Digital Television Terrestrial Broadcasting Primer

	ABOUT THE AUTHOR Pan Feng is currently a research fellow in the Center for Signal Processing, Nanyang Technological University, Singapore. His experience includes 15 years of teaching and research in digital image processing and video engineering. He has offered numerous training courses for industry in these areas.

Digital television (DTV) is a new type of broadcasting technology that will globally transform television as we now know it. DTV refers to the complete digitization of the TV signal from transmission to reception. By transmitting TV pictures and sounds as data bits and compressing them, a digital broadcaster can carry more information than is currently possible with analog broadcast technology. This will allow for the transmission of pictures with HDTV resolution for dramatically better picture and sound quality than is currently available, or of several SDTV programs concurrently. The DTV technology can also provide high-speed data transmission, including fast Internet access.

This article is an introduction to the principles behind digital television broadcasting, including audio/video coding and multiplexing, data scrambling and conditional access, channel coding, and digital modulations. The article also compares the three major DTV standards: ATSC-T, DVB-T, and ISDB-T.

Figure 1:  Diagram of the DTTB system

The block diagram of a digital-terrestrial-television broadcasting system (DTTB) is shown in Figure 1. The video, audio and other service data are compressed and multiplexed to form elementary streams. These streams may be multiplexed again with the source data from other programs to form the MPEG-2 Transport Stream (TS). A transport stream consists of Transport Packets that are 188 bytes in length.

The FEC encoder takes preventive measures to protect the transport streams from errors caused by noise and interference in the transmission channel. It includes Reed-Solomon coding, outer interleaving, and convolutional coding. The modulator then converts the FEC protected transport packets into digital symbols that are suitable for transmission in the terrestrial channels. This involves QAM and OFDM in DVB-T and ISDB-T systems, or PAM and VSB in ATSC-T. The final stage is the upper converter, which converts the modulated digital signal into the appropriate RF channel. The sequence of operations in the receiver side is a reverse order of the operations in the transmitter side.

Data compression technology makes digital television broadcasting possible with a smaller frequency bandwidth than that of an analog system. Among the many compression techniques, MPEG is one of the most accepted for all sorts of new products and services, from DVDs and video cameras to digital television broadcasting. The MPEG-2 standard supports standard-definition television (SDTV) and high-definition television (HDTV) video formats for broadcast applications.

MPEG video compression exploits certain characteristics of video signals, namely, redundancy of information both inside a frame (spatial redundancy), and in-between frames (temporal redundancy). The compression also removes the psychovisual redundancy based on the characteristics of the human vision system (HVS) such that HVS is less sensitive to error in detailed texture areas and fast moving images. MPEG video compression also uses entropy coding to increase data-packing efficiency.

Figure 2:  DCT-based intraframe coding

The intraframe coding algorithm (Figure 2) begins by calculating the DCT coefficients over small non-overlapping image blocks (usually 8x8 in size). This block-by-block processing takes advantage of the image's local spatial correlation properties. The DCT process produces many 2D blocks of transform coefficients that are quantized to discard some of the trivial coefficients that are likely to be perceptually masked. The quantized coefficients are then zigzag scanned to output the data in an efficient way. The final step in this process uses variable length coding to further reduce the entropy.

Figure 3:  Motion-compensated interframe coding

Interframe coding (Figure 3), on the other hand, exploits temporal redundancy by predicting the frame to be coded from a previous reference frame. The motion estimator searches previously coded frames for areas similar to those in the macroblocks of the current frame. This search results in motion vectors (represented by x and y components in pixel lengths), which the decoder uses to form a motion-compensated prediction of the video. The motion-estimator circuitry is typically the most computationally intensive element in an MPEG encoder (Figure 4). Motion-compensated interframe coding, therefore, only needs to convey the motion vectors required to predict each block to the decoder, instead of conveying the original macroblock data, which results in a significant reduction in bit-rate.

