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At a meeting hosted in New York by Columbia University, the Moving Picture Experts Group (MPEG) completed definition of MPEG-2 Video, MPEG-2 Audio, and MPEG-2 Systems. MPEG therefore confirmed that it is on schedule to produce, by November 1993, Committee Drafts of all three parts of the MPEG-2 Standard, for balloting by its member countries. To ensure that a harmonized solution to the widest range of applications is achieved, MPEG, an ISO/IEC working group designated JTC1/SC29/WG11, is working jointly with the ITU-TS Study Group 15 Experts Group for ATM Video Coding. MPEG also collaborates with representatives from other parts of ITU-TS, and from EBU, ITU-RS, SMPTE, and the North American HDTV community. The final approval of ISO/IEC 13818-1 (MPEG-2 Systems), ISO/IEC 13818-2 (MPEG-2 Video) and ISO/IEC 13818-3 (MPEG-2Audio) as International Standard (IS) was given by the 29th meeting of ISO/IEC JTC1/SC29/WG11 (MPEG) held in Singapore in November 1994. |
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The MPEG-2 concept is similar to MPEG-1, but includes extensions to cover a wider range of applications. The primary application targeted during the MPEG-2 definition process was the all-digital transmission of broadcast TV quality video at coded bitrates between 4 and 9 Mbit/sec. However, the MPEG-2 syntax has been found to be efficient for other applications such as those at higher bit rates and sample rates (e.g. HDTV). The most significant enhancement over MPEG-1 is the addition of syntax for efficient coding of interlaced video (e.g. 16x8 block size motion compensation, Dual Prime, et al). Several other more subtle enhancements (e.g. 10-bit DCT DC precision, non-linear quantization, VLC tables, improved mismatch control) are included which have a noticeable improvement on coding efficiency, even for progressive video. Other key features of MPEG-2 are the scalable extensions which permit the division of a continuous video signal into two or more coded bit streams representing the video at different resolutions, picture quality (i.e. SNR), or picture rates. MPEG-1 was optimized for CD-ROM or applications at about 1.5 Mbit/sec. Video was strictly non-interlaced (i.e. progressive). The international co-operation had executed so well for MPEG-1, that the committee began to address applications at broadcast TV sample rates using the CCIR-601 recommendation (720 samples/line by 480 lines per frame by 30 frames per second or about 15.2 million samples/sec including chroma) as the reference. Unfortunately, today's TV scanning pattern is interlaced. This introduces a duality in block coding: do local redundancy areas (blocks) exist exclusively in a field or a frame ?
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What are the typical MPEG-2 bitrates and picture quality ? Here are some examples of typical frame sizes in bits :
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Where will we see MPEG-2 in everyday life ?
In September 1993, the consortium of 85 European companies signed an agreement to fund a project known Digital Video Broadcasting (DVB) which will develop a standard for cable and terrestrial transmission by the end of 1994. The scheme will use MPEG-2. This consortium has put the final nail in the coffin of the D-MAC scheme for gradual migration towards an all-digital, HDTV consumer transmission standard. The only remaining analog or digital-analog hybrid system left in the world is NHK's MUSE (which will probably be axed in a few years).
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What did MPEG-2 add to MPEG-1 in terms of syntax/algorithm ?
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What are the scalable modes of MPEG-2 ? Scalable video is permitted only in the Main+ and Next profiles. Currently, there are four scalable modes in the MPEG-2 toolkit. These modes break MPEG-2 video into different layers (base, middle, and high layers) mostly for purposes of prioritizing video data. For example, the high priority channel (bitstream) can be coded with a combination of extra error correction information and decreased bit error (i.e. higher Carrier-to-Noise ratio or signal strength) than the lower priority channel. Another purpose of scalability is complexity division. For example, in HDTV, the high priority bitstream (720 x 480) can be decoded under noise conditions were the lower priority (1440 x 960) cannot. This is graceful degradation. By the same division however, a standard TV set need only decode the 720 x 480 channel, thus requiring a less expensive decoder than a TV set wishing to display 1440 x 960. This is simulcasting. A brief summary of the MPEG-2 video scalability modes:
Other scalability modes were experimented with in MPEG-2 video (such as Frequency Scalability), but were eventually dropped in favor of methods that demonstrated similar quality and greater simplicity. |
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What is the TM (Test Model) rate control and adaptive quantization technique ? Test model was not by any stretch of the imagination meant to be the show-stopping, best set of algorithm. It was designed to exercise the syntax, verify proposals, and test the relative performance of proposals in a way that could be duplicated by co-experimentors in a timely fashion. Otherwise there would be more endless debates about model interpretation than actual time spent in verification. The MPEG-2 Test Model (TM) rate control method offers a dramatic improvement to the Simulation Model (SM) method used for MPEG-1. TM's improvements are due to more sophistication pre-analysis and post-analysis routines. Rate control and adaptive quantization are divided into three steps:
