quality, will be enabled. even with the loss of some data from the data stream. The present invention in this way provides means to enable the transmission and reception of live, fully dual, fully moving video data and accompanying audio data over existing telephone lines with the concomitant benefit of allowing parties, in remote, visual locations. and audibly communicate with each other. Also, if one of the parties to the connection is a television studio that originates a video broadcast, the received video communications will be and can be transmitted and received between any of the two points served by the conventional telephone network, at a cost that does not deteriorate the wide use. These and other objects and advantages of the present invention will become more apparent to those skilled in the art after consideration of the accompanying drawings and the following description of the preferred embodiments, which are considered a form of illustration and example only, but are not constructed in no way to limit the invention described and claimed herein.

BRIEF DESCRIPTION OF THE DRAWINGS Figure 1 is a schematic diagram showing the end of the transmission / reception of a typical embodiment of the present invention together with its connection to the existing telephone network. Figure 2 is a flow diagram of the video encoder process. Figure 2A is a flow diagram of the audio encoder process. Figure 2B is a flowchart of the video decoder process. Figure 2C is a flow diagram of the audio decoder process. Figure 3 is a schematic diagram showing a first embodiment of the format of a fixed length packet carrying audio / video data. Figure 3A is a schematic diagram showing a second embodiment of the format of a fixed-length packet carrying audio / video data. Figure 3B is a schematic diagram showing a third embodiment of the format of a fixed length packet carrying audio / video data.

Figure 3C is a schematic diagram showing a fourth embodiment of the format of a fixed length packet carrying audio / video data. Figure 4 is a flow chart of the data transmission process. Figure 4A is a flowchart of the data reception process. Figure 5 is a schematic diagram showing the hardware components of the branch unit.

DESCRIPTION OF THE PREFERRED MODALITIES Figure 1 shows components of the invention included in the end of transmission / reception of a connection that transmits and receives audio / video packets according to the invention. A monitor 2 capable of presenting a broadcast quality video image, said video image has been transmitted from a remote location on the existing telephone network 4 through the packages created in accordance with the present invention, is connected to an encoder / video / audio decoder 6 of a design known in the art capable of using or decoding conventional data encoding a protocol such as, for example, AVI, MPEG-I or MPEG-II. An announcer 9 capable of projecting audio data is also connected to the video / audio encoder / decoder 6. A microphone or other device 8 capable of collecting audio data is also connected to the video / audio encoder / decoder 6. Finally, a camera 10 capable of receiving video data is also connected to the video / audio encoder / decoder 6. The video and audio data, which have either been encoded or decoded by the audio / video encoder / decoder 6, are already are encoded as decoded by the video and audio encoder / decoder process 12. The packets are either transmitted to a remote location being sent in a stream or are received from a remote location in a stream and the stream is broken in their data encoded constituents by the process of transmitting / receiving data 14. A modem 16, capable of interconnecting between existing telephone lines and a location for receiving and transmitting video and audio data, and compatible with some variation of the DSL protocol, such as, for example, CDSL (Consumer DSL) or ADSL (Asynchronous DSL), receives and transmits the packets of the invention to and from existing local telephone lines 18. A referral unit 20, comprising a sensor 22, and capable of distinguishing between an ordinary voice call and a DSL call carrying the packets of the invention, is connected to telephone lines existing premises 18. Branch 20 is installed in the central office of the telephone company. The branch unit 20 allows an ordinary voice call to pass unimpeded to the existing telephone company voice switch 24 at the central office. However, a DSL call carrying the packets of the invention is connected in branch 26 directly over the existing long distance telephone network 28 through the branch unit 20, thus deriving the existing telephone company voice switch 24. Figure 2 is an illustrative flow chart of the video encoder process 29. A typical video / audio encoder / decoder 6, employing the encoding process 29, accepts a video input 30, and feeds it through a 32 block processor, which divides it into many blocks comprising an image. An image or frame thus divided into blocks is stored in the memory 34. When the next image or frame is obtained from the video input 30, a motion sensor 36 determines the amount of change or movement in the image. Based on the coding algorithm, Figure 2 based on a conventional coding algorithm, mainly H.320, the motion compensator 38 decides which block (if any) has changed or which is in agreement. If, based on the coding algorithm, the change in a portion of the image is sufficient, a data mixer or subtractor 40 determines the exact difference and stores the new image in memory 34. The difference between the two frames stored in the memory 34 is selected, by the selectors 42, and processed by a quantizer 44, an inverse quantizer 46, an orthogonal processor 48, and a reverse orthogonal processor 50. An output encoder 52 assembles an output data stream 54, the which includes information regarding the coding process (type, frame lengths, resolution, etc.).