Figure 4:  Block diagram of an MPEG-2 video compression system

Unlike video, the three current DTV standards use three different audio coding schemes: Dolby AC-3 for ATSC, MPEG audio and Dolby AC-3 for DVB, and MPEG-AAC for ISDB. However, these audio standards use a similar technique called perceptual coding and support up to six channels—right, left, center, right surround, left surround, and subwoofer—often designated as 5.1 channels. A perceptual audio coder exploits a psycho-acoustic effect known as masking (Figure 5). This psycho-acoustic phenomenon states that when sound is broken into its constituent frequencies, those sounds with relatively lower energy adjacent to others with significantly higher energy are masked by the latter and are not audible.

Figure 5:  Audio perceptual masking

AC-3 is one of the most popular audio compression algorithms used in DTV, movie theater, and home theater systems. AC-3 makes use of the psycho-acoustic phenomenon to achieve great data compression. In the encoding process (Figure 6), a modified DCT algorithm transforms the audio signal into the frequency domain, which generates a series of frequency coefficients that represent the relative energy contributions to the signal of those frequencies.

Figure 6:  Dolby AC-3 audio coding block diagram

By analyzing the incoming signal in the frequency domain, psycho-acoustically masked frequencies are given fewer (or zero) bits to represent their frequency coefficients; dominant frequencies are given more bits. Hence, besides the coefficients themselves, the decoder must receive the information that describes how the bits are allocated so that it may reconstruct the bit allocation. In AC-3, all of the encoded channels draw from the same pool of bits, so channels that need better resolution can use the most bits.

The output coefficients generated by the time-domain to frequency-domain transformation are typically represented in a block floating-point format to maintain numeric fidelity. Using the block floating-point format is one way to extend the dynamic range in a fixed-point processor. It is done by examining a block of (frequency) samples and determining an appropriate exponent that can be associated with the entire block. Once the mantissas and exponents are determined, the mantissas are represented using the variable bit-allocation scheme described above; the exponents are DPCM coded and represented with a fixed number of bits (Figure 6).

MPEG audio is a type of forward adaptive bit allocation, while AC-3 uses hybrid adaptive bit allocation, which combines both the forward and backward adaptive bit allocation. The main advantage of MPEG audio is that the psycho-acoustic model resides only in the encoder. When the encoder is upgraded, legacy decoders continue to decode newly coded data. However, the disadvantage is that it could have a heavy overhead for complicated music pieces.

Audio and video encoders deliver elementary stream outputs. These bit streams, as well as other streams carrying other private data, are combined in an organized manner and supplemented with additional information to allow their separation by the decoder, synchronization of picture and sound, and selection by the user of the particular components of interest. This is done through packetization specified in MPEG-2 systems layer. The elementary stream is cut into packets to form a packetized elementary stream (PES). A PES starts with a header, followed by the content of the packet (payload) and the descriptor. Packetization provides the protection and flexibility for transmitting multimedia steams across the different networks. In general, a PES can only contain the data from the same elementary stream.

Elementary, Packetized Elementary, and Transport Streams
In broadcasting applications, a multiplex usually contain different data streams (audio and video) that might even come from different programs. Therefore, it is necessary to multiplex them into a single stream—the transport stream. Figure 7a shows the process of multiplexing. A transport stream consists of fixed-length transport packets, each exactly 188 bytes long. The header contains important information such as the synchronization byte and the packet identifier (PID). PID identifies a particular PES within the multiplex.

Figure 7:  (a) The process of multiplexing. (b) The structure of a transport packet.

It is necessary to include additional program-specific information (PSI) within each transport stream in order to identify the relationship between the available programs and the PID of their constituent streams. This PSI consists of the four tables: program associate table (PAT), program map table (PMT), network information table (NIT), and conditional access table (CAT).

Within a transport stream, the reserved PID of 0 indicates a transport packet that contains a PAT. The PAT associates a particular PID value with each program that is currently carried in the transport multiplex. This PID value identifies the PMT for that particular program. The PMT contains details of the constituent elementary streams for the program. Program 0 has a special meaning within the PAT and identifies the PID of the transport packets that contains the optional NIT. The contents of the NIT are private to the broadcaster and are intended to contain network-specific information. The CAT is identified by a PID of 1 and contains information specific to any conditional access or scrambling schemes that are in use.