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MPEG-2 Video is a generic method for compressed representation of video sequences using a common coding syntax defined in the document ISO/ IEC 13818 Part 2 (CD: Nov. 1993, DIS: March 1994) by the International Organization for Standardization (ISO) and the International Electrotechnical Commission (IEC), in collaboration with the International Telecommunications Union (ITU) as Recommendation H.262. The MPEG-2 Video Standard specifies the coded bit stream for high-quality digital video. As a compatible extension, MPEG-2 Video builds on the completed MPEG-1 Video Standard (ISO/IEC IS 11172-2), by supporting interlaced video formats and a number of other advanced features, including features to support HDTV. As a generic International Standard, MPEG-2 Video is being defined in terms of extensible profiles, each of which will support the features needed by an important class of applications. At the Sydney MPEG meeting, the MPEG-2 Main Profile was defined to support digital video transmission in the range of about 2 to 15 Mbits/sec over cable, satellite, and other broadcast channels, as well as for Digital Storage Media (DSM) and other communications applications. Building on this success at the New York meeting, MPEG experts from participating countries in Asia, Australia, Europe, and North America further defined parameters of the Main Profile and Simple Profile suitable for supporting HDTV formats. MPEG experts also extended the features of the Main Profile by defining a hierarchical/scalable profile. This profile aims to support applications such as compatible terrestrial TV/ HDTV, packet-network video systems, backward compatibility with existing standards (MPEG-1 and H.261), and other applications for which multi-level coding is required. For example, such a system could give the consumer the option of using either a small portable receiver to decode standard definition TV, or a larger fixed receiver to decode HDTV from the same broadcast signal. The technical definition of MPEG-2 Video has been completed. This was a critical milestone, and MPEG-2 Video was scheduled for a Committee Draft in November 1993. |
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What are MPEG-2 VIDEO Main Profile and Main Level ? It is inappropriate to talk about MPEG-2 profiles without also talking about levels. The 4 profiles define the colorspace resolution, and scalability of the bitstream.
The combination of a profile and a level produces an architecture which defines the ability of a decoder to handle a particular bitstream. MPEG-2 Video Main Level is analogous to MPEG-1's CPB, with sampling limits at CCIR-601 parameters (720 x 480 x 30 Hz). Profiles limit syntax (i.e. algorithms), whereas Levels limit parameters (sample rates, frame dimensions, coded bitrates, etc.). Together, Video Main Profile and Main Level (abbreviated as MP@ML) normalize complexity within feasible limits of 1994 VLSI technology (0.5 micron), yet still meet the needs of the majority of application users.
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At what bitrates is MPEG-2 video optimal ? The Test subgroup has defined a few examples :
There are two separate reasons for the popularity of 544 pixel/line :
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How does MPEG video really compare to TV, VHS, laserdisc ? VHS picture quality can be achieved for source film video at about 1 million bits per second (with proprietary encoding methods). It is very difficult to objectively compare MPEG to VHS. The response curve of VHS places -3 dB at around 2 MHz of analog luminance bandwidth (equivalent to 200 samples/line). VHS chroma is considerably less dense in the horizontal direction than MPEG source video (compare 80 samples/ line to 176!). From a sampling density perspective, VHS is superior only in the vertical direction (480 lines compared to 240), but when taking into account interfield magnetic tape crosstalk and the TV monitor Kell factor, not by all that much. VHS is prone to timing errors (which can be improved with time base correctors), whereas digital video is fully discretized. Pre-recorded VHS is typically recorded at very high duplication speeds (5 to 15 times real time playback), which leads to further shortfalls for the format that has been with us since 1977. Broadcast NTSC quality can be approximated at about 3 Mbit/sec, and PAL quality at about 4 Mbit/sec. Of course, sports sequences with complex spatial-temporal activity need more like 5 and 6 Mbit/sec, respectively. Laserdisc is a tough one to compare. Disc is composite video (NTSC or PAL) with up to 425 TVL (or 567 samples/line) response. Thus it could be said that laserdisc has 567 x 480 x 30 Hz resolution. The carrier-to-noise ratio is typically better than 48 dB. Timing is excellent. Yet some of the clean characteristics of laserdisc can be achieved at 1.15 Mbit/sec ( SIF rates), especially for those areas of medium detail (low spatial activity) in the presence of uniform motion. This is why some people say MPEG-1 video at 1.15 Mbit/sec looks almost as good as Laserdisc or Super VHS. Regardless of the above figures, those clever proprietary encoding algorithms can push these bitrates even lower. |
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Why film does so well with MPEG ? Several reasons, really:
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What are some pre-processing enhancements ?