Figure 2A shows the process of audio encoder 55. An audio input 57 is sent to an analog-to-digital converter. The data is sent to a memory block for short-term storage in a memory 61. A time control module 63, which takes a clock pulse 65 from the video encoder stream 54, determines where to insert the data in the output stream 54 (see Figure 2) to synchronize the audio to the video. The audio data is inserted as a bit (s) at particular locations in the video stream 54 through the audio multiplexer 67 to form a combined audio / video stream 69. Figure 2B shows the video decoding process 71. The input decoder 73 converts the decoder input current 108 (see Figure 4A) into blocks for short-term storage in the memory 75. A block processor 77 evaluates the "changed" blocks and through the use of a block inserter 79, replaces those blocks that have changed to memory 75. This information is then sent as a complete video frame 81. Figure 2C shows the audio decoder process. The decoder input stream 108 is sent through an audio demultiplexer 83 to intercept the bits containing audio information. The audio data is sent to the memory 85 as short-term storage, while a clock pulse 87, taken from the video stream 81 (see Figure 2B), is used by a time control module 89 to determine when take out the audio, so that it is in an appropriate time relationship with the video. The audio data is sent through an analog-to-digital converter 91 to convert the data back to an analog audio signal. Figures 3, 3A, 3B and 3C show formats of several packet modes of a fixed length of 27 bytes. When sending and receiving packets of a fixed length directly from and at the sending and receiving location, the existing telephone company equipment is not required to perform processing and / or frame input to the buffer that could encourage transmission. This is extremely important for moderate video transmission. The packages do not contain address / destination information and site hardware information since they are not sent to a pre-determined location directly by the telephone company's equipment. The packets will be sent to the appropriate destination based on the telephone number dialed by the DSL 16 modem (see Figure 1). Sending and receiving packets from and to the end-user locations allows high-speed transmission even over the last link from the end-user location to the central office of the telephone company. Figure 3 shows a first mode of the format of the packets. The packets start with a two-byte start flag 56 and end with a two-byte 58-end flag. These flags indicate the telephone company equipment where each packet starts and ends and conforms to the specifications of the telephone company, which are a function of a particular telephone company network on which the packets are traveling. After the start flag 56 there are two bytes 60, which confirm the packet length. A byte 62 indicating the type of coding algorithm used to create the audio / video data in the packet follows the bytes 60 that confirm the length. The length confirming bytes 60 and coding type byte 62 are ignored by the telephone company equipment, being read by the hardware in the sending and receiving locations and only by the derivation units. Finally, twenty bytes of the actual encoded audio / video data 64 searched for to be transmitted form the rest of the packet. The format of the audio / video data is determined by the type of coding algorithm used to create the audio / video data. Figure 3A shows a second mode of the format of the packets. The second mode differs from the first mode in that it does not include the length confirmation bytes 60 found in the first mode. If the hardware, both in the sending and receiving ends, it is standardized to recognize that each packet is always 67 bytes long, any error can be calculated using only the start 56 and stop 58 flags. This could allow the audio data / video 66 occupy an increased length of 22 bytes. However, the first mode may be necessary in a situation where the hardware is located at either end of transmission or reception that interacts with the packets and is not programmed to be aware of the length of the packet. Figure 3B shows a third mode of packet format. The third mode differs from the first mode in that one byte 62 indicating the type of coding in the first mode has been enlarged to two bytes 68 in the third mode due to the possible release of additional coding algorithms in the future. For example, there is the possibility that MPEG-III and MPEG-IV can be released soon. A single byte allows the identification of up to eight coding algorithms, but the identification of more coding algorithms may be necessary. Increasing this byte to two bytes could allow additional coding algorithms to be signaled to the encoding / decoding devices. The use of two bytes 68 for the type of coding algorithm could decrease the length 70 of the audio / video data by one byte when compared to the audio / video data 64 in the first mode, leaving the new length of the audio / video data 70 in 19 bytes. Figure 3C is a fourth mode of packet format. This differs from the third modality of the format of the packets in that the two bytes 72 that indicate the length of the packet in the third mode have been eliminated in the fourth mode for the same reason that they were eliminated in the second mode. The elimination of the length information in the fourth mode allows the audio / video data 74 to be increased by two bytes in length, leaving the length of said audio / video data 74 in 21 bytes. Figure 4 is a flow chart of the data transmission process, which assembles the output data stream from the encoder to a stream of packets. The encoder output data stream 76 (marked 54 in Figure 2) is read by a data counter 78, and passed to the frame memory 80, being stored in a first input or first output (FIFO) base . The data counter 78 sends flags to the frame memory processor 82 to notify the frame memory 80 of the lack or corruption of bits. The frame memory processor 82 sends the bit location information of the missing or corrupted bits to the image corrector 84, which determines an appropriate correction action and sends the corrected bits to the frame memory 80. The correction process error will be an average process so that if a corrupt or missing bit is detected, the image corrector 84 will take the value of the bit before and the value of the bit after the corrupt or missing bit and insert an average of two instead of the corrupt or missing bit. The signal is transmitted from the frame memory 80 in a FIFO base to the packer 86, which produces packets in one of the formats represented in the modes shown in Figures 3, 3A, 3B and 3C. The packets are then sent through a network interface converter 88, which inputs the signal to conform to the connection specifications of a DSL modem, said connection specifications being, for example, standard protocols such as 10Base-T, Thinnet and TCP / IP. The DSL modem 90 then sends the pre-formatted packets to the central office via the local telephone line 92. Figure 4A is a flow diagram of the data reception process., which disassembles a stream of packets to an input data stream of the decoder. A packet stream is received from the local telephone line 94 through a DSL 96 modem and is sent through a network interface converter 98, which removes the introduction to the buffer placed in the packets through the converter of network interface 88 at the end of the transmission of the connection. The packets are then sent through a data counter 100 and a frame memory 102. The data counter 100 sends flags to the frame memory processor 104, which informs the frame memory 102 that bits 106 are going to be discarded as part of the start and stop flags, the length bytes, and the encoding type bytes. The data, without the scrapped bits, is sent as a decoder input stream 108 to a decoder, which converts the data to a variable image on a presentation monitor with accompanying sound projected from a speaker. Figure 5 is a schematic diagram showing the hardware components of the branch unit 112 installed in the central office of the telephone company. When an off-hook condition is generated by the telephone company telephone or a splitter connected to a DSL modem, the condition is sensed by a line condition detector 110 within the bypass unit 112. The input signal on the line Local telephone 114 is then sent to a data detection circuit 116, which "hears" the telephone line 114 to determine if the packages of the invention are present. If no packets are present, the data detection circuit 116 sends the signal to a standard telephone company voice load coil 118 as required by the voice switches and then that signal is transmitted 120 to the company voice network switch standard telephone If the packets are detected on the line, the data detection circuit 116 derives the standard telephone company voice network switch and sends 122 packets directly out of the telephone company's long distance network. The incoming voice calls and the packets 124 are received by the branch unit 112, which, through a directional combiner 126, sends the signal to the intended receiver. Although preferred embodiments have been described, it will be understood by those skilled in the art that various modifications, changes or alterations may be made to the invention described and presented herein without departing from its scope or equivalents as claimed in the appended claims. Other modifications, very numerous to mention, will easily occur to those skilled in the art.