Navigating an MPEG-2 Multiplex
MPEG-2 PSI tables only give information concerning the multiplex. The DVB standard adds complementary tables (DVB-SI) to allow the user to navigate the available programs and services by means of an electronic program guide (EPG). DVB-SI has four basic tables and three optional tables to serve this purpose. The decoder must perform the following main steps in order to find a program or a service in an MPEG-2 transport multiplex.

1. As soon as the new channel is acquired (synchronized), the decoder must filter the PID 0 packets to acquire the PAT sections and construct the PAT to provide the available choice (services currently available on the air) to the user
   
   
2. Once the user choice is made, the decoder must filter the PID corresponding to the PMT of this program and construct the PMT from the relevant sections. If there is more than one audio or video stream, the user should be able to make another choice.
   
   
3. The decoder must filter the PID corresponding to this choice.

The audio/video decoding can now start. The part of this process that is visible to users is the interactive presentation of the EPG associated with the network, which can be built by means of the PSI and DVB-SI tables in order to allow them to easily navigate the available programs and services. Similar tables, Program and System Information Protocol (PSIP) tables, are also available in the ATSC system.

DTV services will either be pay-per-view or at least include some elements that are not freely available to the public. DVB defined a standard for a "Common Interface for Conditional Access and other Digital Video Broadcasting Decoder Applications" to enable an Integrated Receiver Decoder (IRD) to de-scramble programs broadcast in parallel, using different conditional access (CA) systems. By way of inserting a PCMCIA module into the common interface, you can sequentially address different CA systems by that IRD. MultiCrypt describes the simultaneous operation of several CA systems. The MultiCrypt approach has the additional advantage that it does not require agreements between networks, but it is more expensive to implement. Other applications, such as Ethernet connection or electronic commerce, may also utilize the DVB-CI connector.

SimulCrypt is another way of providing the viewer with access to programs. In this case, commercial negotiations between different service providers have led to a contract that enables the viewer to use the one specific CA system built into the IRD to watch all the programs, irrespective of the fact that these programs were scrambled under the control of different CA systems. At the moment, DVB supports both MultiCrypt and SimulCrypt, while ATSC only supports the later.

The transmission channels used for digital television broadcasting are, unfortunately, rather error-prone due to a lot of disturbances (such as noise, interference, and echoes). However, a digital TV signal, after almost all its redundancy is removed, requires a very low bit error-rate (BER) for good performance. A BER of the order of 10^-10 corresponds to an average interval of some 30 minutes between errors. Therefore it is necessary to take preventive measures before modulation in order to allow detection and, as far as possible, correction in the receiver of most errors introduced by the physical transmission channel. These measures are called, collectively, forward error correction (FEC). FEC requires that redundant data is added to the original data prior to transmission, allowing the receiver to use these redundant data to detect and recover the lost data caused by the channel disturbance.

Figure 8:  Forward error correction coding process

Figure 8 illustrates the successive steps of the forward error correction encoding process used in digital television broadcasting. Strictly speaking, energy dispersal is not part of the error correction process. The main purpose of this step is to avoid long strings of 0s or 1s in the transport stream, in order to ensure the dispersal of energy in the channel. Broadcasting standards often use the terms inner coding and outer coding. Inner coding operates just before the transmitter modulates the signal and just after the receiver demodulates the signal. Outer coding applies to the extreme input and output ends of the transmission chain. Inner coding is usually convolutional in nature, with optimal performance under conditions of steady noise interference. Outer coding is a Read-Solomon code that is usually more effective for correcting burst errors.