The basic idea of multidimensional/multirate pre-processing is to apply source video whose resolution (sampling density) is greater than the target source and reconstruction sample rates. This follows the basic principles of oversampling, as found in A/D converters.
Most detail is contained in the lower harmonics anyway. Sharp-cut off filters are not widely practiced,
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Why use advanced pre-filtering techniques ? Think of the DCT and quantizer as an A/D converter. Think of the pre-filter as the required anti-alias prefilter found before every A/D. The big difference of course is that the DCT quantizer assigns a varying number of bits per sample (transform coefficient). Judging on the normalized activity measured in the pre-analysis stage of video encoding, and the target buffer size status, you have a fairly good idea of how many bits can be spared for the target macroblock, for instance. Other pre-filtering techniques mostly take into account: texture patterns, masking, edges, and motion activity. Many additional advanced techniques can be applied at different immediate layers of video encoding (picture, slice, macroblock, block, etc.). |
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What are some advanced encoding methods ?
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MPEG is developing the MPEG-2 Audio Standard for low bitrate coding of multichannel audio. MPEG-2 Audio coding will supply up to five full bandwidth channels (left, right, center, and two surround channels), plus an additional low frequency enhancement channel, and/or up to seven commentary/multilingual channels. The MPEG-2 Audio Standard will also extend the stereo and mono coding of the MPEG-1 Audio Standard (ISO/ IEC IS 11172-3) to half sampling-rates (16 kHz, 22.05 kHz, and 24 kHz), for improved quality for bitrates at or below 64 kbits/s, per channel. MPEG produced an updated version of the MPEG-2 Audio Working Draft, and is on track for achieving a Committee Draft specification by the November MPEG meeting. The MPEG-2 Audio multichannel coding Standard will provide backward-compatibility with the existing MPEG-1 Audio Standard (ISO/ IEC IS 11172-3). Together with ITU-RS, MPEG is organizing formal subjective testing of the proposed MPEG-2 multichannel audio codecs and up to three non backward compatible (NBC) codecs. The NBC codecs are included in order to determine whether an NBC mode should be introduced as an addendum to the standard. If the results show clear evidence that an NBC mode improves the performance, a formal call for NBC proposals will be issued by MPEG, with a view to incorporate these features in the audio syntax. MPEG-2 audio attempts to maintain as much compatibility with MPEG-1 audio syntax as possible, while adding discrete surround-sound channels to the original MPEG-1 limit of 2 channels (Left, Right or matrix center and difference). The main channels (Left, Right) in MPEG-2 audio will remain backwards compatible, whereas new coding methods and syntax will be used for the surround channels. A total of 5.1 channels are included that consist of the two main channels (L,R), two side/rear, center, and a 100 Hz special effects channel (hence the ".1" in "5.1"). At this time, non-backwards compatible (NBC) schemes are being considered as an ammedment to the MPEG-2 audio standard. One such popular system is Dolby AC-3. |
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MPEG is developing the MPEG-2 Systems Standard to specify coding formats for multiplexing audio, video, and other data into a form suitable for transmission or storage. There are two data stream formats defined: the Transport Stream, which can carry multiple programs simultaneously, and which is optimized for use in applications where data loss may be likely, and the Program stream, which is optimized for multimedia applications, for performing systems processing in software, and for MPEG-1 compatibility. Both streams are designed to support a large number of known and anticipated applications, and they retain a significant amount of flexibility such as may be required for such applications, while providing interoperability between different device implementations. The Transport Stream is well suited for transmission of digital television and video telephony over fiber, satellite, cable, ISDN, ATM, and other networks, and also for storage on digital video tape and other devices. It is expected to find widespread use for such applications in the very near future. The Program Stream is similar to the MPEG-1 Systems standard (ISO/ IEC 11172-1). It includes extensions to support new and future applications. Both the Transport Stream and Program Stream are built on a common Packetized Elementary Stream packet structure, facilitating common video and audio decoder implementations and stream type conversions. This is well-suited for use over a wide variety of networks with ATM/ AAL and alternative transports. In New York, MPEG completed definitions of the features, syntax, and semantics of the Transport and Program Streams, enabling product designers to proceed. Among other items, the Transport Stream packet length was fixed at 188 bytes, including the 4-byte header. This length is suited for use with ATM networks, as well as a wide variety of other transmission and storage systems. |
