15/30 Mbit/s Universal Digital TV Codec Using a Median Adaptive Predictive Coding Method

This paper describes a 15/30 Mbit/s TV codec with a new approach to high-efficiency coding for TV signals, i.e., median adaptive prediction. The 15/30 Mbit/s codec, commonly applicable to NTSC, PAL, and SECAM (525/60 and 625/50) systems, uses adaptive prediction incorporating a motion-compensated interframe, an interfield, and an intrafield predictor. Its performance for digital transmission is presented. This universal codec is designed, based on CCIR recommendations concerning digital TV coding parameters for studios (Rec. 601) and general principles on long-distance digital TV transmission (Rec. 604). A field trial of 15 Mbit/s digital TV transmission using this codec between earth stations with a 30 m diameter antenna and a 5 m diameter antenna is reported.     

Picture Coding

A survey of recent German research in the field of picture coding is presented. The described coding methods are mainly based on extended differential pulse-code modulation (DPCM) techniques. A unified model of a feedback switched quantizer for picture coding is explained. Theoretical results show a 9-dB gain in SNR over that of DPCM. A simple realization is shown. For encoding the 1-MHz videotelephone signals, a two-stage coding system consisting of a two-dimensional DPCM in the first stage and a dot interlaced frame repeating codec in the second stage is described. A DPCM combined with a relevancy detector and runlength coder is used for coding the 5-MHz video signals of a high-resolution videotelephone. A special scanning technique assures compatibility with ordinary videotelephones. Good picture quality is obtained by coding the luminance and chrominance signals of color TV separately with DPCM and switched quantization into a 34 Mbit/s signal.     

32 Mbit/s Transmission of NTSC Color TV Signals by Composite DPCM Coding

A composite DPCM coding System was developed, which is capable of coding and transmitting an NTSC color TV signal without component separation at a 32.064 Mbit/s rate. The DPCM prediction method used is based on the algorithm that a luminance component prediction valuehat{y}and a carrier chrominance component prediction valuehat{c}are calculated individually, and then the composite signal prediction valuehat{x}is determined byhat{x} = hat{y} + hat{c}. In order to utilize horizontal blanking (HBL) intervals for transmitting active video signals,HBL signals are not transmitted in each line but their representative signals are transmitted once a frame during a vertical blanking interval. A dual word-length coding and quantizing method is adopted, which uses 4 bit and 8 bit words with average word-lengths of 4.4 bits/sample. Codec equipment was fabricated, and coding and transmitting experiments were conducted, using NTT's PCM-100 M digital repeatered line. Experimental results show that this composite DPCM coding system can be employed for digital transmission of NTSC color TV signals, such as color ITV signals, at a 32.064 Mbit/s rate.     

Simplified 6.3 Mbit/s Codec for Video Conferencing

A color TV codec with a 6.3 Mbit/s transmission bit-rate has been simplified by a newly developed coding method. This method is the combination of a 4-to-1 dot interlace system and 4 bit/pel DPCM. A digital filtering technique for reducing coding distortion is also employed. In comparison with an interframe codec having the same transmission bit-rate, the amount of codec circuitry is halved. Subjective tests were performed to investigate the coded picture quality. It was found that this codec can present acceptable pictures for video conferencing, and that the picture quality for rapidly moving subjects is the same as that of the interframe codec operating at the same bit-rate. A video conferencing system via a digital satellite link has been constructed for experiments by employing this codec. The experiments showed that the satellite link has a sufficiently low transmission error rate for the coded pieture transmission. Moreover, it has been proved that transmission delay does not pose any problems for normal conferences.     

