KR101528647B1 - Code enhanced staggercasting - Google Patents

Abstract

A method and architecture for handling signal communication between an encoder and a decoder operating in accordance with an ATSC standard adapted for mobile handheld transmission is disclosed. The method and apparatus include transmitting packets and duplicate packets in accordance with time, space and frequency diversity to improve redundant error handling.

Description

[0001] CODE ENHANCED STAGGERCASTING [0002]

This application is related to U.S. Provisional Patent Application No. 61 / 003,041 entitled " Code Enhanced Staggercasting ", which is incorporated herein by reference, and U.S. Provisional Patent Application No. 61 / 003,041 entitled " ATSC M / H Mobile Broadcast for Portable Services & 002,977.

The present invention relates to data transmission in a multi-mode transmission system. In particular, the invention relates to a transmission system in which a plurality of code rates within a single standard transmission protocol such as ATSC can be used for data transmission.

Over the past several decades, video transmission systems have moved from analog to digital formats. In the United States, the transition from the National Television System Committee (NTSC) analog television system to the Advanced Television Systems Committee (ATSC) A / 53 digital television system is at a final stage. The A / 53 standard defines "video encoder input scanning formats and preprocessing and compression parameters of a video encoder, preprocessing and compression parameters of an audio encoder input signal format and audio encoder, service multiplexing and transmission layer characteristics and norm specifications, Specifications of the parameters of the system including the RF / transmission subsystem ". The A / 53 standard defines how source data (e.g., digital audio and video data) should be processed and modulated into signals to be transmitted over-the-air. This process adds redundant information to the source data so that the receiver can recover the source data even if the channel adds noise and multipath interference to the transmitted signal. The redundancy information added to the source data reduces the effective rate at which the source data is transmitted, but increases the likelihood that the source data is successfully recovered from the received signal.

The ATSC A / 53 standard development process focused on HDTV and fixed reception. The system was designed to maximize the video bit rate for large high definition televisions that are already entering the market. Transmission broadcasting under the ATSC A / 53 standard, however, presents difficulties for mobile receivers. There is a need for improvements to the standard for robust reception of digital television signals by mobile devices.

Recognizing this fact, in 2007, ATSC created a process to develop standards that broadcasters would be able to deliver television content and data to mobile and handheld devices via their digital broadcast signals. . In response, several proposals were received. The resulting standard, called ATSC-M / H, is intended to be backward compatible with ATSC A / 53, enabling the operation of existing ATSC services on the same RF channel without adversely affecting existing receiver equipment .

As with some of the proposed ATSC-M / H systems, many systems for transmission to mobile devices perform periodic transmissions. These systems may include a preamble in their transmissions to aid operation of the receiver system. Preambles typically contain known information that parts of the receiving system can use for training for reception enhancement, which can be particularly useful in difficult environments such as those found in mobile operation. These systems can also encode data at varying code rates. For example, the code rate or information rate of a Forward Error Correction (FEC) code, such as a convolutional code, refers to a portion of the total amount of information that is not duplicated. The code rate is typically a fraction. If the code rate is k / n, for every k bits of useful information, the coder will generate n bits of data as a whole, of which nk is redundant.

Existing ATSC M / H proposals include the use of separable block codes to enable code-enhanced time and frequency diversity. In the example of a transmission coded at 1/2 speed, mobile data is input into an FEC coder, which outputs 2 bytes for each input byte. The two bytes represent the original data and the redundant data. The receiver can receive either the original data or the duplicate data at the receiver threshold of the original data. When both streams are received, a coding gain advantage is given so that the receiver can recover the data at a threshold below the original data. The mobile and gait behavior of telecommunications equipment raises some of the biggest challenges due to other obstacles as well as extreme transmission channel disturbances that exist due to buildings and moving vehicles. A system that provides data in a redundant manner may be used. It would be desirable to utilize redundant information through frequency, time and space diversity to improve reception on mobile devices.

According to one aspect of the present invention, a method is provided.

According to another aspect of the present invention, another method is provided.

As described herein, the present invention enables diversity in a mobile broadcasting system, such as the proposed ATSC-M / H system, while allowing backward compatibility with legacy transmission and reception paths such as ATSC A / Provided are a method and apparatus for transmitting data using city and data redundancy. While the present invention has been described as having a preferred design, the present invention may be further modified within the spirit and scope of the present disclosure. This application is therefore intended to cover any variations, uses, or adaptations of the invention using the general principles of the invention. This application is also intended to cover such departures from the present disclosure as come within known or customary practice in the art to which this invention pertains and which fall within the scope of the appended claims. For example, the described techniques may be designed for different kinds of data, or may be applied to transmission systems that use different coding, error correction, redundancy, interleaving or modulation techniques.

1 is a block diagram of one embodiment of a terrestrial broadcast transmitter for mobile / handheld reception of the present disclosure;
2 is a block diagram of one embodiment of a portion of an exemplary mobile / handheld data stream of the present disclosure.
3 is a block diagram of one embodiment of an exemplary data frame of the present disclosure;
4 is a block diagram of one embodiment of a terrestrial broadcast receiver for mobile / handheld reception of the present disclosure;
5 is a block diagram of one embodiment of a decoder of the present disclosure;
6 is a block diagram of another embodiment of a decoder of the present disclosure;
7 is a block diagram of a terrestrial broadcast environment in accordance with the present invention;
8 is a block diagram of one embodiment of a portion of a transmitter in accordance with the present disclosure;
The examples provided herein illustrate preferred embodiments of the present invention, and these examples should not be construed as limiting the scope of the present invention in any way.

Referring now to the drawings, and more particularly to FIG. 1, a block diagram of one embodiment of a terrestrial broadcast transmitter for mobile / handheld reception of the present disclosure is shown. The embodiment 100 of FIG. 1 includes a plurality of signal transmission means, such as an MPEG transport stream source 110, an ATSC M / H preprocessing path 115, and a global ATSC A / 53 processing path. The components in the ATSC-M / H preprocessing path 115 include a packet interleaver 120, a serial concatenated block coder 125, a packet deinterleaver 130, an MPEG transport stream header corrector 135 And a preamble packet inserter 140. The raw ATSC A / 53 processing path 145 includes a data randomizer 150, a Reed-Solomon encoder 155, a byte interleaver 160, a trellis encoder 165, a sync inserter 170, a pilot inserter 175 and a modulator 180. [

In the ATSC-M / H preprocessing flow, MPEG transport data 112 incoming from MPEG transport stream source 110 is received at packet interleaver 120. The packet interleaver 120 rearranges the sequenced number of bytes into different sequences to improve bit error rate and frame error rate performance. In this illustrative embodiment, the packet interleaver 120 takes the bytes in a row-wise order from a fixed number of consecutive packets, and outputs the bytes in column units. In this manner, all of the first bytes of packets are grouped together, and all of the second bytes of packets are grouped together, continuing to the last bytes of the packets. Each source packet is an MPEG transport stream packet from which the sink bytes have been removed, so the length of each packet is 187 bytes. The number of packets in each code frame is equal to the number of source symbols required for the GF (256) serial concatenated block code.

