RU46397U1 - INFORMATION TRANSMISSION SYSTEM - Google Patents

Abstract

The claimed utility model allows you to create a digital broadcast television system for mobile phones and interactive video surveillance systems for objects from a mobile phone in networks with an unlimited bandwidth of the communication channel. This is achieved due to the fact that its transmitting part additionally contains a buffer unit connected to the inputs of the multiplexing unit, the output of which is connected to the input of the transmission unit, and the receiving part additionally contains an analysis unit that analyzes the current throughput of the communication channel and predicts its change, input which is connected to the output of the receiving unit, and the output through the feedback channel to the control input of the multiplexing unit, while the outputs of the compression unit forming several output shackles with various compression ratios are connected to the inputs of the buffer block. 1 n.p. f-ly, 1 Fig.

Description

The utility model relates to the field of telecommunications, namely to the transmission of audio / video information via cellular communication channels with the display of a synchronized continuous audio / video stream on a mobile phone screen. This utility model can be widely used for digital broadcast television on mobile phones and interactive monitoring of objects from a mobile phone.

Information transmission systems are known, characterized in that the video / audio information received from the source is digitized, compressed (compressed), then noise-resistant coding is performed for transmission to the communication channel, then the transmitted information is decoded with restoration from interference, then decompression and audio / video information at the receiving part (1).

A known information transmission system used to transmit audio / video information in cellular networks (2). The system contains a series-connected digitization unit, a compression unit,

a transmission unit, which together form a transmitting part connected via a communication channel with a receiving part, which is formed by a series-connected receiving unit and a decoding unit.

The operation of the system consists in the fact that the transmitting part generates a compressed video / audio stream with a fixed output bit rate (output bit rate is a parameter that sets the mode of information received per unit time at the output of the compression unit). Then, via the Internet and a cellular communication channel, it transfers it to a mobile phone. The mobile phone buffers the input video / audio stream, then decompresses it and displays it. In this case, stringent requirements are imposed on the correspondence between the bandwidth of the communication channel and the output bit rate of the generated video / audio stream on the transmitting side.

For continuous display of the audio / video stream on the receiving part of the system (mobile phone screen), a strict correspondence between the output video / audio stream and the communication channel bandwidth is required, which significantly limits the implementation of the system on cellular networks with an unlimited channel bandwidth. Fluctuations in the bandwidth of the data transmission channel lead to a loss of continuity and synchronization of the display of the audio / video stream. Significant delay in displaying the audio / video stream due to buffering on the mobile phone. The channel switching process is complicated

- the procedure for turning on a specific channel takes at best about 2-3 minutes.

The technical result obtained by implementing the claimed utility model is expressed in reduced requirements for the communication channel, in particular, bandwidth fluctuations, as well as in the possibility of self-tuning to various output video / audio streams depending on the channel bandwidth, which significantly improves user characteristics and allows you to work in a wide range of data transfer speeds.

The specified technical result is achieved in that the information transmission system containing the transmitting and receiving parts connected by a communication channel, the transmitting part comprising a digitizing unit, the output of which is connected to the compression unit, as well as a transmission unit whose output is the output of the transmitting part of the system, the receiving part which contains a receiving unit connected to the input of the decoding unit, converting the information into an output signal format, connected to the input of the display unit, characterized in that the transmitting part additionally contains a buffer unit connected to the inputs of the multiplexing unit, the output of which is connected to the input of the transmission unit, and the receiving part additionally contains an analysis unit that analyzes the current throughput of the communication channel and predicts its change, the input of which is connected to the output of the reception unit, and the output through feedback channel - with

the control input of the multiplexing unit, while the outputs of the compression unit, forming several output streams with various compression ratios, are connected to the inputs of the buffer unit.

The drawing shows a block diagram of an information transmission system.

The indicated system comprises a digitization block 1, a compression block 2, a buffer block 3 (containing separate nodes 3.1, 3.2, ..., 3.N for buffering various output streams), a multiplexing block 4, a transmission block 5, a communication channel 6, block 7 reception, analysis unit 8, decoding unit 9, display unit 10, feedback channel 11. Blocks 1, 2, 3, 4, 5 form the transmitting part 12 of the system, and blocks 7, 8, 9, 10 form the receiving part 13.

The transmitting part 12 is intended for generating streams with a given output bit rate and transmitting them to the communication channel 6. The receiving part 13 is intended for receiving, decoding and displaying information.

Block 1 provides the conversion of the analog signal into a digital code, carrying out frame-by-frame input and conversion.

Block 2, processing digitized video information, forms a series of streams of the same content (video / audio) with different output bit rates (for example, 10 Kbit / s, 15 Kbit / s, 20 Kbit / s, etc., the lower limit is the minimum bandwidth the capacity of channel 6, the upper limit is the maximum throughput of channel 6, the discreteness of the row should be minimal and determined

minor changes in the perception of the image when switching from stream to stream and the technical capabilities of the server to form multiple streams). The implementation of the video information compression algorithm during the formation of the output stream is such that a constantly set output bit rate is supported by changing the compression ratio of the input video signal. Fluctuations in the output bit rate are insignificant (no more than 10%).