Claims (16)

1. - A system for sending, in at least a first location, over telephone lines, first audio / video data recorded in at least said first location, receiving, in at least said first location, over the telephone lines, a reproduction of second audio / video data in at least one second location, send, in at least said second location, over the telephone lines, the second audio / video data recorded in at least said second location, and receive, in at least a second location, over telephone lines, a reproduction of the first audio / video data recorded in at least said first location, the system comprises: a. means for converting the first audio / video data, in at least said first location, to a first stream of packets that will be sent over said telephone lines; b. means for converting a second stream of packets into at least said first location to the reproduction of the second audio / video data that will be presented in at least said first location; c. means for converting the second audio / video data, in at least said second location, to the second stream of packets that will be sent over the telephone lines; d. means for converting the first stream of packets into at least one second location to reproduction of the first audio / video data that will be presented in at least said second location; and e. a total of at least two means for deriving the telephone company voice switch in at least two telephone company headquarters, thereby allowing the first packet stream and the second packet stream to pass directly between at least the first local telephone line and at least the second local telephone line, respectively, and at least one long-distance network; said system being able to send the first audio / video data and the second audio / video data, and receive said reproduction of the first audio / video data and said reproduction of the second audio / video data in a totally double way , totally in movement and alive.

2. A system according to claim J, wherein said means for converting the first audio / video data comprise: a. an audio / video encoder / decoder producing an audio / video encoder output data stream; b. a packer accepting a processed form of the audio / video encoder output data stream after processing said audio / video encoder output data stream between the audio / video encoder / decoder and the packer, said packer producing a stream of packets not inserted into the buffer; and c. a Digital Subscriber Line (DSL) modem accepting the first packet stream and sending the first packet stream on at least one first local telephone line, the first packet stream being produced by entering the buffer of said packet stream not inserted into the buffer.

3. A system according to claim 2, wherein said audio / video encoder / decoder comprises: a. a block processor that accepts video data included in the first audio / video data and divides said video data into a plurality of blocks comprising an image; b. a memory for storing a plurality of images; c. a motion sensor for determining an amount of movement between said image and an immediately following image; d. a motion compensator to decide which block, if any, has changed and to what degree between said image and said immediately following image; and. a data subtractor which determines an exact difference between said image and the immediately following image and stores said immediately following image in the memory if a change between the image and the immediately following image is sufficient under a coding algorithm used; F. at least one selector for selecting a difference between said image and the immediately following image stored in the memory; g. a quantizer, an inverse quantizer, an orthogonal processor, and an inverse orthogonal processor for processing said difference between said image and the immediately following image; and h. an output encoder that accepts the input of said block processor and said inverse quantizer and outputs a stream of video data.

4. A system according to claim 3, wherein said means for converting the first audio / video data comprises: a.- an analog-to-digital converter that accepts audio data included in the first audio / video data. and convert audio data from analog to digital; b. a memory that stores the audig data sent to the memory by the analog-to-digital converter; c. a time control module that accepts a clock pulse of said video data stream and determines where to insert the audio data into the video data stream to synchronize audio to video; and d. an audio multiplexer that accepts the video data stream and accepts the audio data from the memory and combines said audio data and said video stream according to the determination of the time control module to produce a data stream of audio / video output.

5. A system according to claim 4, wherein said means for converting the first audio / video data comprises: a. a data counter that accepts the audio / video encoder output data stream and outputs flags corresponding to missing or corrupted bits in the audio / video output data stream; b. a frame memory that accepts the audio / video encoder output data stream of the data counter and outputs said processed audio / video encoder data stream to the packer; c. a frame memory processor that accepts said flags and notifies the frame memory of said missing or corrupted bits: d. an image corrector that accepts the bit location information of said missing or corrupted bits of the frame memory processor, determines appropriate correction for the missing or corrupted bits, and sends correct bits to the frame memory; and e. a network interface convert which accepts said packet stream not input to the buffer of the packer, inputting said buffer stream into the buffer memory to confirm with connection specifications of the DSL modem, thus producing the first current of the packet. packets, and send said first packet stream to the DSL modem.