The interleaved data is then coupled to a Galois Field (GF) 256 serial concatenated block coder (SCBC) 125. The GF (256) Serial Concatenated Block Code (SCBC) 125 decoder will take different forms depending on the rate mode for the current symbols. Generally, this is constituted by constituent decoders which repeatedly decode soft information in a turbo decoding manner. The SCBC 125 codes the packet interleaved data in one of a plurality of forms depending on the desired data rate and codeword length. The SCBC 125 consists of one or more component GF (256) codes cascaded in series, which is linked by GF (256) code optimized block interleavers to improve overall code performance. This may then be followed by a GF (256) puncture optionally to achieve the desired codeword length.

In particular, the finite field GF (p ⁿ ) is a mathematical set containing a finite number of elements p ⁿ (the values of p and n are integers). A specific finite field is defined using the constructor polynomial g (x). Each element of a finite field can be represented by a unique bit pattern having n bits. Also, the unique polynomial of the p ^{n -th} order can be associated with each element, where each coefficient of the polynomial is between 0 and p-1. Also, mathematical operations in finite fields have important properties. The addition of two elements of a finite field GF (p ⁿ ) is defined as an element associated with a polynomial with coefficients that are modulo-p sum of the coefficients of the polynomials associated with the two elements being added. Similarly, the multiplication of two elements is defined as a modulo multiplication of the polynomial g (x) associated with the finite field and the polynomials associated with the two elements. Addition and multiplication operators are defined on the finite field so that the sum and multiplication of any two elements of the finite field are elements of the finite field. The attribute of the Reed-Solomon codeword is that the result of multiplying each byte of the codeword by an element of the finite field is another valid Reed-Solomon codeword. Also, adding two Reed-Solomon codewords on a byte basis yields other Reed-Solomon codewords. The old A53 standard defines a 256-element finite field GF (2 ⁸ ) and an associated generator polynomial g (x) for use in the Reed-Solomon algorithm. The properties of finite fields also create the ability to generate syndromes on codewords to determine errors. There are other important attributes of codewords.

In an exemplary embodiment, two codewords or packets generated by a half-rate byte-code encoder provide redundancy for the original codeword and a duplicate of the originally entered codeword Lt; / RTI > The two codewords can also be described as organized and unstructured data. It is important to note that codewords representing the structured and unstructured data can be arranged to form larger data structures. In a preferred embodiment, codewords may be organized to form a rugged data stream of data packets. The robust data stream includes organized packets that are copies of data packets in stream portion A and unstructured packets generated by processing byte-code encoders in stream portion A '. Unstructured packets also include packets that can be derived from other organized and unstructured packets of a robust data stream. Packets in a robust data stream may also be composed of organized bytes and unstructured bytes. In these embodiments, the organizational byte is a copy of the bytes of the content data, and the unstructured byte is derived from other organized and unstructured bytes.

The redundant or unstructured codeword or packet output by the byte-code encoder is the result of multiplying each byte of the incoming codeword or packet by the element b of the finite field GF (256). In one embodiment, the MPEG transmission source 110 generates a message M, which consists of M (1), M (2), ..., M (187) M (2) is the second byte of the message, etc., followed by byte-code encoder 104 generates code words A and A 'from code word M as follows.

(1)

A (i) = M (i) (i = 1, 2, ..., 187)

(2)

A '(i) = b * M (i) (i = 1,2, ..., 187)

The value b is a predetermined (non-zero) element of the same finite field GF 256 that can be used by the Reed-Solomon encoder 155. In an exemplary embodiment, the value of the b element is two. It is clear that using the same finite element for both the byte-code encoder and the reed-solomon encoder enables operations between the two encoders based on the properties of the finite field. The byte-code encoder 125 encodes all of the bytes of the data packet, including the bytes that form the header containing the PID, to generate one or more unstructured packets of the robust data stream. Thus, the PID of each unstructured packet is byte-code encoded and may no longer represent a PID value that can be recognized for the receiving device.

It is clear that any packets encoded by the embodiment of the transmitter depicted by the encoder 100 may be decoded by an embodiment of a decoder used in a sphere receiver conforming to the A53 standard. The decoder in the globally receiver provides packets of the robust data stream to the data decoder. The robust data stream includes unstructured packets that are encoded using a byte-code encoder that will result in data content that will be correctly decoded by the decoder in the old receiver but can not be recognized by the old receiver. However, since these packets have a PID that is not associated with a conventional or a globular data format in the Program Map Table (PMT), the content decoder in the globally receiver ignores these non-functional packets of the robust data stream.

The byte-code encoder 125 generates unstructured packets for each organizational packet using Equation 2 above and provides these two packets to a spherical 8-VSB encoder for transmission to obtain a valid data rate of 1/2 (I.e., 1 byte input, 2 byte output). As noted above, the bytecode encoder 125 may generate different valid data rates using different encoding rates. In some embodiments, the byte-code encoder generates one byte-encoded packet for every two source packets M _A and M _B received from the MPEG TS source 110, A robust data stream having a 2/3 rate including one unstructured packet calculated as shown in FIG.

(3)

_{M AB (i) = M A} (i) * b 1 + M B (i) * b 2 (i = 1, 2, ..., 187)

Where M _A and M _B are successive organized packets generated by the data generator 102 and b ₁ and b ₂ are predetermined elements of a finite field such as a finite field used by the Reed-Solomon encoder 155 . In an exemplary embodiment, the value of the elements b ₁ and b ₂ is two. In some embodiments, the values of b ₁ and b ₂ may not be the same. The byte-code encoder 125 provides M _A , M _B and M _AB to the spherical 8-VSB encoder 130 for further encoding and transmission.

The byte-coder encoder 125 may generate robust data streams (i.e., those with lower data rates) using different coding rates by including additional input data packets for generating redundant packets. Another embodiment of the byte-code encoder 125 is by adopting four unstructured packets M _A , M _B , M _C and M _D from the MPEG TS source 110 and the following calculated five packets Thereby generating a data stream of 4/9 speed.

(4)

_{M AB (i) = M A} (i) * b 1 + M B (i) * b 2 (i = 1, 2, ..., 187)

(5)

M _CD (i) = M _C (i) * b ₃ + M _D (i) * b ₄ (i = 1,2, ..., 187)

(6)

_{M AC (i) = M A} (i) * b 5 + M C (i) * b 6 (i = 1, 2, ..., 187)

(7)

M _BD (i) = M _B (i) * b ₇ + M _D (i) * b ₈ (i = 1,2, ..., 187)

(8)

_{M ABCD (i) = M AB} (i) * b 9 + M CD (i) * b 10 (i = 1, 2, ..., 187)

The values b ₁ , b ₂ , ..., b ₁₀ are predetermined elements selected from a finite field. In an exemplary embodiment, the values for b ₁ , b ₂ , ..., b ₁₀ are two. Also, as shown in equation (8), the packet M _ABCD is a duplicate packet generated from only the other duplicate packets, in particular, from the M _AB and M _CD packets. The duplicate packet M _ABCD can instead be generated using the elements of the duplicate packets M _AC and M _BC . In some embodiments of the MPEG transmission source generator 110, the removal of one or more unstructured packets may be performed with an operation known as puncturing. For example, a punctured rate of 4/8 can be generated by not creating one of the packets (i.e., M _{ABCD in} this case) that will only use duplicate packets, Because it contains data. Any packet or codeword can be removed. However, the removal of a packet or codeword containing the smallest amount of unique data may be optimal. Code puncturing can be used to vary the number of packets transmitted to meet certain constraints on the number of packets or code words being transmitted.