Universal digital TV codec—unicodec

The UNICODEC, developed under the INTELSAT R & D Program, is a TV codec which can transmit and receive any standard signal of NTSC, PAL, or SECAM, by using 15 Mb/s or 30 Mb/s transmission rates (switchable) with two high quality sound channels and one voice circuit. These rates include rate 239/255 BCH FEC for video information. The codec uses the 4:2:2 digital interface for component coding based on CCIR Recommendation 601. The UNICODEC can provide the flexibility to deal with the composite coding now needed in mixed analogue/digital networks, but can also handle either composite NTSC, PAL, SECAM or MAC (multiplexed analogue components), or digital component signals (Y, R-Y, B-Y) in the future. The key technique used here is called MAP (median adaptive prediction) coding, which is one of the adaptive DPCM methods for accurate and simple control in high quality TV transmission; one prediction out of three (interframe, intrafield or interfield) is transmitted without the need for many control bits. There is also a refresh mode for burst error or synchronization recovery. The UNICODEC is expected to provide double capacity with higher quality pictures compared to conventional analogue transmission (FM in 18 MHz). Alternatively, smaller size antennas may be used. This paper describes the objectives of the development of the UNICODEC, its main features and results of field tests.     

A single-chip adaptive DPCM video codec

A differential pulse-code modulation (DPCM) video codec with two-dimensional intrafield prediction and adaptive quantizer is presented. An approach for the arithmetic implementation of the DPCM structure and the design of a test chip, fabricated in a 1.5 μm CMOS technology, is described. This is the first VLSI realization of a DPCM codec with adaptive quantizer. For the test chip transmitter or receiver mode, application as part of a three-dimensional interframe codec and processing of luminance or chrominance signals are optional. A line buffer and ten different quantizer characteristics are realized on-chip. Correct operation has been verified up to 26 MHz     

Time-Shared Codecs for Multichannel Delta Modulation and Differential PCM

Current practice for multichannel delta modulation (DM) terminals uses one DM codec per channel-end. This paper describes methods by which one high-speed DM codec can be timeshared over a large number of channels. The methods are also applied to multichannel differential PCM (DPCM) terminals. In both cases the time-shared codecs include step-size adaptation determined by the recent past history of the coded digital signal. A method for digital conversions between multichannel linear PCM and adaptive DPCM (ADPCM) formats is also described.     

Digital TV Receiver for NTSC Color TV Signals with Dual Word-Length DPCM Coding

A video codec system for carrying one broadcast quality NTSC color TV channel at a rate of 42.935 Mbits/s has been proposed. The receiver of the video codec system is developed for recovery of the original signal. The receiver accepts a 42.935 Mbits/s serial TDM data stream with a synchronous clock from the Transmitter. The receiver detects the unique sync code and inserts a horizontal blanking pattern which has been removed from the transmitting data. The audio and video data are separated at the outputs of the demultiplexer. The 4/8 bit dual length code word of the video data is smoothed out by a buffer memory and fed to the DPCM reconstruction loop. The analog NTSC color video signal as well as the audio signal are reconstructed after the D/A conversion. This paper describes the design and development of the receiver portion (Fig. 1) of the codec which is capable of transmitting'one color signal at broadcast standards on a T3 digital link.     

A codec for HDTV signal transmission through terrestrial and satellite digital links

A codec for digital transmission of HDTV is described. The bit-rate compression algorithm is based on advanced techniques such as spatial discrete cosine transform (DCT), temporal differential PCM (DPCM), variable length coding. The codec is designed to operate with both the interlaced studio systems 1125/60 and 1250/50, and, thanks to the inherent flexibility of the packet structure, a wide range of line bit-rates can be used as a compromise between video quality and bit-rate constraints of the digital transmission link. The flexibility of the HDTV codec is highlightened through examples of applications over satellite digital links in practical situations.     

QVGA/CIF Resolution MPEG-4 Video Codec Based on a Low-Power and General-Purpose DSP

This paper describes a quarter video graphic array/common intermediate format (QVGA/CIF) resolution MPEG-4 video codec based on a low-power, general-purpose digital signal processor (DSP) (NEC μPD77210, 160 MHz, 80 mW, 1.5 V). To enhance video codec performance, the codec employs fast algorithms, including, in motion estimation, a successive similarity detection algorithm (SSDA; a fast block matching) whose decision timing for termination of block matching is optimized. Further, the use of a software direct memory access (DMA) queue reduces the wasteful DSP wait cycles that can result from massive access to external frame memories. The resulting codec executes QVGA (320 × 240 pixels) × 15 fps codec, or CIF (352 × 288 pixels) × 15 fps encoding at 384 kbps, in real time, performance levels sufficient for next-generation wireless videotelephony.     