Block 4 switches between the output streams and the receiving part 13 in accordance with the information received from block 8.

Block 5 implements error-correcting coding and packet transmission of the output stream to block 7. It generates packets of a given volume from the output stream and sends them with a frequency that ensures the reception of video / audio information during reception and decoding. For example, with a stream with an output bit rate of 20 Kbps and a packet length of 256 bytes per second, 10 packets will be sent. With the same channel bandwidth, block 7 will receive these 10 packets within 1 second. At the same time, communication channel 6 manages to skip a given amount of information per unit time, margin dT = 0 (where dT = dT1-dT2, dT1 is the temporal length of a packet at a given output bit rate of a stream, dT2 is the real time interval for which block 7 received a packet data). In case the bandwidth of channel 6

higher (40 Kbit / s), the real packet transmission time per unit time is less than the temporal packet length at a given output bit rate, with dT> 0, that is, channel 6 manages to skip a given amount of information with a margin of dT per unit time. If the throughput of channel 6 is lower (10 Kbit / s), the real packet transmission time per unit time is longer than the temporal length of the packet at a given output bit rate, while dT <0, that is, channel 6 does not have time to miss unit of time a given amount of information.

Thus, the parameter dT directly depends on the throughput of channel 6 at a given output bit rate. Its analysis by block 8 in the process of transmitting packets allows us to estimate the current channel throughput and predict the next based on a number of previous values. In this case, the parameter of the channel 6 bandwidth change is calculated as K = T1 / (T1-dT). The predicted value is estimated as a linear approximation (in the simplest case) of changes in a given parameter based on previous values. Then, the current output bit rate of the transmitted stream is compared with the possible predicted bandwidth of channel 6. In this case, the value of the new output bit rate of the stream should be lower than the predicted bandwidth of channel 6. Next,

selection of one of the multiplexed output streams providing the necessary output bit rate.

The continuity of the display at the receiving part 13 during the transition between the various output streams of the transmitting part 12 is supported by the internal buffer of block 7.

The input audio / video signal is digitized in block 1, and fed to the input of block 2, which, on the basis of the MPEG4 encoder (MPEG4-standard for compressing (encoding) audio / video information) forms several output streams with different compression ratios and / or frame frequencies and, accordingly, with different output bit rates, which are received at the inputs of block 3. Block 4 (broadcast multiplexer) provides access from the receiving part 13 of the system to the required stream. At the initial time, access is made to the stream with the minimum bit rate, then, depending on the signal from block 8 carrying information about the bandwidth of channel 6, a stream with a bit rate corresponding to the predicted bandwidth of channel 6 is selected. Block 5 generates information packets and transmits video / audio stream to channel 6. Communication channel 6 is GPRS (EDGE) channel of a cellular operator (GPRS - Packet Data Networks - a technology that is an implementation of a pack hydrochloric GSM standard switching networks previously used only circuit switched technology. Instead of transmitting the continuous data stream through

permanent connection with packet switching, the network is used only if there is data for transmission). Block 7 receives information packets and their channel decoding. Block 9 converts the video / audio stream from MPEG4 format to a format for display on block 10 (on the smartphone screen). At the same time, information from block 7 goes to block 8, which analyzes the current throughput of channel 6 and predicts its change, forming a command to block 4 to switch to a stream whose output bit rate corresponds to the predicted one.

The claimed utility model allows you to create a digital broadcast television system for mobile phones and interactive video surveillance systems for objects from a mobile phone in networks with an unlimited bandwidth of the communication channel.

Sources of information taken into account:

1. R.E.Bykov, K.V. Ivanov, A.A. Mantsvetov "Digital image conversion", Moscow, 1992, pp. 147-150;

2. James F. Cornes, Keith W. Ross, Computer Networks, Moscow, 1996, pp. 536-547 (prototype).

Claims (1)

An information transmission system comprising transmitting and receiving parts connected by a communication channel, the transmitting part comprising a digitizing unit, the output of which is connected to the compression unit, and a transmission unit, the output of which is the output of the transmitting part of the system, the receiving part of which contains a receiving unit connected to the input of the decoding unit, converting the information into an output signal format, connected to the input of the display unit, characterized in that the transmitting part further comprises a buffer unit connected about the inputs of the multiplexing unit, the output of which is connected to the input of the transmission unit, and the receiving part additionally contains an analysis unit that analyzes the current throughput of the communication channel and predicts its change, the input of which is connected to the output of the reception unit, and the output through the feedback channel to the control the input of the multiplexing unit, while the outputs of the compression unit, forming several output streams with various compression ratios, are connected to the inputs of the buffer block.