In addition, the byte-code encoder 125 employs eight data packets M _A , M _B , ..., M _H to generate 19 unstructured packets as follows, thereby producing robust data with a data rate of 8/27 A stream can also be generated.

(9)

_{M AB (i) = M A} (i) * b 1 + M B (i) * b 2 (i = 1, 2, ..., 187)

(10)

M _CD (i) = M _C (i) * b ₃ + M _D (i) * b ₄ (i = 1,2, ..., 187)

(11)

_{M AC (i) = M A} (i) * b 5 + M C (i) * b 6 (i = 1, 2, ..., 187)

(12)

M _BD (i) = M _B (i) * b ₇ + M _D (i) * b ₈ (i = 1,2, ..., 187)

(13)

_{M ABCD (i) = M AB} (i) * b 9 + M CD (i) * b 10 (i = 1, 2, ..., 187)

(14)

M _EF (i) = M _E (i) * b ₁₁ + M _F (i) * b ₁₂ (i = 1, 2, ..., 187)

(15)

_{M GH (i) = M G} (i) * b 13 + M H (i) * b 14 (i = 1, 2, ..., 187)

(16)

M _EG (i) = M _E (i) * b ₁₅ M _G (i) * b ₁₆ (i = 1, 2, ..., 187)

(17)

_{M FH (i) = M F} (i) * b 17 + M H (i) * b 18 (i = 1, 2, ..., 187)

(18)

M _EFGH (i) = M _EF (i) * b ₁₉ + M _GH (i) * b ₂₀ (i = 1, 2, ..., 187)

(19)

_{M AE (i) = M A} (i) * b 21 + M E (i) * b 22 (i = 1, 2, ..., 187)

(20)

_{M BF (i) = M B} (i) * b 23 + M F (i) * b 24 (i = 1, 2, ..., 187)

(21)

M _CG (i) = M _C (i) * b ₂₅ + M _G (i) * b ₂₆ (i = 1,2, ..., 187)

(22)

_{M DH (i) = M D} (i) * b 27 + M H (i) * b 28 (i = 1, 2, ..., 187)

(23)

M _ACG (i) = M _AC (i) * b ₂₉ + M _EG (i) * b ₃₀ (i = 1, 2, ..., 187)

(24)

M _BDFH (i) = M _BD (i) * b ₃₁ + M _FH (i) * b ₃₂ (i = 1, 2, ..., 187)

(25)

M _ABF (i) = M _AB (i) * b ₃₃ + M _EF (i) * b ₃₄ (i = 1,2, ..., 187)

(26)

_{M CDGH (i) = M CD} (i) * b 35 + M GH (i) * b 36 (i = 1, 2, ..., 187)

(27)

_{M ABCDEFGH (i) = M ABCD} (i) * b 37 + M EFGH (i) * b 38 (i = 1, 2, ..., 187)

In addition, punctured code with a data rate of 8/26 can be generated by the byte-code encoder 125 by not generating the least unique data value packet M _ABCDEFGH or other packets generated from only the duplicate packets have.

As noted above, the byte-code encoder can be configured to generate predetermined encoding code rates based on the number of codewords or packets used and the number of codewords or packets formed through a single encoding process. In addition, more complex code rates can be constructed using specific arrangements of the above-described code rate encoders as building blocks or component code rate encoders. In addition, additional processing blocks may be included to form a concatenated byte-code encoder. For example, a concatenated byte-code encoder may use additional interleaving blocks between component byte-code encoders in addition to redundancy to improve the ruggedness of the generated data stream. Various embodiments of redundant and code-enhanced staggered transmission methods will be described below.

After encoding, the data is combined with a packet deinterleaver 130. Packet deinterleaver 130 takes the bytes in order of columns from the resulting SCBC codewords for the original group of packets and outputs the bytes in row-wise order. The original packets are reconstructed and new packets are generated from the parity bytes of the SCBC codewords. Each packet corresponds to a common GF (256) symbol position in all generated SCBC codewords. The number of packets generated in each code frame is nSCBC, where the first kSCBC packets are original data packets and the last (nSCBC-kSCBC) packets are parity packets.

The data is then combined into an MPEG TS header modifier 135, where the MPEG headers are modified. The MPEG TS header modifier may modify the packet identifier (PID) of the MPEG transport stream headers to indicate the code rate used by the error correction technique. Code rates are expressed as fractions of the original number of data bytes divided by the total number of data bytes used. For example, in a 12/52 rate mode that replaces twelve data bytes with 40 parity bytes, each group of 12 bytes uses one R = 1/2 encoder and two R = 12/26 encoders, Each 12/26 encoder uses two R = 2/3 encoders and one 27/26 puncture, resulting in a 12/52 speed mode. R = 27/26 punctures are performed in such a way that the last byte of the 27 bytes is dropped. Two data blocks are used to transmit twelve MPEG TS packets under the 12/52 speed mode. The 12/26 rate mode replenishes 12 data bytes with 14 parity bytes, each group of 12 data bytes uses two R = 2/3 encoders and one R = 27/26 puncture, Resulting in a 12/26 speed mode. R = 27/26 punctures will be performed in such a way that the last byte of the 27 bytes is dropped. One data block is used to transmit twelve MPEG TS packets under the 12/26 rate mode. The 17/26 rate mode replenishes 17 data bytes with 9 parity bytes and each group of 17 data bytes replaces 16 data bytes with 8 parity bytes using one R = 2/3 encoders And one R = 1/2 encoder is used to supplement one data byte with one parity byte, resulting in a 17/26 speed mode. One data block is used to transmit 17 MPEG TS packets under 17/26 speed mode. The 24/208 rate mode replenishes 24 data bytes with 184 parity bytes and each group of 24 data bytes uses 24 R = 1/4 encoders and 8 R = 12/26 encoders, As well as a 24/208 speed mode. R = 27/26 punctures will be performed so that the last of the 27 bytes is dropped. Eight data blocks are used to transmit 24 MPEG TS packets under the 24/208 speed mode.

Each packet using the MPEG protocol typically includes a packet identification portion or a PID. The current system allows more than 8000 possible unique identifiers, but now only 50 are used. The PID is typically one or more bytes of information used to identify the type of data in the packet. Many of the PID portions of the bits now remain reserved and unused. These PIDs may be used to identify a particular error correcting code rate to be assigned to the packet. In order to ensure that the PID is correctly identified by any receiving system, certain rules based on the MPEG protocol must be maintained. The 3 byte header 440 includes a 13 bit packet identifier (PID) that identifies the packet as part of the mobile / handheld transmission. Headers 440 of MPEG packets from the ATSC-M / H stream are modified to include packet identifiers (PIDs) that are not recognized by the older ATSC A / 53 receivers after packet-deinterleaving . Therefore, older receivers should ignore ATSC-M / H specific data to provide backward compatibility.

This data is then coupled to a preamble packet inserter 140, where preamble packets comprised of consecutive MPEG packets form a preamble block. The MPEG packets are formed to have an effective MPEG header with data bytes generated from a PN generator (not shown). The number of data bytes generated from the PN generator depends on the code rate used, e.g., 184 data bytes are generated in the 12/52 rate mode, resulting in a total of 2208 bytes of PN data. According to an exemplary embodiment, the PN generator is a 16-bit shift register with nine feedback taps. Eight of the shift register outputs are selected as output bytes. The ATSC M / H packets are placed between the preamble blocks in the data blocks. All data blocks contain 26 ATSC M / H encoded packets, which have the same coding or 26 ATSC A / 53 encoded packets. Once the preamble packets are inserted 140, an ATSC M / H stream is formed.