Read-Solomon Coding
Outer coding is a Reed-Solomon code that is a subset of BCH cyclic block codes. As its name implies, in block coding, a block of bits is processed as a whole to generate the new coded block. It does not have system memory, such that coding of a data word does not depend on what happens before or after that data occurs. Reed-Solomon code, in combination with the Forney convolutional interleaving that follows it, allows the correction of burst errors introduced by the transmission channel. It is applied individually to all the transport packets in Figure 7a, excluding the synchronization bytes. R-S codes have been recently proved to operate at the theoretical limit of correcting efficiency—no more efficient code can be found. This is why it has been chosen for all DTV standards as outer coding. An R-S code is characterized by three parameters (n, k, t) where n is the size of the block after coding, k is the size of the block before coding and t is the number of correctable symbols. Whether the received codeword is error-free could be checked through a division circuit corresponding to the generate polynomial g(x). For a proper codeword, the remainder is zero. In the event that the remainder is non-zero, a Euclidean algorithm is used to decide the two values needed for error correction: the location of the error and the nature of the error. However if the size of the error exceeds half the amount of redundancy added, the error cannot be corrected.

In the ATSC standard, we find the R-S(207,187,10) code. It adds 20 parity bytes and can correct up to 10 erroneous bytes per packet. In the DVB and ISDB standards, we find the R-S(204,188,8) code. It adds 16 parity bytes and can correct up to 8 erroneous bytes per packet.

Interleaving
The purpose of data interleaving is to increase the efficiency of the Reed-Solomon coding by spreading over a longer time the burst errors introduced by the transmission channel, which could otherwise exceed the correction capacity of the Reed-Solomon coding. Interleaving is normally implemented by using a two-dimensional array buffer, such that the data enters the buffer in rows and then read out in columns. The result of the interleaving process is that a burst of errors in the channel after deinterleaving becomes a few scarcely spaced single-symbol errors, which are more easily correctable.

The interleaver employed in the ATSC standard is a 52-data-segment (intersegment) convolutional byte interleaver. Interleaving is provided to a depth of about 1/6 of a data field (4 ms deep). Only data bytes are interleaved. The interleaver is also synchronized to the first data byte of the data field. Intrasegment interleaving is also performed for the benefit of the trellis coding process. DVB and ISDB use convolutional interleaving, and the interleaving depth is 12.

Inner Code
The inner coding is a 2/3 trellis coding for ATSC, and convolutional coding for DVB and ISDB. Inner coding is an efficient complement to the Reed-Solomon coding and Forney interleaving as it is designed to correct random errors.

ATSC Trellis Coding
The 8-VSB transmission system employs a 2/3 rate (R=2/3) trellis code, with one unencoded bit that is precoded. In creating serial bits from parallel bytes, the MSB is sent out first: (7, 6, 5, 4, 3, 2, 1, 0). The MSB is precoded (7, 5, 3, 1) and the LSB is feedback convolutional encoded (6, 4, 2, 0). Standard four-state optimal Ungerboeck codes are used for the encoding (Figure 9); also shown are the precoder and the symbol mapper.

Figure 9:  2/3 trellis coding and precoder

You can use trellis coding with multi-level signaling, in other words, several multi-level symbols are associated into a group. The waveform that results from a particular group of symbols is called a trellis. If each symbol can have eight levels, then in three symbols there can be 512 possible trellises. In trellis coding, the data are coded such that only certain trellis waveforms represent valid data. If only 64 of the trellises represent error-free data, then two data bits per symbol can be sent instead of three. The remaining bit is a form of redundancy because trellises other than the correct 64 are due to errors. If a trellis is received in which the level of one of the symbols is ambiguous due to noise, the ambiguity can be resolved because the correct level is the one that gives a valid trellis. This technique is known as maximum-likelihood decoding. The 64 valid trellises should be made as different as possible to make the system continue to work with a poorer signal to noise ratio. If the trellis coder makes an error, the outer code will correct it.

