Digital Broadcast Technologies

Digital technology is becoming pervasive in all types of services. As computing power continues to increase, more and more functions can be tackled in the digital domain. An excellent example is the transmission of television pictures. This sample chapter from OpenCable™ Architecture, winner of NCTA Book of the Year Award, introduces a number of key digital broadcast technologies.

Digital technology is becoming pervasive in all types of services. As computing power continues to increase according to Moore’s Law (see Chapter 1, "Why Digital Television?"), more and more functions can be tackled in the digital domain. An excellent example is the transmission of television pictures.

Nevertheless, digital technology is not a panacea. The complexity of the techniques can introduce reliability and quality issues. In addition, only a few engineers thoroughly understand all these techniques, making us more reliant on a smaller number of de facto standard chip-sets.

This chapter introduces a number of key digital technologies. If you are familiar with these technologies, you might want to skim through or skip over this chapter. There are many excellent texts (see the list of references at the end of the chapter) that explain how these digital technologies work. This chapter does not attempt to repeat them, but instead provides a commentary on why these techniques are so important and what they mean in practical terms. This chapter discusses

• Video compression—The basic principles of the video compression algorithms commonly used for entertainment quality video, the importance of choosing the correct parameters for video encoding, and some alternative video compression algorithms.
  

• Audio compression—The basic principles of MPEG-2 audio compression and Dolby AC-3 audio compression.
  

• Data—Arbitrary private data can be carried by the underlying layers.
  

• System information—Tabular data format used by the digital receiver device to drive content navigation, tuning, and presentation.
  

• Timing and synchronization—A mechanism is required to recover the source timing at the decoder so that the presentation layer can synchronize the various components and display them at exactly the intended rate.
  

• Packetization—The segmentation and encapsulation of elementary data streams into transport packets.
  

• Multiplexing—The combining of transport streams containing audio, video, data, and system information.
  

• Baseband transmission—The various mechanisms for carrying digital transport streams: the Digital Video Broadcast (DVB) asynchronous serial interface (ASI), Synchronous Optical Networks (SONET), Asynchronous Transfer Mode (ATM), and Internet Protocol (IP).

• Broadband transmission—The digital transmission payload must be modulated before it can be delivered by an analog cable system. Quadrature Amplitude Modulation (QAM) has been selected as the best modulation for downstream transmission in cable systems. Other modulation techniques include quaternary phase shift keying (QPSK) and vestigial side band (VSB) modulation.

Figure 4-1 shows how each of these techniques are layered. This diagram illustrates the in-band communications stack for a digital cable set-top and is discussed in Chapter 6, “The Digital Set-Top Converter,” in more detail.

Figure 4-1 Layered Model for Digital Television

Video Compression

Image compression has been around for some time but video compression is relatively new. The processing requirements to compress even a single frame of video are large—to compress 30 frames (or 60 fields) per second of video requires massive processing power (delivered by rapid advances in semiconductor technology).

Nonetheless, digital video must be compressed before it can be transmitted over a cable system. Although other compression algorithms exist, the dominant standard for video compression is MPEG-2 (from Moving Picture Experts Group). Although MPEG-2 video compression was first introduced in 1993, it is now firmly established and provides excellent results in cable, satellite, terrestrial broadcast, and digital versatile disk (DVD) applications.

This section discusses

• MPEG-2 video compression—The basics of the MPEG-2 video compression algorithm and why it has become the dominant standard for entertainment-video compression

• Other video compression algorithms—Why other video compression algorithms have their applications and why they are unlikely to challenge MPEG-2 video compression in the entertainment world for some time

• Details on MPEG-2 video compression—Some more details on the use of MPEG-2 video compression and its parameters

MPEG-2 Compression

MPEG-2 video compression is the de facto standard for entertainment video. MPEG-2 video compression is popular for a number of reasons:

• It is an international standard [ISO/IEC IS 13818-2].

• MPEG-2 places no restrictions on the video encoder implementation. This allows each encoder designer to introduce new techniques to improve compression efficiency and picture quality. Since MPEG-2 video encoders were first introduced in 1993, compression efficiency has improved by 30 to 40%, despite predictions by many that MPEG-2’s fundamental theoretical limitations would prevent this.

• MPEG-2 fully defines the video decoder’s capability at particular levels and profiles. Many MPEG-2 chip-sets are available and will work with any main level at main profile (MP@ML)–compliant MPEG-2 bit-stream from any source. Nevertheless, quality can change significantly from one MPEG-2 video decoder to another, especially in error handling and video clip transitions.

• MPEG-2 video compression is part of a larger standard that includes support for transport and timing functions.

Moreover, MPEG-2 is likely to remain as the dominant standard for entertainment video because it has been so successful in establishing an inventory of standard decoders (both in existing consumer electronics products and in the chip libraries of most large semiconductor companies). Additional momentum comes from the quantity of real-time and stored content already compressed into MPEG-2 format. Even succeeding work by the MPEG committees has been abandoned (MPEG-3) or retargeted to solve different problems (MPEG-4 and MPEG-7).

MPEG-2 is a lossy video compression method based on motion vector estimation, discrete cosine transforms, quantization, and Huffman encoding. (Lossy means that data is lost, or thrown away, during compression, so quality after decoding is less than the original picture.) Taking these techniques in order:

• Motion vector estimation is used to capture much of the change between video frames, in the form of best approximations of each part of a frame as a translation (generally due to motion) of a similar-sized piece of another video frame. Essentially, there is a lot of temporal redundancy in video, which can be discarded. (The term temporal redundancy is applied to information that is repeated from one frame to another.)

• Discrete cosine transform (DCT) is used to convert spatial information into frequency information. This allows the encoder to discard information, corresponding to higher video frequencies, which are less visible to the human eye.

• Quantization is applied to the DCT coefficients of either original frames (in some cases) or the DCT of the residual (after motion estimation) to restrict the set of possible values transmitted by placing them into groups of values that are almost the same.

• Huffman encoding uses short codes to describe common values and longer codes to describe rarer values—this is a type of entropy coding.

The foregoing is a highly compressed summary of MPEG-2 video compression (with many details omitted). However, there are so many excellent descriptions of MPEG compression (see DTV: The Revolution in Electronic Imaging, by Jerry C. Whitaker; Digital Compression for Multimedia: Principles and Standards, by Jerry D. Gibson and others; Testing Digital Video, by Dragos Ruiu and others; and Modern Cable Television Technology; Video, Voice, and Data Communications, by Walter Ciciora and others) that more description is not justified here. Instead, the following sections concentrate on the most interesting aspects of MPEG:

• What are MPEG-2’s limitations?

• What happens when MPEG-2 breaks?

• How can compression ratios be optimized to reduce transmission cost without compromising (too much) on quality?