The ATSC-M / H data stream is then sent to a data randomizer 150, a Reed-Solomon encoder 155, a byte interleaver 160, a 12-1 trellis encoder 165, a sink inserter 170, 53 path 145 that includes a modulator 180 and a modulated ATSC A / In data randomizer 150, each byte value is changed according to a known pseudo random number generation pattern. This process is reversed at the receiver to recover the correct data values. With the exception of segment and field sinks, in order to use the allocated channel space with maximum efficiency, it is necessary to make it possible for the transmitted signal frequency response to have a flat noise-like spectrum It is desirable that the 8-VSB bit stream has completely random noise-like properties.

The data is then coupled to a Reed-Solomon encoder 155, where Reed-Solomon (RS) coding provides additional error correction possibilities at the receiver through the addition of additional data to the transmitted stream. In an exemplary embodiment, the RS code used in the VSB transmission system is t = 10 (207, 187). The RS data block size is 187 bytes, and 20 RS parity bytes are added for error correction. The total RS block size of 207 bytes is transmitted in RS code word units. In generating the bytes from the serial bit stream, the MSB will be the first serial bit, and the twenty RS parity bytes are sent to the end of the data block or RS codeword.

The byte interleaver 160 then processes the output of the Reed-Solomon encoder 155. Interleaving is a common technique for handling burst errors that may occur during transmission. Without interleaving, a burst error can have a significant impact on one particular segment of data, thereby making the segment impossible to recalibrate. However, if the data is interleaved before transmission, the effect of the burst error can be effectively distributed across the plurality of data segments. Instead of introducing large errors into one local segment that can not be modified, smaller errors may be introduced into a plurality of segments each separately within a correction capability range of forward error correction, parity bits, or other data integrity techniques . For example, a typical (255, 223) Reed-Solomon code will allow correction of up to 16 symbol errors in each codeword. If the Reed-Solomon coded data is interleaved prior to transmission, the probability that a long error burst will be spread across a plurality of codewords after inverse interleaving is increased and the possibility that more errors than a correctable 16 symbol errors are present in any particular codeword .

The interleaver used in the VSB transmission system is a 52 data segment (inter segment) convolutional byte interleaver. Interleaving is provided at a depth of about 1/6 of the data field (4 ms depth). Only data bytes are interleaved. The interleaver is synchronized to the first data byte of the data field. Intrasegment interleaving is also performed for the benefit of the lattice coding process.

The signal is then coupled to a trellis encoder 165. Grid coding is another form of forward error correction. Unlike Reed-Solomon coding, which treats the entire MPEG-2 packet as a block at the same time, lattice coding is an evolutionary code that keeps track of the ongoing streams of bits as they evolve over time. Thus, Reed-Solomon coding is known in the form of block codes, while lattice coding is a convolutional code.

In ATSC trellis coding, each 8 bit byte is divided into a stream of four 2 bit words. In the lattice coder, each arriving 2-bit word is compared to the past history of the previous 2-bit words. A 3 bit binary code is mathematically generated to describe the transition from the previous 2 bit word to the current one. These 3-bit codes replace the original 2-bit words and are transmitted to the airwaves as 8-level symbols of 8-VSB (3 bits = 8 combinations or levels). There are three bits for every two bits going into the trellis coder. For this reason, the lattice coder in the 8-VSB system is referred to as a 2/3 rate coder. The signal waveform used with the lattice code is an eight level (three bit) one-dimensional constellation. The transmitted signal is referred to as 8 VSB. 4 state grating encoder will be used.

In an exemplary embodiment, lattice code intra segment interleaving is used. It uses twelve identical lattice encoders and precoders that operate on interleaved data symbols. Code interleaving encodes symbols (0, 12, 24, 36, ...) as a group, encodes symbols (1, 13, 25, 37, ...) as a second group, (2, 14, 26, 38, ...) as a third group, and so on.

Once the data is grid encoded, it is coupled to the sink inserter 170. The sync inserter 170 is a multiplexer that inserts various synchronization signals (data segment sync and data field sync). A two level (binary) four symbol data segment sync is inserted into the eight level digital data stream at the beginning of each data segment. The MPEG sync byte is replaced by a data segment sync. In an exemplary embodiment using ATSC transmission standards, the complete segment will consist of 832 symbols, of which 4 symbols are for data segment sync, and 828 are data and parity symbols. The same sync pattern occurs regularly at 77.3 second intervals, which is the only signal that repeats at this rate. Unlike the data, the four symbols for the data segment sync are neither Reed-Solomon or lattice-encoded nor interleaved. The ATSC segment sync is an iterative 4-symbol (1-byte) pulse that is added to the front of the data segment and replaces the missing first byte (Packet Sync Byte) of the original MPEG-2 data packet. Correlation circuits within the 8-VSB receiver go towards the iterative nature of the segment sync, which is easily contrasted with the background of completely random data. The recovered sync signal is used to generate the receiver clock and recover the data. Segment syncs can be easily recovered by the receiver due to their repetitive nature and long duration. Accurate clock recovery can be achieved even at noise and interference levels beyond levels where accurate data recovery is not possible, which enables fast data recovery during channel changes and other transient conditions.

After sink insertion, the signal is combined with a pilot inserter, where a small DC shift is applied to the 8-VSB baseband signal so that a small residual carrier appears at the zero frequency point of the resulting modulated spectrum . This ATSC pilot signal provides the RF PLL circuits in the 8-VSB receiver with signals independent of the data being transmitted, which they will lock onto. The frequency of the pilot is the same as the suppressed carrier frequency. This can be generated by a small (digital) DC level (1.25) added to all the symbols (data and sink) of the digital baseband data and the sync signals (+1, +3, +5, +7). The power of the pilot is typically 11.3 dB below the average data signal power.

After the pilot signal is inserted, the data is coupled to a modulator 180. The modulator amplitude modulates the 8-VSB baseband signal on an intermediate frequency (IF) carrier. With conventional amplitude modulation, we create a double sideband RF spectrum around our carrier frequency, and each RF sideband is on the other side of the mirror. This represents duplicate information, and one sideband can be discarded without any substantial loss of information. In 8-VSB modulation, the VSB modulator receives a 10.76 mega symbol per second 8-level lattice-encoded composite data signal (with pilot and sink added). The ATV system performance is based on a linear phase raised cosine Nyquist filter response at the transmitter and receiver as shown in FIG. System filter responses are essentially flat across the entire band, except in transition regions at each end of the band. Nominally, the roll-off in the transmitter will have the response of a linear phase root raised cosine filter.

The transmission system includes operations for mobile and portable devices in the burst mode of transmission. Several key advantages in operation in burst mode have been described throughout the foregoing disclosure and include the ability to be received by a new class of devices while maintaining backward compatibility. These new class of devices require a lower level of video resolution than that found in existing broadcast standards and thus also enable other features, including higher coding and compression, as well as operating in the presence of higher noise levels can do. An additional advantage of burst mode operation types is focused on potential device power savings by focusing on using the device only when signals are intended or received for the device.