6. A system according to claim 2, wherein each of at least two means for deriving said telephone company voice switch comprises: a. a line condition detector for detecting an off-hook condition in one of at least a first local telephone line and at least one second local telephone line generated by a source selected from the group consisting of: a telephone, a splitter connected to the telephone line; DSL modem, and a splitter connected to a DSL modem in at least a second location; b. a data detection circuit that receives a signal from the source through one of at least the first local telephone line and at least one second local telephone line and through the line condition detector, determines whether the first packets or the second stream of packets is present in said output, and sends said signal directly to at least one long-distance network if the first or second packet stream is present, thereby bypassing the telephone company voice switch; c. a voice load coil that receives a signal from said data detection circuit if the first and second packet streams are not present in said signal, the voice load coil transmitting the signal to the telephone company voice switch; d. a directional combiner receiving from at least one long-distance network a transmission selected from the group consisting of: a voice call, the first packet stream, and the second packet stream, and send the transmission to the intended receiver.

7. A system according to claim 1, wherein said means for converting the second packet stream comprises: a. a Digital Subscriber Line (DSL) modem that receives the second packet stream from at least one local telephone line; and b. an audio / video encoder / decoder receiving an audio / video decoder input data stream from the DSL modem after processing said second packet stream between the DSL modem and the audio / video encoder / decoder.

8. A system according to claim 7, wherein the audio / video encoder / decoder comprises: a. an input decoder that receives the audio / video decoder input data stream and converts a portion of said audio / video decoder input data stream into blocks forming a video frame; b. a block processor that evaluates those blocks that are changed from a video frame immediately before said video frame; c. a memory that stores blocks forming a video frame and outputs said blocks as a complete video frame in a video output data stream; and d. a block inserter controlled by the block processor, the block inserter replacing said changed blocks in memory.

9. A system according to claim 8, wherein the audio / video encoder / decoder comprises: a. an audio demultiplexer that receives the input data stream from the audio / video decoder and outputs audio data included in the audio / video decoder input data stream; b. a memory that receives said audio data; c. a time control module that uses a block pulse taken from the video output data stream to make said output memory to audio data in appropriate time relative to the video output data stream; and d. a digital-to-analog convert to convert the audio data output from the digital form memory back to an analog audio signal.

10. A system according to claim 7, wherein said means for converting the second stream of packets further comprises: a. a network interface converter that receives the second packet stream from the DSL modem and removes the introduction to the buffer placed in the second packet stream by a second network interface converter in at least a second location; b. a data counter that receives the second packet stream not inserted into the buffer of the network interface converter and generates flags based on the second stream of packets not input to the buffer; c. a frame memory processor that receives said flags from the data counter; and d. a frame memory which receives the second packet stream not input to the data counter buffer, which receives communications from the frame memory processor with respect to which bits of the second packet stream not input to the buffer it discards for enabling the creation of said reproduction of said second audio / video data, which discards said bits of the second packet stream not input to the buffer, and which outputs the audio / video decoder input data stream to the audio / video encoder / decoder.

11. A system according to claim 1, wherein each of the packets are of constant length.

12. A system according to claim 1, wherein each of the packets comprises: a. a start flag; and b. an end flag.

13. A system according to claim 12, wherein each of the packets further comprises data specifying an encoding algorithm through which said audio / video data were encoded before being converted to said packet stream.

14. A system according to claim 13, wherein each of the packets further comprises data specifying the length of each of the packets.

15. A system according to claim 1, wherein each packet of the first packet stream and the second packet stream comprises: a. a start flag; b. data specifying an encoding algorithm through which said audio / video data was encoded; c. encoded audio / video data; and d. a flag to finish.

16. A system according to claim 13, wherein each packet further comprises data specifying the length of said packet.