MPEG Limitations

If MPEG-2 is so perfect, why is there any need for other compression schemes? (There are a great many alternative compression algorithms, such as wavelet, pyramid, fractal, and so on.) MPEG-2 is a good solution for coding relatively high-quality video when certain transmission requirements can be met. However, MPEG-2 coding is rarely used in Internet applications because the Internet cannot generally guarantee the quality of service (QoS) parameters required for MPEG-2–coded streams. These QoS parameters are summarized in Table 4-1.

Table 4-1 MPEG-2 QoS Parameters for Entertainment Quality Video

As you can see from the table, for entertainment-quality video, MPEG-2 typically requires a reasonably high bit rate, and this bit rate must be guaranteed. Video-coding will, in general, produce a variable information rate, but MPEG-2 allows for CBR transmission facilities (for example, satellite transponders, microwave links, and fiber transmission facilities). As such, MPEG-2 encoders attempt to take advantage of every bit in the transmission link by coding extra detail during less-challenging scenes. When the going gets tough—during a car chase, for example—MPEG-2 encoders use more bits for motion and transmit less detail. Another way to think of this is that MPEG-2 encoding varies its degree of loss according to the source material. Fortunately, the human visual system tends to work in a similar way, and we pay less attention to detail when a scene contains more motion. (This is true of a car chase whether you are watching it or you are in it!)

MPEG-2 coded material is extremely sensitive to errors and lost information because of the way in which MPEG-2 puts certain vital information into a single packet. If this packet is lost or corrupted, there can be a significant impact on the decoder, causing it to drop frames or to produce very noticeable blocking artifacts. If you think of an MPEG-2 stream as a list of instructions to the decoder, you can understand why the corruption of a single instruction can play havoc with the decoded picture.

Finally, MPEG-2 is extremely sensitive to variations in transmission delay. These are not usually measurable in synchronous transmission systems (for example, satellite links) because each bit propagates through the system according to the clock rate. In packet- or cell-based networks, however, it is possible for each packet-sized group of bits to experience a different delay. MPEG-2 was designed with synchronous transmission links in mind and embeds timing information into certain packets by means of timestamps. If the timestamps experience significant jitter (or cell delay variation), it causes distortions in audio and video fidelity due to timing variations in the sample clocks—for example, color shifts due to color subcarrier phase variations.

MPEG-2 Artifacts

What are MPEG artifacts? In practice, all lossy encoders generate artifacts, or areas of unfaithful visual reproduction, all the time; if the encoder is well designed, all these artifacts will be invisible to the human eye. However, the best laid plans sometimes fail; the following are some of the more common MPEG-2 artifacts:

• If the compression ratio is too high, there are sometimes simply not enough bits to encode the video signal without significant loss. The better encoders will progressively soften the picture (by discarding some picture detail); however, poorer encoders sometimes break down and overflow an internal buffer. When this happens, all kinds of visual symptoms—from bright green blocks to dropped frames—can result. After such a breakdown, the encoder will usually recover for a short period until once again the information rate gets too high to code into the available number of bits.

• Another common visible artifact is sometimes visible in dark scenes or in close-ups of the face and is sometimes called contouring. As the name suggests, the image looks a little like a contour map drawn with a limited set of shades rather than a continuously varying palette. This artifact sometimes reveals the macro-block boundaries (which is sometimes called tiling). When this happens, it is usually because the encoder allocates too few quantization levels to the scene.

NOTE

Macro-blocks are areas of 16-by-16 pixels that are used by MPEG for DCT and motion-estimation purposes. See Chapter 3 of Modern Cable Television Technology; Video, Voice, and Data Communications by Walter Ciciora and others, for more details.

• High-frequency mosquito noise will sometimes be apparent in the background. Mosquito noise is often apparent in surfaces, such as wood, plaster, and wool, that contain an almost limitless amount of detail due to their natural texture. The encoder can be overtaxed by so much detail and creates a visual effect that looks as if the walls are crawling with ants.

There are many more artifacts associated with MPEG encoding and decoding; however, a well-designed system should rarely, if ever, produce annoying visible artifacts.

MPEG-2 Operating Guidelines

To avoid visible artifacts due to encoding, transmission errors, and decoding, the entire MPEG-2 system must be carefully designed to operate within certain guidelines:

• The compression ratio cannot be pushed too high. Just where the limit is on compression ratio for given material at a certain image resolution and frame rate is a subject of intense and interminable debate. Ultimately, the decision involves engineers and artists and will vary according to encoder performance (there is some expectation of improvements in rate with time, although also some expectation of a law of diminishing returns). Table 4-2 gives some guidance based on experience.

Table 4-2 MPEG-2 Resolution Versus Minimum Bit Rate Guidelines



• The transmission system must generate very few errors during the average viewing time of an event. For example, in a two-hour movie, the same viewers may tolerate very few significant artifacts (such as frame drop or green blocks). In practice, this means that the transmission system must employ forward error correction (FEC) techniques.

Other Video Compression Algorithms

There are a great many alternative video compression algorithms, such as wavelet, pyramid, fractal, and so on (see Chapter 7 of Digital Compression for Multimedia: Principles and Standards by Jerry D. Gibson and others). Many have special characteristics that make them suitable for very low bit rate facilities, for software decoding on a PC, and so on. However, it is unlikely that they will pose a significant threat to MPEG-2 encoding for entertainment video in the near future.

Details of MPEG-2 Video Compression

The following sections detail some of the more practical aspects of MPEG-2 video compression:

• Picture resolution—MPEG-2 is designed to handle the multiple picture resolutions that are commonly in use for broadcast television. This section defines what is meant by picture resolution and how it affects the compression process.

• Compression ratio—MPEG-2 can achieve excellent compression ratios when compared to analog transmission, but there is some confusion about the definition of compression ratios. This section discusses the difference between the MPEG compression ratio and the overall compression ratio.

• Real-time MPEG-2 compression—Most of the programs delivered over cable systems are compressed in real-time at the time of transmission. This section discusses the special requirements for real-time MPEG-2 encoders.

• Non–real-time MPEG-2 compression—Stored-media content does not require a real-time encoder, and there are certain advantages to non–real-time compression systems.

• Statistical multiplexing—This section explains how statistical multiplexing works in data communications systems and what special extensions have been invented to support the statistical multiplexing of MPEG-2 program streams.

• Re-multiplexing—Re-multiplexing, or grooming, of compressed program streams is discussed, including a recent technique that actually allows the program stream to be dynamically reencoded to reduce its bit rate.

Picture Resolution

MPEG-2 compression is a family of standards that defines many different profiles and levels. (For a complete description of all MPEG-2 profiles and levels, see Chapter 5 of DTV: The Revolution in Electronic Imaging by Jerry C. Whitaker.) MPEG-2 compression is most commonly used in its main profile at its main level (abbreviated to MP@ML). This MPEG-2 profile and level is designed for the compression of standard definition television pictures with a resolution of 480 vertical lines.