Burst mode operations such as those described above may utilize periods that do not require high data transmission of the signal to maintain maximum performance of the legacy system and receiver. The burst mode operation may be based on processing of signals based on the so-called new information processing rate which may vary according to current broadcast signal characteristics.

Backward compatibility with older systems is maintained by focusing on burst mode operations at a data packed level by introducing information about new program identifiers. The new program identifiers allow a new class of equipment to recognize the data without affecting the operation of the existing equipment. There is support for additional spheres by including an overlay structure to maintain the spherical signal transmission operation during certain burst mode profiles.

Further utilization of frequency or spatial diversity as well as temporal diversity, along with the complete redundancy of organized and unstructured data and the ability of the signal to be recovered from one of the totally separable codewords, Can be increased. This will be further discussed in the discussion of FIG.

2, a block diagram of one embodiment of a portion of an exemplary mobile / handheld data stream 200 of the present disclosure is shown. 26 ATSC M / H coded packets are grouped into one data block. In a global ATSC transmission, all data blocks typically have the same coding, but this is not physically required. The preamble blocks are two blocks long and have 52 ATSC M / H coded packets. The first MPEG packet following the preamble block is a control packet containing system information. After randomization and forward error correction processing, data packets are formatted into data frames for transmission, and data segment sync and data field sync are added.

The ATSC-M / H data stream 200 comprises bursts having a predetermined number of data blocks 230 suitable for a preamble block 210 and a subsequent selected data rate mode. According to an exemplary embodiment, each data block 230 consists of 26 MPEG packets. Each data frame consists of two data fields, each containing 313 data segments. The first data segment of each data field is a unique synchronization signal (data field sync) and includes a training sequence used by an equalizer in the receiver. The remaining 312 data segments carry the equivalent of data from one 188 byte transport packet and its associated FEC overhead, respectively. The actual data in each data segment comes from some transport packets due to data interleaving. Each data segment consists of 832 symbols. The first four symbols are transmitted in binary form and provide segment synchronization. This data segment sync signal also represents the sync byte of the 188 byte MPEG compatible transport packet. The remaining 828 symbols in each data segment carry the remaining 187 bytes of the transport packet and the data equivalent for its associated FEC overhead. These 828 symbols are transmitted as 8 level signals and therefore carry 3 bits per symbol. Thus, 828x3 = 2484 bits of data are carried within each data segment, which is a requirement for transmitting protected transport packets.

The ATSC M / H data stream consists of a sequence of blocks, each consisting of 26 packets from one of the old VSB A / 53 system or the ATSC M / H system. The ATSC M / H data stream consists of bursts of blocks, each burst having a preamble block followed by Nb data blocks, where Nb is a system variable parameter and is a function of the total ATSC M / H data rate to be transmitted . Each data block is encoded in one of the defined ATSC M / H rate modes. This rate mode applies to the entire data block. For each burst of blocks, the highest coded FEC rates (i.e., the lowest fractions) in the burst of blocks are delivered first and the lowest coded FEC rates (i.e., the highest fractions) So that data blocks are transmitted starting from the preamble block such that any subsequent data blocks have a robustness equal to or less than the current data block. ATSC A / 53 8-VSB coded 26 packets of rectangular data blocks may be placed in one or more blocks for spherical overlay operation.

Referring now to FIG. 3, a data frame 300 in accordance with the present invention is shown. The illustrated data frame 300 is organized for transmission, and each data frame consists of two data fields each containing 313 data segments. The first data segment of each data field is a unique synchronization signal (data field sync) and includes the training sequence used by the equalizer in the receiver. The remaining 312 data segments carry the equivalent of data from one 188 byte transmission packet and its associated FEC overhead, respectively. The actual data in each data segment comes from some transport packets due to data interleaving. Each data segment consists of 832 symbols. The first four symbols are transmitted in binary form and provide segment synchronization. This data segment sync signal also represents the sync byte of the 188 byte MPEG compatible transport packet. The remaining 828 symbols of each data segment carries the data equivalent for the remaining 187 bytes of the transport packet and its associated FEC overhead. These 828 symbols are transmitted as 8 level signals and thus carry 3 bits per symbol. Thus, 828x3 = 2484 bits of data are carried within each data segment, which is a requirement for transmitting protected transport packets.

187 data bytes + 20 RS parity bytes = 207 bytes

207 bytes x 8 bits / byte = 1656 bits

The 2/3 rate lattice coding requires 3/2 x 1656 bits = 2484 bits.

The exact symbol rate is given by Equation 1 below.

(1) S _r (MHz) = 4.5 / 286 x 684 = 10.76 ... MHz

The frequency of the data segment is given by Equation 2 below.

(2) f _seg = S _r /832=12.94...X 10 ³ data segment / sec

The data frame rate is given by Equation 3 below.

(3) f _frame = f _seg /626=20.66 ... frames / second

The symbol rate S _r and the transmission rate T _r will have their frequencies fixed to each other.

The 8 level symbols combined with binary data segment sync and data field sync signals are used to suppressed-carrier modulate a single carrier. However, before transmission, most of the lower sidebands will be removed. The resulting spectrum is flat, except in band edges where the nominal square root raised cosine response yields 620 kHz transition regions. At the suppressed carrier frequency of 310 kHz from the lower band edge, a small pilot is added to the signal as described above.

Referring now to FIG. 4, an embodiment of a terrestrial broadcast receiver 400 for mobile / handheld reception of the present disclosure is shown. The receiver 400 includes a signal receiving element 410, a first tuner 420, a second tuner 425, a first pre-equalizer demodulator 430, a second demodulator demodulator 430, An equalizer 450, a post-equalizer correction processor 460, a transmission decoder 470, and a tuner controller 480. The controller 440 may be any of the following:

The signal receiving component 410 is operable to receive signals comprising audio, video and / or data signals (e.g., television signals, etc.) from one or more signal sources, such as terrestrial broadcast systems and / do. According to an exemplary embodiment, the signal receiving element 410 is implemented as an antenna, such as a log periodic antenna, but may also be implemented as any type of signal receiving element. The antenna 410 in this exemplary embodiment operates to receive audio, video, and data signals that are transmitted over ATSC M / H terrestrial over frequency bandwidth. ATSC signals are generally transmitted over a frequency range of 54 to 870 MHz with a bandwidth of about 6 MHz per channel. The subchannels may be time multiplexed. The signals are coupled from the antenna through a coaxial cable or a transmission line such as a printed circuit board trace.

The first and second tuners 420 and 425 operate to perform a signal tuning function responsive to the control signal from the tuner controller 480. In accordance with an exemplary embodiment, each tuner 420, 425 receives a different time or frequency diverse RF signal from one of the one or more antennas 410, filters and frequency downconverts the RF signal, (I.e., single-ended or multi-stage down-converted) to generate an intermediate frequency (IF) signal. The RF and IF signals may include audio, video and / or data content (e.g., television signals, etc.) and may include analog signal standards (e.g., NTSC, PAL, SECAM, etc.) and / or digital signal standards (e.g., ATSC, QAM, QPSK, etc.). Each tuner 420 is operative to convert a received ATSC M / H signal from a carrier frequency to an intermediate frequency. For example, the tuner may convert the 57 MHz signal received at antenna 410 to a 43 MHz IF signal. A light demodulator demodulator 430 is operative to demodulate the IF signal from the tuner 420 into a baseband digital stream. Demodulator 435 is operative to demodulate the IF signal from tuner 425. The baseband digital streams are then coupled to an equalizer.