DVB Convolutional Coding and Puncturing
In DVB, convolutional coding is used, followed by code puncturing. Typically, a 1/2 convolutional consists of two FIR filters. These two FIR filters convolve with the input bit stream, which produces two outputs that represent different parity checks on the input data so that bit errors can be corrected. Clearly, there will be two output bits for every input bit; therefore the code rate is 1/2. Any rate between 1/1 and 1/2 would still allow the transmission of original data, but the amount of redundancy would vary. Failing to transmit the entire 1/2 output is called puncturing and it obtains any required balance between bit rate and error correcting capability. In DVB systems, as well as in ISDB systems, 1/2, 2/3, 3/4, 4/5, 5/6, 7/8 are all possible code rates.

Until now we do not see much difference among the three DTV systems. Differentiation occurs due to the different modulation schemes of the systems. This section briefly describes principles behind those modultion schemes.

ATSC 8-VSB System
The ATSC 8-VSB system was developed by the Advanced Television Systems Committee in the U.S. The framing structure of the transmitted signal is an important aspect of the ATSC standard. It accommodates the transport stream requirements, as well as mitigates channel inter-propagation effects such as multipath and impulse noise.

The transport packet for ATSC consists of 188 bytes, including a sync byte. At the transmitter, this is altered in two ways. First the sync byte is stripped off, leaving 187 bytes to be transmitted. Then 20 bytes are added to this for the the Reed-Solomon error correction, giving 207 bytes transmitted in each packet, which amounts to 1656 bits. The trellis coding at rate 2/3 increases this to 2484 bits, or 828 symbols, since eight-level coding gives three bits per symbol. A special waveform, known as the data segment sync, is added to the head of this packet and occupies four normal symbol periods. The total modified transmission stream packet now occupies 832 symbol periods, or a total time of 77.3 µs at the symbol rate of 10.76 megasymbols per second. This resulting new data packet is now called a data segment.

Figure 10:  VSB data segments and framing structure

Periodically, at intervals of 313 packets or 24.2ms, a special data segment known as a field sync is inserted. The field sync carries training data used by the adaptive equalizer in the receiver to estimate what echoes may be present due to multipath interference. The form of the data segment and overall framing structure is shown in Figure 10.

Figure 11:  Nominal VSB channel occupancy

The eight-level symbols combined with the binary data segment sync and data field sync signals are used to generate a suppressed-carrier-modulate carrier. Before transmission, however, most of the lower sideband is removed. The resulting spectrum is flat, except for the band edges where a nominal square-root raised-cosine response results in 620 kHz transit bands. The nominal VSB transmission spectrum is shown in Figure 11. The spectrum includes a small pilot signal at the suppressed carrier frequency, 310 kHz from the lower band edge.

DVB-T OFDM System
A European consortium of public and private sector organizations—the Digital Video Broadcasting Project—developed the DVB-T OFDM system. The system uses a larger number of carriers-per-channel modulated in parallel via an FFT process, a technique referred to as orthogonal frequency division multiplex (OFDM). In case of multipath interference, echoes could cause severe interference to the main signal. Therefore, long symbol duration is necessary to suppress the echo interference. OFDM can achieve long symbol duration within the same bandwidth using parallel modulation. In OFDM, symbols are demultiplexed to modulate many different carriers (a few thousand), each of which occupies a much narrower bandwidth. Hence, the symbol duration could be increased, though the total bandwidth remains the same. These carriers are chosen to be orthogonal to each other so that they are separable in the decoder. The modulated symbols are frequency multiplexed to form the OFDM baseband signal, which is then up-converted to RF signal for transmission.

The OFDM transmission system allows the selection of different levels of QAM modulation. Moreover, a guard interval with selectable width (1/4, 1/8, or 1/16 of the symbol duration) separates the transmitting symbols, which gives the system an excellent capability for coping with multipath distortion. OFDM modulation also supports a single frequency network, such that in the single coverage area, multiple transmitters are used to transmit the same data using the same frequency at the same time. The DVB-T system can operate in either a 2k mode or 8k mode. The 2k mode uses a maximum of 1705 carriers, while in 8k mode the carrier number is 6817. The 2k mode system has short symbol duration, so it is suitable for a small single-frequency network (SFN) network with limited distance between transmitters. The 8k mode is used in a large SFN network where the transmitters could be up to 90 km apart.