The resolution of a picture describes how many pixels are used to describe a single frame. The higher the resolution, the more pixels per frame. In many cases, the luminance information is coded with more pixels than the chrominance information.

Chroma subsampling takes advantage of the way the human eye works by sampling the chrominance with less detail than the luminance information. In the MPEG-2 main profile (MP), the chrominance information is subsampled at half the horizontal and vertical resolution compared to the luminance information. For example, if the luminance information is sampled at a resolution of 480 by 720, the chrominance information is sampled at a resolution of 240 by 360, requiring one-fourth the number of pixels. This arrangement is called 4:2:0 sampling. The effect of 4:2:0 sampling is to nearly halve the video bandwidth compared to sampling luminance and chrominance at the same resolution.

Compression Ratio

The compression ratio is a commonly misused term. It is used to compare the spectrum used by a compressed signal with the spectrum used by an equivalent NTSC (National Television Systems Committee) analog signal. Expressed this way, typical compression ratios achieved by MPEG-2 range from 6:1 to 14:1. Why is the term confusing?

If you take the same video signal and modulate it as an analog signal (uncompressed) and compress it using MPEG-2, you have two very different things. The analog signal is an analog waveform with certain bandwidth constraints so that it fits into 6 MHz, whereas the MPEG-2 elementary stream is just a string of bits that cannot be transmitted until further processing steps are taken. These steps include multiplexing, transport, and digital modulation, and they all affect how much bandwidth is required by the compressed signal.

To compare apples with apples, you must take the same video signal and convert it to an uncompressed digital signal (this is actually the first step in the compression process and is termed analog-to-digital conversion or simply sampling). You can now compare the uncompressed digital signal with the MPEG-2 compressed elementary stream for a true comparison of the input bit rate and the output bit rate of the compression process. A full-resolution uncompressed video signal sampled in 4:2:0 (see the previous section, “Picture Resolution”) requires 124.416 Mbps. MPEG-2 can squeeze this down to about 4 Mbps with little or no loss in perceived quality; this is a true compression ratio of 124:4 or 31:1. This is very different than the commonly quoted range of 6:1 to 14:1.

To continue the math, take the 4 Mbps video elementary stream and add an audio stream at 192 Kbps to create a program stream at 4.192 Mbps. Add information to describe how the streams are synchronized and place the data into short transport packets for efficient multiplexing with other streams. You now have a payload of approximately 4.3 Mbps. Using 64-QAM modulation (see the section Broadband Transmission in this chapter), six 4.3 Mbps streams fit into its 27 Mbps payload. Thus, we could express this as a 6:1 compression ratio.

This is all very confusing! In this example, a video signal with a 31:1 MPEG-2 video compression ratio is roughly equivalent to an overall compression ratio of 6:1. (If the example employs 256-QAM and statistical multiplexing, you might achieve an overall compression ratio of 12:1 although the MPEG-2 video compression ratio is still 31:1.)

In this book, the terms MPEG-2 video compression ratio and overall compression ratio will be used to distinguish these very different measures.

Real-Time MPEG-2 Compression

Real-time compression is commonly used at satellite up-links to compress a video signal into a digital program stream as part of the transmission (or retransmission) process. Very often, the encoder runs for long periods of time without manual intervention. There must be sufficient headroom in the allocated bit rate to allow the encoder to operate correctly for all kinds of material that it is likely to encode. (Headroom refers to available, but normally unused, bits that are allocated to allow for the video compression of difficult scenes.) Each channel requires a dedicated encoder, so price is a definite issue for multichannel systems. The encoder must also be highly reliable, and in many cases automatic switching to a backup encoder is required.

Non–Real-Time MPEG-2 Compression

Non–real-time encoders are technically similar to real-time encoders, but have very different requirements. In fact, they may encode in real-time but their application is to encode to a stored media (such as a tape or disc), and a highly-paid compressionist usually monitors the compression of each scene. (Compressionists are studio engineers who not only understand how to operate the encoding equipment but also apply their artistic judgment in selecting the best trade-off between compression ratio and picture quality.) Therefore, encoder price is less of an issue and performance is extremely important because the compressed material will be viewed over and over again. In the case of digital versatile disks (DVDs), no annoying visible artifacts, however subtle, can be tolerated, because the picture quality will be carefully evaluated by a magazine reviewer.

Statistical Multiplexing

Statistical multiplexing is a technique commonly used in data communications to extract the maximum efficiency from a CBR link. A number of uncorrelated, bursty traffic sources are multiplexed together so that the sum of their peak rates exceed the link capacity. Because the sources are uncorrelated, there is a low probability that the sum of their transmit rates will exceed the link capacity. However, although the multiplex can be engineered so that periods of link oversubscription are rare, they will occur. (See Murphy’s law!) In data communications networks, periods of oversubscription are accommodated by packet buffering and, in extreme cases, packet discard. (The Internet is a prime example of an oversubscribed, statistically multiplexed network where packet delay and loss may be high during busy periods.)

Video material has a naturally varying information rate—when the scene suddenly changes from an actor sitting at a table to an explosion, the information rate skyrockets. Although MPEG-2 is designed to compensate by encoding more or less detail according to the amount of motion, the encoded bit rate may vary by a ratio of 5 to 1 during a program.

This makes MPEG-2 program streams excellent candidates for statistical multiplexing, except for the fact that MPEG-2 is extremely sensitive to delay and loss. As such, statistical multiplexing cannot be used for MPEG-2 if there is any probability of loss due to oversubscription.

Therefore, statistical multiplexing has been specially modified for use with MPEG by the addition of the following mechanisms:

• A series of real-time encoders are arranged so that their output can be combined by a multiplexer into a single multi-program transport stream (MPTS). Each encoder has a control signal that instructs it to set its target bit-rate to a certain rate.

• The multiplexer monitors the sum of the traffic from all the encoders as it combines them, and in real-time decides whether the bit rate is greater or lower than the transmission link capacity.

• When one encoder has a more challenging scene to compress, it requests that its output rate be allowed to rise. The hope is that one of the other encoders will have less-difficult material and will lower its output rate.

However, there is a significant probability that all the encoders could be called upon to encode a challenging scene at the same time. When this happens, the aggregate bit rate will exceed the link capacity. A conventional statistical multiplexer would discard some packets, but in the case of MPEG-2, this would be disastrous and almost guarantee poor-quality video at the output of the decoders.

Instead, the multiplexer buffers the additional packets and requests that the encoders lower their encoded bit rate. The buffered packets are delayed by only a few milliseconds, but MPEG-2 is extremely sensitive to delay variation. The multiplexer can fix this within limits; as long as the decoder pipeline does not underflow and the timestamps are adjusted to compensate for the additional time they are buffered, the decoder continues to function normally.