The tuner controller 480 operates to receive commands from the transport decoder 470 in response to the signal level and frequency of the tuned channel or the desired tuned channel. The tuner controller 480 generates control signals for controlling the operation of the tuners 420 and 425 in response to these received commands.

The equalizer controller 440 generates an error term in response to the decoded data received from the demodulators 430 and 435. This provides the capability for a data-oriented equalizer. The equalizer controller 440 estimates the error between the received data and the decoded data and generates an error term. The error term is supplied to the equalizer 450 and minimized.

The equalizer 450 is operative to receive the tuned and demodulated MPEG streams from the light modulator demodulators 430 and 435 and calculates the equalizer coefficients applied to the equalization filter in the equalizer to generate the error signal. The equalizer 450 operates to compensate for transmission errors such as attenuation and intersymbol interference. The equalizer includes a matched filter that performs roll-off filtering that operates to cancel the intersymbol interference. During the equalizer training period, the previously selected training signal is transmitted over the channel and a correctly delayed version of the signal previously stored in the receiver is used as the reference signal. The training signal is usually a pseudo noise sequence long enough to allow the equalizer to compensate for channel distortion. An equalizer in accordance with an exemplary embodiment of the present invention is operative to store a plurality of pseudo noise sequences, each pseudo-random sequence corresponding to a code rate. When the equalizer 450 receives the pseudo-random sequence training signal, the equalizer compares a portion of the received sequence with a plurality of stored sequences. If a match is made, the code rate associated with the received sequence is used by the decoder to decode the received data after the training sequence.

The downstream corrector processor 460 and the transport decoder 470 are operative to perform error correction and to decode the MPEG data stream. These elements are shown and discussed in detail in Figures 5 and 6.

The receiver can be configured to operate with a single tuner and demodulator by time sharing the tuner and receiving different frequencies and different times. Instead, the tuner is configured with a bandwidth wide enough to simultaneously receive two signals so that both signals can be tuned to different IF frequencies and each of these IF frequencies can be processed simultaneously or time multiplexed by a demodulator . In a single tuner, using either time or frequency diversity, packet combinations do not occur in the equalizer but instead occur in the code because the equalizer must follow the received signals. This gives three possibilities to receive the packet correctly: the probability of correctly receiving the first packet, correctly receiving the second packet, or correctly receiving the combination after the byte decoder. In contrast to receiving a single packet, if coding is being used to combine packets, this reduces the minimum amount of signal-to-noise ratio needed to receive a signal that is substantially error-free. For example, at a code rate of 1/2, the minimum threshold is reduced from 15 dB for a single packet without coding to 7 dB for two packets with coding and 3.5 dB for four packets with coding.

Referring now to FIG. 5, a block diagram of one embodiment of a decoder 500 for use in a receiver system is shown. Decoder 500 includes circuitry adapted to assist in decoding data received by the receiver using redundant packets, such as unstructured packets in the data stream described above. The decoder 500 may also decode the encoded data, generally using the older or existing A53 standard.

At the decoder 500, after the initial tuning, demodulation and processing by other circuits (FIG. 4), the grating decoder 502 receives the incoming signal. The lattice decoder 502 is connected to a convolutional de-interleaver 504. The output of the convolutional de-interleaver 504 is connected to a byte-code decoder 506. The byte-code decoder 506 has an output connected to the Reed-Solomon decoder 508. The output of the Reed-Solomon decoder 508 is connected to a de-randomizer 510. The output of the inverse randomizer 510 is connected to a data decoder 512. Data decoder 512 provides an output signal for use in the remainder of the receiver system, such as video display or audio playback.

According to the conventional or older A53 standard, the lattice decoder 502 includes a signal demultiplexer, 12 2/3 rate lattice decoders, and a signal multiplexer. The demultiplexer distributes digital samples between twelve 2/3 speed lattice decoders and the multiplexer multiplexes the estimates generated by each of the twelve 2/3 speed lattice decoders. An inverse interleaver 504, such as a convolutional interleaver, deinterleaves the stream of lattice decoded bit estimates to generate sequences or packets arranged to include 207 bytes. The packet arrangement is performed in conjunction with determining and identifying the location of synchronization signals that are not shown. The Reed-Solomon error correction circuit 508 considers each of the 207-byte sequences generated by the de-interleaver 504 as one or more code words, and any bytes in the codewords or packets may be corrupted Or not. This determination is often performed by calculating and evaluating a set of syndromes or error patterns for codewords. If a corruption is detected, the Reed-Solomon error correction circuit 508 attempts to recover the corrupted bytes using the information encoded in the parity bytes. The resulting error corrected data stream is then de-randomized by the de-randomizer 510 and then provided to a data decoder 512 that decodes the data stream according to the type of content being transmitted. Typically, the combination of lattice decoder 502, deinterleaver 504, Reed-Solomon decoder 508 and inverse randomizer 510 is identified as an 8-VSB decoder in the receiver. It is important to note that, in general, a conventional receiver for receiving signals conforming to the older A53 standard performs the receiving process in the reverse order of the transmitting process.

The received data in the form of bytes of data in the data packets are decoded by the trellis decoder 502 and deinterleaved by the deinterleaver 504. The data packets may contain 207 bytes of data and may also be grouped into groups or 24, 26, or 52 packets. The lattice decoder 502 and the de-interleaver 504 can process the byte-encoded data as well as the incoming rectangle format data. Based on a predetermined packet transmission sequence also known by the receiver, the byte-code decoder 506 determines whether the packet is a byte-code encoded or packet contained within a robust data stream. If the received packet is not from a byte-code encoded data stream, the received packet is provided to the Reed-Solomon decoder 508 without any further processing in the byte-code decoder 506. [ The byte-code decoder 506 may also include an inverse randomizer that multiplies the data stream during encoding or removes a known sequence of added constants. It is important to note that robust data streams include both the same bytes as the organizational packets and the original data, and both bytes that contain unstructured packets and redundant data.

If the byte-code decoder 506 determines that the received packet is a byte-code encoded packet belonging to a robust or robust data stream, the packet may be decoded along with other packets including the same data stream. In one embodiment, byte-code encoded packets of the same data stream are decoded by multiplying the inverse of the value of the element that was used to develop the byte-coded packet with each byte in the packet. The decoded values of the bytes of the unstructured packet are compared to the values of the bytes of the organizational packet and the values of any bytes in the two non-identical packets are deleted (i. E., Set to 0) Lt; / RTI > A systematic packet from which error bytes have been deleted can be decoded using Reed-Solomon decoding performed in the Reed-Solomon decoder 508 thereafter. A further description of other embodiments of byte-code decoders will be discussed below.

The bytecode decoder 506 may also be adapted to operate as a block coder for decoding the encoded signals as shown in Fig. For example, the bytecode decoder 506 may include a packet deinterleaver similar to the packet interleaver 120 and a packet de-interleaver similar to the packet de-interleaver 130. In addition, the bytecode encoder function may be adapted to decode the GF (256) serial concatenated block coding (SCBC) signal. The bytecode decoder 506 may further comprise an identifier block used to identify encoded data for identification of mobile or ATSC M / H receive and / or a priori training packets. The identifier block may also include a packet identifier block, for example, to determine whether headers in incoming packets include a PID used for mobile reception.