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What about the Grand Alliance ? The Grand Alliance was formed in May 1993 by seven organizations (AT&T Corp., General Instrument Corp. (GI), Massachusetts Institute of Technology (MIT), Philips Consumer Electronics, David Sarnoff research center, Thomson, Zenith Electronics Corp.) to evaluate technologies and to decide on key elements that will be at the heart of the best of the best HDTV system. The video compression and transport technologies selected by the Grand Alliance are based on the MPEG-2 standards. The scanning formats selected are focused primarily on computer-friendly progressive scanning, while offering and interlaced mode important to some broadcasters. They agreed to use the MPEG-2 Video and Systems syntax, including B-pictures. Both interlaced (1440 x 960 x 30 Hz) and progressive (1280 x 720 x 60 Hz) modes will be supported. The Alliance then had to settled upon a modulation (QAM or VSB), convolution (MS or Viterbi), and error correction (RSPC, RSFC) specification. Laboratory tests in early 1993 showed better performance for a variant of VSB modulation and broadcast and cable carriage of digital HDTV signals via 8-VSB and 16-VSB modulation were tested under field conditions in Charlotte, North Carolina USA. The audio technology selected is a six-channel, compact-disc-quality digital surround sound system. The last major technical decision, the broadcast and cable transmission subsystem, is expected in early 1994 following testing of competing technologies. The Grand Alliance, now called ATSC (Advanced Television Systems Committee), suggests positions to the Department of State for their use in international standards organizations. ATSC proposes standards to the Federal Communications Commission. On April 12, 1995, ATSC Members approved the Digital Television Standard for HDTV Transmission. |
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Integrated Error Robust Solutions for MPEG-2 / MPEG-4 Advanced Audio Coding Whenever digital data is transmitted in real-time, i.e. without the chance of re-transmission, whether in packet oriented networks (e.g. Internet) or in stream oriented networks (e.g. digital broadcast systems or mobile communication networks), the receiver of this digital data will have to cope with transmission errors. In compressed audio, decoding of corrupted bitstreams can lead to annoying artifacts that heavily reduce the audio quality. Those artifacts may even damage the listeners ears or electronic equipment. Thus, measures have to be taken to deal with such transmission errors. Four different approaches can be combined:
Error Detection for AAC
Error Concealment for AAC
For AAC an optimized concealment solution was developed. It uses a psychoacoustics based selection between insertion of shaped noise and predicted harmonics, enabling very effective improvement of perceived audio quality. Usually, only very minor artifacts are audible, even under critical error conditions. Read more about Error Concealment. Error Protection for AAC
Error Resilient AAC
Huffman Codeword Reordering (HCR) to avoid error propagation within spectral data
Using these tools it becomes possible to recover much more data from a corrupted AAC bitstream. Furthermore errors can be detected more easily and concealed more efficiently. The combination of error resilience techniques and unequal error protection leads to an error robust coding scheme with less additional overhead. Due to its flexibility, it is suited for a broad range of applications with different error characteristics of the transmission channels. |
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Fraunhofer IIS-A offers custom tailored software solutions for error detection and protection, error concealment and error resilience optimized for AAC. We offer the adaptation of these methods to the specific requirements of broadcasting and mobile communication systems or other error-prone transmission channels. Some of these solutions are already, or will become available as implementations on different DSP platforms. Simple Visual Profile
The MPEG-4 standard is designed to play the role of a "global multimedia language". Within its source coding parts, MPEG-4 embraces a multitude of natural and synthetic audio and video coding schemes and representations, providing a generic tool set for numerous applications for the transmission and storage of audiovisual signals. In addition, MPEG-4 introduces and supports completely new concepts of object-based user interactivity as well as a rich feature set to maintain a useful quality of service within error-prone channels. The MPEG-4 Simple Visual Profile provides efficient, error resilient coding of rectangular video objects, suitable for applications on mobile networks or any other low-rate video application, like transmitting video signals as programm-associated data via digital audio broadcasting systems or video streaming via the Internet. Video scenes with low complexity (e.g. interviews, news) require a few 10 kbps, using a small image size at a low frame rate (e.g. QCIF 176 x 144 @ 8.33 fps). Video scenes with higher complexity and larger picture sizes (e.g. CIF 352 x 288 @ 12.5 fps) require several 100 kbps to provide an acceptable image quality. Fraunhofer-IIS develops specific solutions for audiovisual applications, with a focus on the combination of low-rate high-quality sound (e.g. coded with MP3, MPEG-2 AAC, or compliant to several MPEG-4 Audio Profiles) with low-rate good-quality images. One example is the development of "picture radio" services for the digital broadcasting systems of WorldSpace (satellite radio for Africa, Asia and South America) and EU-147 DAB (terrestrial radio used