Some statistical multiplexers use a technique called look ahead statistical multiplexing (pioneered by DiviCom—see http://www.divi.com/). In this technique, the material is encoded or statistics are extracted in a first pass, the information is passed to the multiplexer (while the original input video is passing through a pipeline delay), and bit rates are assigned for each encoder; so when the real encoding happens, a reasonable bit rate is already assigned. This solves some of the nasty feedback issues that can happen in less sophisticated designs.

Reencoding

Until recently, it was impossible to modify an encoded MPEG-2-bit stream in real-time. It is now possible, however, to parse the MPEG-2 syntax and modify it to reduce the bit rate by discarding some of the encoded detail. This technique was pioneered by Imedia Corporation (http://www.imedia.com/) and allows the feedback loop between the MPEG encoders and the multiplexer to be removed. In a reencoding (or translation) system, the multiplexer is used to combine a number of variable bit rate MPEG-2 streams. If, at any instant in time, the aggregate bit rate of all of the streams exceeds the transmission link capacity, the multiplexer will reencode one or more of the streams to intelligently discard information to reduce their bit rate. Unlike statistical multiplexing, where the multiplexer could not discard any bits, the multiplexer reduces the bit rate by discarding some information—for example, fine detail.

Reencoding is a very useful technique to use whenever a number of multi-program transport streams are groomed—that is, a new output multiplex is formed from program streams taken from several input multiplexes. Without some means of adapting the coded rate, re-multiplexing would result in considerable inefficiency and the output multiplex would contain fewer channels.

A second application of reencoding is in Digital Program Insertion (DPI). DPI splices one single-program transport program stream into another so that the viewer is unaware of the transition. It can be used to insert local advertisements into a broadcast program. Reencoding allows the inserted segment to be rate-adapted to the segment that it replaces. DPI is discussed in more detail in Chapter 15, "OpenCable Headend Interfaces."

Although reencoding techniques are extremely useful, feedback-controlled statistical multiplexing is superior from a compression-efficiency perspective when it is possible to collocate encoders and multiplexers. Hence, feedback-controlled statistical multiplexing tends to dominate at original encoding sites that include statistical multiplexing, whereas reencoding is appropriate at nodes where grooming of statistically multiplexed signals needs to be performed.

Audio Compression

Audio compression is a companion to video compression, but the techniques are very different. First, the audio signal requires much less bandwidth than the video signal. For example, a stereo audio pair sampled at 48 KHz and using 16-bit samples requires 1.536 Mbps (compared to 124 Mbps for the video signal). It would be quite possible to send uncompressed audio. Moreover, audio compression cannot achieve the same compression ratio as video; a typical rate for the stereo audio pair is 192 Kbps, an 8:1 compression ratio.

Nevertheless, it is easy to make an economic argument for audio compression. For example, in a 40 Mbps transmission link, an additional two video channels can be carried if audio compression is used (assuming 4 Mbps video).

There are two leading contenders for audio compression for entertainment quality audio: MPEG audio compression and Dolby AC-3.

MPEG-1 Layer 2 (Musicam)

MPEG-1 Layer 2 audio compression, also known as Musicam, is specified in ISO/IEC IS 13818-3. MPEG audio compression delivers near–CD quality audio using a technique called sub-band encoding. MPEG audio compression is used mainly in Europe and is used by most direct-to-home satellite providers in the United States.

MPEG audio compression is a two-channel system, but it can encode a Dolby Pro-Logic signal, which includes two additional channels for rear and center speakers. (This is analogous to the way in which a Dolby Pro-Logic signal is carried by a BTSC [Broadcast Television System Committee] signal as part of an NTSC transmission).

Dolby AC-3

Dolby AC-3 is a more advanced system than MPEG audio compression. AC-3 was selected as the audio compression system for digital television in North America and is specified by ATSC Standard A/52. AC-3 encodes up to six discrete channels: left, right, center, left-rear, right-rear, and sub-woofer speakers. The sub-woofer channel carries only low frequencies and is commonly referred to as a 0.1 channel because it has such a limited frequency range. Thus, AC-3 5.1 gets its name from five full channels and a 0.1 channel. In addition, AC-3 has a two-channel mode that can be used to carry stereo and Dolby Pro-Logic encoded signals.

AC-3 also uses sub-band encoding but provides a number of advantages over MPEG audio compression:

• In AC-3 5.1, surround-sound effects are reproduced much more faithfully than is possible with Dolby Pro-Logic because of the additional channels.

• AC-3 5.1 distributes the available transmission link capacity between the 5.1 channels so that more bits are used for those channels containing more information at any particular time. This method effectively makes them statistically multiplexed discrete digital channels.

• AC-3 5.1 is a consumer-grade version of the theatrical Digital Theater Sound (DTS) system, so sound tracks may be directly transferred from the theatrical release.

AC-3 has been chosen as the audio system for digital terrestrial broadcasting and for DVD in the United States. It also has been selected by OpenCable.

Other Audio Compression Algorithms

As with video, many other audio compression algorithms exist. For example, RealAudio is commonly used for sending audio over the Internet, and MP3 has caused a stir in recordable audio. MP3 uses MPEG-1 Layer 3 audio compression, which is considerably more sophisticated than MPEG-1 Layer 2. Many audio compression algorithms have special characteristics that make them suitable for very low bit rate facilities, for software decoding on a PC, and so on. However, it is unlikely that they will pose a significant threat to Dolby AC-3 encoding for entertainment-quality audio in the near future.

Data

Many applications require a data communications path from the headend to the set-top. Part 6 of the MPEG-2 standard (ISO IEC 13818-6) describes how arbitrary data packets may be segmented into MPEG-2 transport packets at the headend and reassembled at the set-top. This mechanism is the basis for the broadcast carousel (see Chapter 8, "Interactive Services").

System Information

System information (SI) is a tabular data format used by the digital receiver device to drive content navigation, tuning, and presentation. See Figure 5-1, which shows that the system information can be sent as part of the in-band digital channel. Cable systems can also transmit system information in an out-of-band channel, as described in Chapter 5, "Adding Digital Television Services to Cable Systems."

There are several standards for system information:

• ATSC A/56—Also known as SI, A/56 was adopted by the cable industry before it was made obsolete by ATSC. DVS-011 and DVS-022 are SCTE standards based on A/56; they are used for basic navigation functions. (Program guide information is sent separately in a proprietary format.)

• DVS-234—Also known as SI, DVS-234 is an SCTE-proposed standard that seeks to establish a common service information standard for out-of-band cable channels. (See the section Out-of-Band Channels in Chapter 16, "OCI-N: the Network Interface.")