It is important to note that in the preferred encoder the byte-code encoding precedes the Reed-Solomon encoding of the data packets. However, in the decoder 500 shown here, the incoming data is byte-code decoded before being Reed-Solomon decoded. The byte-code and the Reed-Solomon code operations are linear on the finite field 256 used in the A53 standard, and the linear operators are permutations because they are infinitesimal in the finite field. It is advantageous to perform block decoding first before Reed Solomon because there are soft decoding algorithms that make it practical to have an iterative decoding algorithm. Byte-code encoding provides a soft decoding algorithm, which allows iterative decoding or turbo decoding with higher reliability to recover errors in the received signal, so reordering is important. As a result, performing byte-code decoding prior to Reed-Solomon decoding results in an improvement in receiver performance as measured in terms of bit error rate and signal-to-noise ratio.

Referring now to FIG. 6, a block diagram of another embodiment of a decoder 600 for use in a receiver is shown. Decoder 600 includes additional circuitry and processing for receiving and decoding signals that are adversely affected by signal transmission on a transmission medium such as airborne electromagnetic waves. The decoder 600 is capable of decoding both robust data streams and spherical data streams.

At the decoder 600, a signal that follows the original processing is provided to the equalizer 606. An equalizer 606 is connected to the trellis decoder 610, which provides two outputs. The first output from the trellis decoder 610 provides feedback and is connected back to the equalizer 606 as a feedback input. The second output from the trellis decoder 610 is connected to the convolutional deinterleaver 614. Convolution deinterleaver 614 is connected to byte-code decoder 616, which also provides two outputs. The first output from the byte-code decoder 616 is reconnected to the trellis decoder 610 as a feedback input via the convolutional interleaver 618. The second output from the byte-code decoder 616 is connected to the Reed-Solomon decoder 620. The output of the Reed-Solomon decoder 620 is connected to an inverse randomizer 624. The output of the inverse randomizer 624 is connected to a data decoder 626. The Reed-Solomon decoder 620, the inverse randomizer 624 and the data decoder 626 are connected and functioning in a manner similar to the Reed-Solomon, inverse randomizer and data decoder blocks described in Figure 5, It will not be described further.

An input signal from a front end processing (e.g., antenna, tuner, demodulator, A / D converter) of a receiver (not shown) is provided to an equalizer 606. The equalizer 606 processes the received signal to completely or partially remove the transmit channel effect in attempting to recover the received signal. Various removal or equalization methods are well known to those skilled in the art and will not be discussed herein. The equalizer 606 may include a plurality of sections of processing circuitry including a Feed-Forward Equalizer (FFE) section and a Decision-Feedback-Equalizer (DFE) section.

An equalized signal is provided to the lattice decoder 610. The lattice decoder 610 generates a set of decision values provided in the DFE section of the equalizer 606 as one output. The lattice decoder 610 may also generate intermediate decision values that are also provided in the DFE section of the equalizer 606. The DFE section adjusts the values of the filter taps in the equalizer 606 using the decision values together with the intermediate decision values from the lattice decoder 610. The adjusted filter tap values remove interference and signal reflections present in the received signal. The iterative process allows the equalizer 606 to dynamically adapt to potentially changing signal transmission environment conditions over time with the help of feedback from the grating decoder 610. [ It is important to note that a repetitive process can occur at a rate similar to the incoming data rate of a signal such as 19 Mb / s for a digital television broadcast signal. The iterative process can also occur at a higher rate than the incoming data rate.

The lattice decoder 610 also provides a lattice decoded data stream to the convolution deinterleaver 614. Convolution deinterleaver 614 operates similarly to the deinterleaver described in FIG. 5 and generates deinterleaved bytes organized in data packets. The data packets are provided to the byte-code decoder 616. As described above, packets that are not part of the robust data stream are simply passed to the Reed-Solomon decoder 620 via the byte-code decoder 616. When the byte-code decoder 616 identifies a group of packets as part of the robust data stream, the byte-code decoder 616 uses the redundancy information in the unstructured packets to initialize the bytes in the packets as described above Decode.

The byte-code decoder 616 and the lattice decoder 610 operate in a repetitive manner, referred to as a turbo decoder, to decode the robust data stream. In particular, the lattice decoder 610 provides a first soft decision vector for each byte of packets contained in the robust data stream to the byte-code decoder 616 after inverse interleaving by the convolutional interleaver 614 do. Typically, the lattice decoder 610 generates a soft decision as a vector of probability values. In some embodiments, each probability value in a vector is associated with a value that a byte associated with the vector may have. In other embodiments, the vector of probability values is generated for every half-nibble (i.e., two bits) contained in the organizational packet, which allows the 2/3 rate lattice decoder to estimate 2-bit symbols Because. In some embodiments, the lattice decoder 610 combines the four soft decisions associated with the four half nibbles of the bytes to produce one soft decision, which is a vector of the probabilities of the values that the byte may have. In these embodiments, soft decisions corresponding to the byte are provided to the byte-code decoder 616. In other embodiments, the byte-code decoder separates the soft decisions for the bytes of the organized packet into four soft decision vectors, where each of the four soft decisions is associated with a half nibble of the byte.

The byte-code decoder 616 generates a first estimate of the bytes containing the packets using the soft decision vector associated with the bytes containing the packets of the robust data stream. The byte-code decoder 616 uses both the organized and unstructured packets to generate a second soft decision vector for each byte of packets containing the robust stream, and after re-interleaving by the convolutional interleaver 618, And provides a soft decision vector to the lattice decoder 610. The lattice decoder 610 then generates an additional iteration of the first soft decision vector using the second soft decision vector, which is provided to the byte-code decoder 616. The lattice decoder 610 and the byte-code decoder 616 repeat in this manner until the soft decision vectors generated by the lattice decoder and byte decoder converge or a predetermined number of iterations are performed. The byte-code decoder 616 then uses the probability values in the soft decision vector for each byte of the organized packets to generate a hard decision for each byte of the organized packets. The hard decision values (i.e., decoded bytes) are output from the byte-code decoder 616 to the Reed-Solomon decoder 620. The lattice decoder 610 may be implemented using a Maximum a Posteriori (MAP) decoder and may operate on either byte or half nibble (symbol) soft decisions.

It is important to note that turbo decoding typically uses repetition rates associated with carrying decision data between blocks, higher than incoming data rates. The number of possible iterations is limited by the ratio of data rate to repetition rate. As a result, to the extent practical, the higher repetition rate in the turbo decoder generally improves the error correction result. In one embodiment, a repetition rate of eight times the incoming data rate may be used.

The soft input soft output byte-code encoder as described in FIG. 6 may include vector decoding functions. Vector decoding involves grouping bytes of data, including organized and unstructured bytes. For example, for a byte-code encoded stream at half speed, one organized byte and one unstructured byte will be grouped. Two bytes have more than 64,000 possible values. The vector decoder determines or estimates the probability for each of the possible values of the two bytes and generates a probability map. A soft decision is made based on weighting the probabilities of some or all of the probabilities and based on the Euclidean distance for possible codewords. If the error of the Euclidean distance falls below the threshold value, a hard decision can be made.