ISDB-T BST-OFDM System
The Association of Radio Industries and Businesses (ARIB) in Japan developed the ISDB-T system. It uses a modulation method referred to as Band Segmented Transmission (BST) OFDM, which consists of a set of common basic frequency blocks called BST-Segments. Each segment has a bandwidth corresponding to 1/14th of the channel bandwidth. BST-OFDM provides hierarchical transmission capabilities by using different punctured coding rates, modulation schemes, and guard intervals on different BST-segments. Thus different segments can meet different service requirements. By transmitting OFDM segment groups with different transmission parameters, you get hierarchical transmission.

Generally speaking, each system has its own unique advantages and disadvantages. Table 1 summarizes the main characteristics of the three DTV systems.

	ATSC 8-VSB	DVB-T OFDM	ISDB-T BST-OFDM
Source Coding
Video	Main profile syntax of ISO/IEC 13818-2 (MPEG-2 video)
Audio	Dolby AC-3	MPEG-2 Audio or Dolby AC-3	MPEG-2 Audio or AAC Audio
Transport Stream	MPEG-2 Transport Stream
Channel Coding Coding
Outer Coding	R-S (207, 187, t=10)	R-S (204, 188, t=8)
Outer Interleaver	52 R-S block interleaver	12 R-S block interleaver
Inner Coding	2/3 trellis code	Punctured convolutional code: 1/2, 2/3, 3/4, 5/6, 7/8, constraint length=7, polynomials 171, 133
Inner Interleaver	12-to-1 trellis code interleaver	Bit-wised interleaving & frequency interleaving	Bit-wised interleaving & time and frequency interleaving
Data Randomization	16-bit PRBS
Modulation
Symbol Mapping		QPSK, xQAM	DQPSK/QPSK, xQAM
Guard Interval		1/32, 1/16, 1/8, 1/4
Hierarchical	No	Yes
No. of Carriers	1	2k and 8k FFT	2k, 4k, and 8k FFT
Bit Rates	19.3 Mbps	3.7-31.7 Mbps	4.06-21.47 Mbps
HDTV Capability	Yes	Increased convolutional coding rate, say, 3/4 is required. This needs additional 1.5dB of power.

Table 1:  Main characteristics of DTV systems

The ATSC system is more robust in an added white Gaussian noise (AWGN) channel, has higher spectrum efficiency, lower peak-to-average power ratio, and is more robust to impulse noise. It also has comparable performance to DVB and ISDB systems at low-level multipath distortion and against analog TV interference. Therefore the ATSC 8-VSB system could be more advantageous for a single transmitter system and for providing HDTV service within a 6 MHz channel to fixed receivers.

The DVB-T system has performance advantages with respect to high-level (up to 0 dB), long-delay multipath distortion. DVB-T could be advantageous for services requiring large-scale, single-frequency networks and for mobile reception. Hierarchical channel coding and modulation, which uses multi-resolution constellation on OFDM carriers, is also available to provide two-tier services within one DTV channel.

The ISDB-T system, which uses the same modulation and channel-coding scheme as the DVB-T system, has similar performance advantages to the DVB-T system. It was designed to operate under large-scale SFN and, particularly, in a mobile reception environment. The depth of the time interleaver can be selected to improve the quality of the mobile reception and immunity against impulse noise. The band-segmented transmission allows the use of up to three different modulation schemes and coding rates on different segments to meet various service requirements and interference conditions.

In conclusion, the period of research and development in digital television broadcasting is largely over, and actual digital television services are now offered in many countries. DTV brings about new services and applications, such as home shopping and home banking, which will bring convenience to users. Interactive TV can allow users to call up program-related information on demand, thus enhancing viewing pleasure. This spells a technical progression similar to, but much profound than, the transition from black-and-white to color television.

References