• ATSC A/65—Also known as Program and System Information Protocol (PSIP), A/65 has been adopted for terrestrial digital broadcast. In addition to basic navigation functions (such as channel numbering), PSIP can carry content description information that can be used by the programmer to deliver information for electronic program guides. (See the section In-Band Channels in Chapter 16.)

MPEG-2 Systems Layer

The MPEG Systems Committee has defined a systems layer, specified by ISO-IEC 13818-1. (The MPEG-2 systems layer is commonly and confusingly known as MPEG-2 Transport because it defines a transport packet for transmission purposes.) The MPEG-2 systems layer combines the various components of a digital program into a multi-program transport stream. These components include

• Compressed video

• Compressed audio

• Data

• Timing information

• System information

• Conditional access information

• Program-related data

The MPEG-2 systems layer includes the following functions:

• Timing and synchronization—The transmission of timing information in transport packets to allow the receiver to synchronize its decoding rate with the encoder

• Packetization—The segmentation and encapsulation of elementary data streams into 188-byte transport packets.

• Multiplexing—The mechanisms used to combine compressed audio and video streams (elementary streams) into a transport stream.

• Conditional access—Provision for the transmission of conditional access information in the transport stream.

Timing and Synchronization

MPEG-2 transport includes timing and synchronization functions that assume a constant-delay network. The system timing information is transferred using program clock reference (PCR) timestamps, which are placed in the adaption header of certain packets. The PCR timestamps are used by the decoder to synchronize the decoder clock very accurately to the encoder. In many designs, the decoder clock is used to synthesize the NTSC color subcarrier, which must be controlled to an extremely fine tolerance (about 30 Hz).

In addition, the decoder uses the system timing as a reference for presentation and display timestamps, which are used to synchronize audio and video components of a single program transport stream.

Packetization

The MPEG-2 systems layer is packet-based and allows great flexibility in allocating bit rate between a number of program streams. The packets are of a fixed length, and each contains 184 bytes of payload and has a fixed–length, 4-byte link header, as depicted in Figure 4-2. Within each header is a 13-bit packet identifier (PID) that identifies the stream.

Figure 4-2 MPEG-2 Transport Packet Format

MPEG-2 transport packets use an Asynchronous Transfer Mode; that is, there is no direct relationship between the MPEG-2 system time and the clock used by the physical link. This approach supports great flexiblity in the choice of link layer (see the section “Broadband Transmission,” later in this chapter), but it does make timing recovery more complicated.

Not surprisingly, MPEG-2 packets are very similar to ATM cells:

• MPEG-2 packets are fixed-length.

• The identifier has only local significance on the link.

However, MPEG-2 packets are very different from IP packets:

• IP packets are variable-length.

• The identifier (IP address) has global significance in the network.

Multiplexing

The MPEG-2 systems layer supports a two-level hierarchy of multiplexing: the single-program transport stream and the multi-program transport stream.

Individual program elementary streams (PESs) are packetized into MPEG-2 transport packets and multiplexed together to form a single program transport stream. An elementary stream map is included that describes the structure of the MPEG-2 multiplex; the program map table (PMT), as shown in Figure 4-3, tells the decoder which PID values to select for audio and video for that program. A program corresponds to what is traditionally called a channel—for example, HBO, CNN, and so on.

Figure 4-3 Program Association and Program Map Tables.

Multiple single-program transport streams are multiplexed together to form a multiprogram transport stream (or system multiplex). A program stream map is included to describe the structure of multiplex; the program association table (PAT) has an entry for each program and contains a pointer to the PMT. The PAT is always transmitted using a PID of 0.

Multi-program transport streams can be combined or split into fragments by extracting the PAT from each stream and reconstructing a new PAT. This re-multiplexing operation is called grooming. See Chapter 15, “OpenCable Headend Interfaces,” for more details.

Conditional Access

Part of the MPEG-2 transport stream is used to carry conditional access information. Conditional access is used for security, allowing only authorized decoders to access a video stream. The MPEG-2 systems committee was careful not to specify the conditional access system; instead, it standardized a mechanism that could be used by any conditional access system.

Transport packets with a PID equal to 1 (PID 1) are used to carry the conditional access table (CAT), shown in Figure 4-4. The CAT and the PMT can support multiple conditional access messages, and this mechanism is used for simulcrypt systems in Europe and for the Harmony agreement in North America that allows for dual conditional access (see the article "Multiple Conditional Access Systems," by Michael Adams and Tony Wasilewski, in Communications Technology).

Figure 4-4 Conditional Access Table

Limitations of MPEG-2 Systems Layer

The MPEG-2 systems layer was designed to support constant-delay broadcast networks. MPEG-2 transport packets can be adapted to travel over a communications network (as distinct from a communications link), but the QoS must be tightly controlled. The QoS required by MPEG can be provided by connection-oriented networks (for example, SONET or ATM networks). Connectionless, IP networks are evolving to adopt QoS and traffic engineering capabilities, driven initially by a very strong desire to make practical voice-over-IP; the net result is likely to be reasonable ways to engineer MPEG-over-IP (see Quality of Service—Delivering QoS on the Internet and in Corporate Networks by Paul Ferguson and Geoff Huston).

Transmission Mechanisms

There are a number of ways to transmit a multiplexed MPEG-2 transport stream. In some cases, where a dedicated synchronous transmission facility is available (such as SONET), the digital payload can simply be transmitted over a physical channel. In other cases, where it is necessary to transfer the payload over a shared network, higher-level network functions, such as routing and re-multiplexing, are supported by the network.

There are also two main categories of transmission networks: baseband and broadband. In baseband transmission, the entire physical media (twisted-pair, coaxial, laser-link, and so on) is dedicated to the transmission of a bit-stream. This makes transmission very robust but limits overall transmission rates to the maximum speed of a single transmitter or receiver. In broadband networks, each channel is modulated onto a carrier frequency in such a way as not to interfere with any of the other channels. Although broadband networks are not as inherently robust as baseband networks, they achieve a higher transmission capacity at lower cost than baseband networks.

This section covers baseband and broadband transmission as follows:

• Baseband transmission—ATM, SONET, SDH, IP, and DVB ASI provide a baseband transmission facility that can be used to carry MPEG transport streams. This section discusses each of them in turn and compares their suitability.

• Broadband transmission—QPSK, QAM, and VSB are alternative broadband modulation schemes that are commonly used to carry MPEG transport streams. This section discusses their application in cable systems.

Although baseband and broadband transmission are discussed separately, it is possible to layer multiple baseband channels onto a broadband system, and these techniques are by no means exclusive. For example, in the Orlando Full Service Network (see Chapter 11, “On-Demand Cable System Case Studies”), multiple ATM links were modulated and combined into a broadband cable system for delivery to the set-top.