As described in Figures 5 and 6, byte-code decoders can decode robust data streams encoded by the above-described byte-code encoders, including encodings by simple byte-code encoders or concatenated byte-code encoders have. The byte-code encoders of Figures 5 and 6 describe decoding a robust data stream encoded by a simple or component byte-code encoder that involves only a single encoding step. Concatenated byte-code decoding involves decoding codewords or bytes coming in at least two decoding stages in addition to intermediate processing such as deinterleaving, deinterlacing and re-insertion.

Referring to FIG. 7, an exemplary broadcast environment 700 in accordance with the present invention is shown. A first transmitter 720, a second transmitter 720, and a mobile receiver 730 are shown. The first transmitter 710 is located at a distance d1 from the mobile receiver 730 and the second transmitter 720 is located at a distance d2 from the mobile receiver 730. [

Utilizing detachable and redundant codewords, a first codeword may be transmitted from the first transmitter 710 and a second codeword may be transmitted from the second transmitter 720. [ This reduces the occurrence of overall signal loss by changing the propagation paths and angles. This change reduces the probability of total signal blanks or a completely destructive multipath. Signal reception can be further improved by transmitting each codeword at different frequencies and / or different times.

This embodiment of spatial and frequency diversity can utilize the original "white space" between existing broadcast channels in the coverage area without increasing the equalizer complexity of current receivers. These proposed embodiments are also particularly well suited to (but not limited to) burst mode transmissions, which are currently being considered for advanced broadcast transmission systems. In burst mode transmissions, a single tuner in the receiver can still receive complete transmissions by receiving asynchronous content at multiple frequencies. Complete reception can be achieved by receiving only one of two or more bursts from two or more sources including primary and secondary transmitters. Signal synchronization can be maintained through a number of known techniques, including techniques already in use for SFNs.

Referring to FIG. 8, an exemplary embodiment of a portion of a transmitter 800 in accordance with the present invention is shown. This figure illustrates an arrangement for a particular implementation of a code enhanced deeply interleaved staggercasting structure within the physical layer of a signal transmission system. Implementations within the receiver will yield similar arrangements rearranged for decoding and demodulation as opposed to encoding and modulation. In operation, processing entails identifying and generating information for redundancy or staggercast operations. This content is received via the channel output 805. Next, the content is provided to a coder comprising two or more parallel encoding branches suitable for nominally generating a standard coded burst mode signal. Next, each coding branch processes its supplied signal portion. Next, one branch is delayed by the delay RAM 815 by a predetermined amount. The amount of delay may represent the number of cycles of the signal, such as a field sync signal, and may also be delayed to an equivalent time representation of a future or subsequent burst transmission time. Each of the signals on each branch is encoded at their respective input terminals 810, 820. The inputs may include spherical deinterlacing and packet deinterlacing. For time diversity signals, the non-delayed coding branch signal may be combined 825 with a previously coded and delayed branch signal that includes a portion of the previously processed data content, . The combined data is then decoded (830) and separated (840) into different output stages. Each output stage 845, 850 can combine spherical interleaving and spherical encoding. The processes for the receiver may be substantially reversed and substantially similar to the processes for the transmitter.

For spatial diversity or frequency diverse transmissions, the content is combined from the channel input before or after the delay RAM 815, and is optionally coupled to another delay RAM 855. The content may then be combined into additional channels for transmission comprising operations as shown for FIG. One advantage of space and frequency diversity transmitters is that they use cooperative transmission. In this example, two television broadcasters or one transmitter with two transmitters or frequencies can transmit a cooperative broadcast duplicate packet with a first broadcast packet on one transmitter. Thus, each broadcaster transmits two packets (one burst and one redundant burst), but by simply sending a redundant burst of cooperative broadcasters, these duplicate packets are transmitted on different frequencies and / or different transmitters, . The cooperative transmission gives the advantages of full frequency diversity and possibly space diversity without increasing the data output or bandwidth of each cooperative transmitter.

In addition to the inherent advantages associated with the relationship of meaningful time interleaving of data content and operation in burst mode transmission, each branch of the modified code enhanced data can be transmitted on separate frequencies. In this manner, frequency diversity in addition to time diversity can be achieved. For example, a first burst comprising a portion of the staggercast content may be provided or transmitted on a broadcast channel specified by the first broadcaster after code improvement. A second burst containing the remainder of the staggercast content may be provided or transmitted on a second broadcast channel at a later time, possibly after a code improvement, by a second broadcaster. Each broadcast channel represents a different frequency spectrum of operation. The resulting operation further assures recovery of the original data content by introducing a frequency diversity operation into the existing time diversity system.

While the invention has been described with reference to particular embodiments, it will be understood that modifications may be made which will fall within the scope of the invention. For example, the various processing steps may be implemented separately or in combination, and may be implemented in general purpose or dedicated data processing hardware. In addition, various encoding or compression methods may be used for video, audio, image, text, or other types of data. In addition, packet sizes, rate modes, block coding, and other information processing parameters may be changed in different embodiments of the present invention.

Claims (14)

CLAIMS 1. A method for encoding data,
Encoding the data in an encoder to generate a redundant packet that is a packet that is a systematic packet and a packet that is non-systematic, each packet including the data, Wherein the unstructured packet comprises a new codeword that provides redundancy to the input codeword; And
Coupling said packet and said duplicate packet to a transmitter at an interface
/ RTI > The method according to claim 1,
The packet and the redundancy packet being coupled to different transmitters,
Wherein the different transmitters are spatially diverse. The method according to claim 1,
Wherein the packet and the redundant packet are transmitted at different frequencies. The method according to claim 1,
Wherein the packet and the duplicate packet are transmitted at different times. CLAIMS 1. A method for receiving a signal,
Receiving a first packet that is an organizational packet at an interface, the organizational packet including a copy of an input codeword;
Receiving a second packet that is a non-textured packet at the interface, the second packet being a duplicate version of the first packet, the unstructured packet including a new codeword providing redundancy to the input codeword;
Combining the first and second packets at a processor; And
Decoding the first and second packets at a decoder
/ RTI > 6. The method of claim 5,
Wherein the first packet is received at a first frequency and the second packet is received at a second frequency. 6. The method of claim 5,
Wherein the first packet is received at a first time and the second packet is received at a second time. An encoder for encoding data to generate duplicate packets, which are packets that are organized packets and unspecified packets, the organized packets comprising a copy of an input codeword, the unstructured packet comprising a new codeword that provides redundancy to the input codeword Included -; And
An interface for coupling the packet to a first transmitter and coupling the duplicate packet to a second transmitter,
/ RTI > 9. The method of claim 8,
Wherein the packet is transmitted at a first frequency and the duplicate packet is transmitted at a second frequency. 9. The method of claim 8,
Wherein the packet is transmitted at a first time, and the duplicate packet is transmitted at a second time. An interface for receiving a first packet that is a structured packet and a second packet that is a non-textured packet, the structured packet including a copy of an input codeword, the unstructured packet including a new codeword that provides redundancy to the input codeword -;
A decoder for decoding the first packet and the second packet to generate a first decoded packet and a second decoded packet; And
A processor for combining the first decoded packet and the second decoded packet to generate a video signal,
/ RTI > 12. The method of claim 11,
Wherein the second packet is a duplicate version of the first packet. 12. The method of claim 11,
Wherein the first packet is received at a first frequency and the second packet is received at a second frequency. 12. The method of claim 11,
Wherein the first packet is received at a first time and the second packet is received at a second time.

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