in Europe). Here, the low-rate video signal is transmitted as programm-associated data channel synchronized to the main audio program. Besides system-specific work, Fraunhofer-IIS developed a special rate control for the video encoder, introducing some acceptable delay in order to maintain a good overall image quality at a given fixed frame rate. Fraunhofer-IIS develops and licenses encoder and decoder software for the MPEG-4 Simple Visual Profile, running on Unix and Windows platforms. Furthermore, Fraunhofer-IIS develops and licenses DSP-based MPEG-4 video solutions. DSPs handle signal processing tasks very efficiently and reliably within the I/O peripherals of a computer system, saving precious workload on such computer for any other of its data processing tasks. As one example, Fraunhofer-IIS developed an MPEG-4 Simple Visual Profile encoder using two Texas Instruments' TMS 320 C 62xx running at 200 MHz. Each DSP is equipped with 16 Mbyte of SDRAM memory. Fast FIFOs are used for the connection between the DSPs as well as for the connection between the video input interface and one of the DSPs. Composite and Y/C input for PAL, NTSC and SECAM are supported. In addition, an optional serial digital interface (SDI) is prepared. Software download and bitstream output are provided via an ISA compatible interface. This dual-DSP encoder provides the following performance:
Due to the increasing number of DSPs within the family of C6xxx DSPs, there are options for video encoders running on one low-cost low-power derivative as well as on one single high-performance platform (including audio encoding). |
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History
Without data reduction, digital audio signals typically consist of 16 bit samples recorded at a sampling rate more than twice the actual audio bandwidth (e.g. 44.1 kHz for Compact Disks). So you end up with more than 1.400 Mbit to represent just one second of stereo music in CD quality. By using MPEG audio coding, you may shrink down the original sound data from a CD by a factor of 12, without losing sound quality. Factors of 24 and even more still maintain a sound quality that is significantly better than what you get by just reducing the sampling rate and the resolution of your samples. Basically, this is realized by perceptual coding techniques addressing the perception of sound waves by the human ear. Using MPEG audio, one may achieve a typical data reduction of:
By exploiting stereo effects and by limiting the audio bandwidth, the coding schemes may achieve an acceptable sound quality at even lower bitrates. MPEG Layer-3 is the most powerful member of the MPEG audio coding family. For a given sound quality level, it requires the lowest bitrate - or for a given bitrate, it achieves the highest sound quality. Sound Quality
*) Fraunhofer uses a non-ISO extension of MPEG Layer-3 for enhanced performance ("MPEG 2.5") In all international listening tests, MPEG Layer-3 impressively proved its superior performance, maintaining the original sound quality at a data reduction of 1:12 (around 64 kbit/s per audio channel). If applications may tolerate a limited bandwidth of around 10 kHz, a reasonable sound quality for stereo signals can be achieved even at a reduction of 1:24. For the use of low bit-rate audio coding schemes in broadcast applications at bitrates of 60 kbit/s per audio channel, the ITU-R recommends MPEG Layer-3. (ITU-R doc. BS.1115) Details
The process to find the optimum gain and scalefactors for a given block, bit-rate and output from the perceptual model is usually done by two nested iteration loops in an analysis-by-synthesis way: Inner iteration loop (rate loop) The Huffman code tables assign shorter code words to (more frequent) smaller quantized values. If the number of bits resulting from the coding operation exceeds the number of bits available to code a given block of data, this can be corrected by adjusting the global gain to result in a larger quantization step size, leading to smaller quantized values. This operation is repeated with different quantization step sizes until the resulting bit demand for Huffman coding is small enough. The loop is called rate loop because it modifies the overall coder rate until it is small enough. Outer iteration loop (noise control/distortion loop) To shape the quantization noise according to the masking threshold, scalefactors are applied to each scalefactor band. The systems starts with a default factor of 1.0 for each band. If the quantization noise in a given band is found to exceed the masking threshold (allowed noise) as supplied by the perceptual model, the scalefactor for this band is adjusted to reduce the quantization noise. Since achieving a smaller quantization noise requires a larger number of quantization steps and thus a higher bitrate, the rate adjustment loop has to be repeated every time new scalefactors are used. In other words, the rate loop is nested within the noise control loop. The outer (noise control) loop is executed until the actual noise (computed from the difference of the original spectral values minus the quantized spectral values) is below the masking threshold for every scalefactor band (i.e. critical band). |
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The Institution of Electrical Engineers organised a one day colloquium in London entitled MPEG-2 - what it is and what it isn't in January 1995. The digest from the colloquium (Digest No: 1995/012) includes the following eight papers:
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