Baseband Transmission

In baseband networks, the MPEG-2 transport packets must be adapted so they can be carried by the network. It was understood by the architects of the Grand Alliance system that it would be necessary to distribute MPEG-2 transport streams over a number of different network types, and considerable effort was spent in choosing an efficient mapping of MPEG-2 packets into ATM.

Asynchronous Transfer Mode

Asynchronous transfer mode (ATM) is a connection-oriented network protocol that can be used to build wide-area network (WAN) switched networks. ATM can provide guaranteed QoS metrics, which are sufficient to support MPEG transport streams. However, ATM adds an additional 12% overhead over native MPEG-2 transmission.

The following are characteristics of ATM:

• Multiplexing structure—ATM is a data-link protocol that can carry video, audio, or data in fixed-length cells that carry a 48-byte payload. Each cell has a 5-byte header that is used to identify the connection. ATM supports bandwidth-on-demand—by varying the cell rate, connections of any required bandwidth may be created.

• MPEG-into-ATM mapping—The mapping of MPEG-2 packets into ATM packets was studied by the Grand Alliance (GA), and the MPEG-2 packet size was set at 188 bytes to facilitate mapping into ATM adaptation layer type 1 or 2, as shown in Figure 4-5.

Figure 4-5 Grand Alliance Mapping for AAL Type 1 or 2

Since that time, an alternative adaptation layer, AAL type 5, has become popular in ATM applications, as illustrated in Figure 4-6. AAL type 5 mapping has been nicknamed straight-8 mapping because it packs two transport packets into eight ATM cells.

Figure 4-6 Straight-8 Mapping for AAL type 5

• Error detection and recovery—The header of each ATM cell is protected by a header error check sequence to protect the connection identifier from becoming corrupted. However, error detection of the payload is the responsibility of the ATM adaptation layer. AAL type 1 and 2 do not support error detection, whereas AAL type 5 computes a 32-bit cyclic redundancy check (CRC) across the payload. This provides a valuable error detection mechanism for data, but it is not much use for streaming (that is, video or audio) applications because there is no time to retransmit a corrupted packet.

• Timing—ATM specifies a timing mechanism for synchronous payloads called Synchronous Residual Time Stamp (SRTS). However, this mechanism is designed for telephone carrier emulation (for example, T1 or E1 emulation) and is not applicable to MPEG-2 transmission. Studies have shown that it is possible to transmit MPEG-2 across ATM networks without the need for system timestamp correction if the ATM cell delay variation is tightly controlled (see Testing Digital Video, by Dragos Ruiu and others). In addition, it is possible to use timing recovery mechanisms to “de-jitter" an MPEG-2 transport stream carried over ATM before delivering it back to a constant delay transmission system (such as QAM).

• Switching—The ATM cell structure was designed so that switches can be implemented entirely in hardware making multigigabit switches possible. ATM switches are extremely cost-effective for this reason. In video-on-demand applications, a switching function is required to deliver a program stream from a server to a particular set-top. Because MPEG-2 switches are not commercially available, ATM switches are often used to build video-on-demand networks (see the section Time Warner Full Service Network in Chapter 11).

• Limitations—ATM adaptation requires additional hardware to map MPEG-2 transport packets into cells at source and then to reassemble the cells into packets at the destination. This adaptation overhead is justified only if a true wide-area switching service is required. Few distribution applications have been deployed that require a switching function, and for broadcast systems the cost of ATM is not justified [Adams].

Synchronous Optical Networks

Synchronous Optical Network (SONET) is a North American standard specified by Bellcore for digital optical transmission. There is an equivalent European standard called the Synchronous digital hierarchy (SDH), which is specified by the International Telecommunications Union (ITU). SONET is a link-layer protocol that carries synchronous payloads in multiples of 50.112 Mbps (within a 51.84 Mbps STS-1). Similarly, SDH carries synchronous payloads in multiples of 150.336 Mbps (within a 155.52 Mbps STM-1). The SONET STS-3c is identical to the SDH STM-1.

The following are characteristics of SONET/SDH:

• Mapping—SONET and SDH are ideal candidates for the carriage of MPEG-2 transport streams, but no direct mapping exists for MPEG-2 into SONET or SDH payloads. An alternative approach is to map a multi-program transport stream (for example, the entire payload of a DVB ASI link) into a single ATM Virtual Channel. The ATM Virtual Channel is then mapped into a SONET payload. This approach has the advantage that only a single ATM segmentation and reassembly (SAR) process is required.

• Error detection and recovery—Optical networks run error-free if properly maintained, and failures are usually catestrophic due to equipment failure of a fiber cut. For this reason, SONET includes mechanisms for error monitoring and protection switching. A block check is used to monitor the error rate on each SONET link (which is usually zero). If the error rate exceeds a set threshold, a protection switch will be made to a spare link (assuming it is available and its error rate is below threshold). SONET protection switches take less than 50 milliseconds to complete but, despite this, are quite noticeable to a customer watching MPEG-2 compressed video.

Internet Protocol

IP is widely used as a data communications protocol. Recently, there has been considerable interest in using IP to carry telephony, audio, and video services. Although IP was not designed with QoS in mind, there has been considerable effort to provide QoS over IP networks (see Quality of Service—Delivering QoS on the Internet and in Corporate Networks by Paul Ferguson and Geoff Huston).

The following provides a brief description of IP:

• Multiplexing structure—IP is a network protocol that can carry arbitrary data in variable-length packets. IP supports bandwidth-on-demand—by varying the packet rate, flows of any required bandwidth may be created.

IP’s variable-length packet is a disadvantage in delay-sensitive applications, because larger packets can delay shorter packets by taking considerable time to traverse a link. However, as link speed increases, this effect is less noticeable. By using a technique called packet over SONET (PoS) to map IP packets directly into high-bandwidth SONET payloads, delays due to large packets are greatly reduced. (For example, at OC-12 rates of 622 Mbps, a 4 KB packet occupies the link for only 51 microseconds.)

• Error detection and recovery—Like ATM, IP’s approach to error detection and recovery is datacentric. That is, errors may be detected by computing a CRC-32 across the packet, and if an error is detected the Transmission Control Protocol (TCP) is used to retransmit the packet.

For streaming applications, retransmission is not useful, so user datagram protocol (UDP) is used because it has no payload error checking and requires less overhead than TCP.

• Routing—IP routers were first constructed from general purpose minicomputers. However, router design has evolved to the point that routers can be implemented almost entirely in hardware. By placing the packet-forwarding function in hardware, the forwarding delay can be reduced by orders of magnitude, and high-performance routers can approach ATM switches in delay performance.

However, IP routing is fundamentally very different from ATM switching in architectural terms. Classical IP routing is completely connectionless, which means that there is no state knowledge in the IP routers about the packet flow. This makes IP routing very flexible and obviates any need for connection management; however, it also means that by nature it is impossible to predict the load on any particular link or router. Thus, there is a statistical probability that a particular route may become congested, and this can interfere with a particular IP flow. The effect to the user is that video may freeze, or voice may become garbled for some period of time until the congestion clears.

• Limitations—IP’s ability to support connectionless networks is its Achilles’ heel for any application requiring QoS guarantees. Connectionless networks support dynamic reconfiguration by rerouting around congested or failed links in the network. This makes packet delay variation very difficult to control and, as we have seen, MPEG-2 is extremely sensitive to variations in delay. Moreover, IP networks currently do not provide admission control mechanisms, so the only way to obtain bandwidth guarantees is by overprovisioning.

DVB Asynchronous Serial Interface

DVB asynchronous serial interface (ASI) was developed for the interconnection of professional MPEG-2 equipment and is a native baseband transmission facility for MPEG-2 transport streams. DVB ASI uses 8b/10b coding at a line rate of 270 Mbps yielding a maximum payload of 216 Mbps. DVB ASI is designed to use two physical media:

• Coaxial cable—Coaxial cable is less expensive than optical fiber and ideal for interconnecting racks of equipment in the headend. However, coaxial cable attenuation limits the reach to about 5 meters.

• Optical fiber—Optical fiber is more expensive than coaxial cable but supports considerably greater reach. Multimode fiber, which has a reach of several kilometers, is typically used, but there is no physical reason why single-mode fiber transceivers (with a reach of up to 100 kilometers) could not be used.

A brief description of DVB ASI’s characteristics follows:

• Transmission format—The DVB ASI transmission format is shown in Figure 4-7. Each byte is encoded as 10 bits using 8b/10b coding, and each MPEG transport packet is preceded and trailed by at least 2 synchronization bytes. A packet may be interspersed with an arbitrary number of stuff bytes.

• Error detection and recovery—DVB ASI has no mechanisms for error detection or recovery because it is designed to be for interconnection of equipment over short distances.

Figure 4-7 DVC ASI Transmission Format

• Timing—DVB ASI, as its name suggests, is asynchronous. That is, there is no relationship between the line clock and the MPEG system timing. Therefore, each piece of equipment using a DVB ASI input performs timestamping using a high-fidelity local 27 MHz counter. (see the section MPEG-2 Systems Layer, later in this chapter). When MPEG-2 transport packets are output onto the ASI link, they are timestamped again to compensate for any jitter introduced by re-multiplexing.

• Limitations—DVB ASI is limited to the interconnection of equipment over short distances because of its lack of physical reach and error protection mechanisms.

Comparison of Baseband Transmission Alternatives

Table 4-3 compares ATM, SONET/SDH, IP, and DVB ASI for the baseband transmission of MPEG-2 transport streams. Although it is unfair, strictly speaking, to compare data link protocols with network protocols, this distinction is lost on the engineer who needs to decide how to move an MPEG-2 transport stream from one location to another.

Table 4-3 Comparison of Baseband Transmission Alternatives for MPEG-2 Transport

Baseband Transmission Interworking

As discussed in the previous sections on baseband transmission, it is quite common to layer one or more protocols on top of another. Figure 4-8 summarizes the protocol layerings that are, or are in the process of being, standardized.

Figure 4-8 Standard Protocol Layering

From left to right, Figure 4-8 illustrates the following mappings:

• An MPEG-2 multi-program transport stream is mapped into a single ATM Virtual Channel, providing transport of an entire DVB ASI payload over an ATM/SONET network.

• An MPEG-2 single-program transport stream is mapped into a single ATM Virtual Channel for transport over an ATM/SONET network.

• An MPEG-2 single-program transport stream is mapped into IP packets. The IP connection is supported by a single ATM Virtual Channel for transport over an ATM/ SONET network.

• An MPEG-2 single-program transport stream is mapped into IP packets. The IP connection is supported by a direct packet over an SONET (POS) adaptation layer to provide transport over an SONET network.

In networks, simplicity is usually most cost-effective, and DVB ASI is the simplest and least expensive approach for local interconnection of equipment. The mapping of an MPEG-2 MPTS into a single ATM virtual channel is also gaining favor with cable companies that need to deliver MPEG-2 transport streams over some distance.

Broadband Transmission

Transmission in a broadband system uses modulation to separate each channel into a given frequency band. This technique is often called frequency-division multiplexing (FDM). This section discusses the three common modulation techniques used for MPEG-2 transport in North America: QPSK, QAM, and VSB.

Error correction and protection techniques are typically employed in broadband transmission systems to reduce the number of errors introduced by analog transmission; these are also discussed in this section.

Quaternary Phase Shift Keying

Quaternary phase shift keying (QPSK) modulation is very robust in the presence of noise, so QPSK is used for satellite transmission links and for control channel modulation in cable systems (see the section Out-of-Band Communications in Chapter 5).

Figure 4-9 shows how QPSK modulation is applied to a baseband signal. Two bits are encoded per baud. The 2-bit symbol is divided into one in-phase (I) bit and one quadrature-phase (Q) bit, which are each converted to an analog level. These levels are used to modulate the amplitude and phase of a carrier.

Figure 4-9 QPSK Modulator Block Diagram

Figure 4-10 shows the constellation diagram for QPSK and the symbol mapping for each phase angle and amplitude vector. QPSK is very robust because the detector needs to detect only two levels and two phase angles to determine the symbol. (QPSK is equivalent to 4-QAM.)

Figure 4-10 QPSK Constellation Diagram

Quadrature Amplitude Modulation

Figure 4-11 shows how 64-QAM modulation is applied to a baseband signal. Six bits are encoded per baud, which is three times as efficient as QPSK. The 6-bit symbol is divided into three in-phase (I) bits and three quadrature-phase (Q) bits, which are each converted to an analog level. These levels are used to modulate the amplitude and phase of a carrier.

Using the North American standard for 64-QAM modulation (ITU J.83 Annex B), a payload of approximately 27 Mbps is achieved within a 6 MHz channel. This is an efficiency of 4.5 bits per baud—considerably less than the theoretical maximum of 6 bits per baud. There are two main reasons for this:

• The entire 6 MHz bandwidth cannot be used, and guard-bands need to be introduced on either side of the signal to prevent interference between adjacent channels.

• Error correction and protection mechanisms introduce some overhead (see the section Forward Error Correction, later in this chapter).

Figure 4-11 64-QAM Modulator Block Diagram

Figure 4-12 shows the constellation diagram for 64-QAM. There are 64 different phase angle and amplitude vectors and, for this reason, 64-QAM is less robust because the detector needs to differentiate between these to determine the symbol. In practice, 64-QAM requires a carrier-to-noise ratio in excess of 22 dB to work (see Chapter 4 of Modern Cable Television Technology; Video, Voice, and Data Communications by Walter Ciciora and others, for more details).

Figure 4-13 shows how 256-QAM modulation is applied to a baseband signal. Eight bits are encoded per baud, and this is 33% more efficient than 64-QAM. The 8-bit symbol is divided into four in-phase (I) bits and four quadrature-phase (Q) bits, which are each converted to an analog level. These levels are used to modulate the amplitude and phase of a carrier.

Using the North American standard for 256-QAM modulation (ITU J.83 Annex B), a payload of approximately 38.8 Mbps is achieved within a 6 MHz channel. This is an efficiency of 6.47 bits per baud—considerably less than the theoretical maximum of 8 bits per baud. A payload of 38.8 Mbps was chosen because it is sufficient to carry two HDTV channels.

Figure 4-12 64-QAM Constellation diagram

Figure 4-13 256-QAM Modulator Block Diagram

Figure 4-14 shows the constellation diagram for 256-QAM. There are 256 different phase angle and amplitude vectors, making the points on the constellation closer together; for this reason, 256-QAM is less robust than 64-QAM. In practice, 256-QAM requires a carrier-to-noise ratio in excess of 28 dB to work in practice (6 dB more than 64-QAM).

Figure 4-14 256 QAM Constellation Diagram

There has been some discussion of still higher orders of QAM modulation—512-QAM, 768-QAM, or even 1024-QAM. However, there are diminishing returns—1024-QAM increases the payload by only 25% over 256-QAM—and it is unlikely that these will be used in the near future.

Vestigial Side Band

VSB-8 modulation has been adopted for use in terrestrial digital broadcasting (see A.53 Annex D). VSB-8 has a payload of approximately 19.4 Mbps and was designed to carry a single HDTV channel. VSB-8 is a one-dimensional modulation scheme because it uses only amplitude modulation of the carrier (in contrast, QAM is two-dimensional modulation technique because it uses both I and Q components). In VSB-8 modulation, the baseband signal is coded as an 8-level value, so 3 bits are encoded per baud.

Error Correction and Protection

Analog transmission systems are subject to noise, distortion, and interference from other carriers. Therefore, error correction and protection techniques are used to maintain an acceptable error rate.

Three commonly used techniques are discussed in this section: forward error correction, interleaving, and trellis coding.

Forward Error Correction

Forward error correction (FEC) uses a mathematical function to generate a check sequence across the payload data. The check sequence is transmitted with the data, and the same mathematical function is used at the receiver to check for payload errors and to correct errors. This technique is also used in error correcting code (ECC) memory.

The Reed Solomon (RS) function is used in conjunction with QPSK, QAM, and VSB modulation. RS (204,188) t=8 describes a scheme where 16 check bytes are generated for each MPEG-2 packet. This represents an overhead of 8.5% but allows 1- and 2-byte errors to be corrected by the receiver.

Interleaving

Noise pulses can obliterate a signal for several microseconds, and at a 256-QAM payload rate of 38.8 Mbps, this represents hundreds of bits. By itself, FEC is incapable of correcting such long error sequences.

Interleaving effectively spreads the payload data over time. Figure 4-15 shows an example of a block interleaver developed by Scientific Atlanta for the Time Warner Full Service Network (see Chapter 11).

Figure 4-15 Block Interleaver Used in the Time Warner Full Service Network

In this implementation, a noise pulse affects only 1 byte in each row, a situation that can be rectified by the RS correction. The main disadvantage of interleaving is buffer memory and delay, which are 32 KB and 11.5 milliseconds, respectively, in this implementation.

Convolutional interleavers require only half the memory and introduce only half the delay of block interleavers. For this reason, both ITU J.83 Annex B (QAM) and ATSC A/53 Annex D (VSB) use convolutional interleaving. Figure 4-16 shows an example of a convolutional interleaver with an interleave depth of six. The blocks (labeled J) buffer the payload and operate as a shift register. The interleaver (at the modulator) and the deinterleaver (at the demodulator) are synchronized so that the payload is reassembled in its original form.

The effect of interleaving is to distribute the errors due to a noise burst over a period of time so that the errored bits are no longer adjacent, which makes FEC more effective.

Figure 4-16 Example of a Convolutional Interleaver

Trellis Coding

As noise is introduced into a QAM or VSB signal, it perturbs the points on the constellation diagram so that ultimately they overlap and the decoder sees the wrong symbol. Trellis coding adds some redundancy and uses sophistical mathematics at the receiver to determine the best fit of the constellation to a symbol. The trellis coding specified by ITU J.83 Annex B QAM uses a 14/15 coding rate to improve noise immunity by approximately 2 dB. VSB-8, which is designed for more challenging broadcast applications, specifies a 2/3 trellis coding rate, improving noise immunity but reducing payload rate by 33%.

Summary

This chapter introduced a number of important techniques and concepts:

• Video and audio compression algorithms and how they are used to increase the channel capacity of a cable system

• The MPEG-2 systems layer and the multiplexing of multiple program elementary streams into a single physical channel

• Baseband transmission mechanisms for compressed audio and video streams, including ATM, SONET, IP, and DVB ASI

• Broadband transmission mechanisms for compressed audio and video streams, including QPSK, QAM, and VSB modulation

References

Books

Ciciora, Walter, James Farmer, and David Large. Modern Cable Television Technology; Video, Voice, and Data Communications. San Francisco, CA: Morgan Kaufmann Publishers Inc., 1999.

Evans, Brian. Understanding Digital TV—The Route to HDTV. New York, NY: IEEE Press, 1995.

Ferguson, Paul, and Geoff Huston. Quality of Service—Delivering QoS on the Internet and in Corporate Networks. New York, NY: John Wiley and Sons, 1998.

Gibson, Jerry D., Toby Berger, Tom Lookabaugh, Dave Lindberg, and Richard L. Baker.

Digital Compression for Multimedia: Principles and Standards. San Francisco, CA: Morgan Kaufmann Publishers, Inc., 1998.

Hodge, Winston William. Interactive Television—A Comprehensive Guide to Multimedia Technologies. New York, NY: McGraw-Hill, 1994.

Ruiu, Dragos et al. Testing Digital Video. Hewlett-Packard, 1997.

Whitaker, Jerry C. DTV: The Revolution in Electronic Imaging. New York, NY: McGraw-Hill, 1998.

Periodicals

Adams, Michael. "ATM and MPEG-2 in Cable TV Networks, Parts 1 and 2." Communications Technology, December 1995 and February 1996.

Adams, Michael, and Tony Wasilewski. "Multiple Conditional Access Systems."

Communications Technology, April 1997.

Standards

ATSC Standard A/52, Digital Audio Compression (AC-3), 1995.

ATSC Standard A/53, Digital Television Standard, 1995.

ATSC Document A/54, Guide to the Use of the ATSC Digital Television Standard, 1995.