数字电视信号的发射与接收技术

系统地讨论了数字电视信号的发射及接收技术。稳定可靠的信号源与抗扰性能良好的传输方式是保证图像质量的前提,快速无失真的编解码是接收机的基本工作要求。只有从发射到传输再到接收均能保证高质量的工作,才能使数字电视信号真正地实现系统级的应用。文中分别对这3个环节进行了阐述,并展望了数字电视信号处理系统未来的发展方向     

16 Kb/s high quality voice encoding for satellite communication networks

This paper describes applications of adaptive predictive coding (APC) with maximum likelihood quantization (MLQ) which can cover a wide range of coding rates from 4.8 to 16 kb/s for low C/N satellite communication systems, such as maritime, aeronautical mobile and thin-route satellite communication systems, and also for speech and data integration, including digital circuit multiplication equipment (DCME) in business communication systems, such as INTELSAT business services (IBS). A 16 kb/s APC–MLQ hardware codec has been implemented by NEC–7720 DSP chips and the performance has been confirmed in subjective quality of speech through conversational tests. The objective performance has also been evaluated for non-voice signals, such as single and multi-frequency tones, and 1200 and 2400 b/s voiceband data signals. The APC-MLQ codec can transmit the voice-band data at 1200 b/s over two asynchronous tandem links and at 2400 b/s over one link. It was noted that the APC-MLQ codec is superior in speech performance at 16 kb/s to a narrow-band companded FM and meets requirements for low C/N satellite communication systems. For voice and data integration into 16 kb/s for 64 kb/s links, we propose a multi-media multiplexing for low C/N digital satellite communication systems and also a small-scale circuit multiplication system for business use. In these systems, a variable rate coding of APC-MLQ from 4.8 to 16 kb/s can be effectively introduced for voice and data integration.     

A Video Codec Transmitter

The transmitter of a video codec system is designed and realized in ECL hardware. This video codec system is capable of carrying one commercial NTSC color television channel with a program audio channel at a rate of 42.9 Mb/s. The transmitter interfaces with a DPCM processor. The transmitter removes the horizontal blanking interval from the composite television signal, converts the quantized difference signal from the DPCM processor into 4/8-bit dual length code and buffers the codes for constant rate transmission. A line synchronization code is multiplexed with audio and video codes to form a serial data link at 42.9 Mb/s. This can be transmitted on a T3 (44.7 Mb/s) digital carrier. The transmitter is composed of four functional units: controller, coder, buffer and multiplexer. This paper describes the design, layout, testing and evaluation of the high speed digital system that results in 2 to 1 data compression of the digital video.     

Design of a DPCM codec for VLSI realization in CMOS technology

Current perspectives on broad-band communication services have made the realization of a DPCM system for video coding on a single integrated circuit particularly important. A nonadaptive intraframe DPCM system is designed for reducing video transmission bit rate by a factor of two. All functional blocks of a DPCM codec have been specified, and modifications have been investigated for reducing speed requirements. Alternative realizations of functional blocks, e.g., adders, subtractors, table look-up operations, are compared with respect to speed by a simple delay model. A one-chip VLSI implementation of an efficient DPCM codec will be possible with a 2-µm CMOS technology.     

Low-complexity subband encoding for HDTV images

The transmission of high-definition television (HDTV) signals on available digital networks and satellites requires the adoption of sophisticated compression techniques to limit the bit rate requirements and to provide high-quality and reliable service to customers. For processing and transmission of image signals, a low-complexity codec without visible degradation is desired. A low-complexity intraframe subband image coding algorithm is developed. The low band is DPCM encoded and the high bands are PCM encoded. An efficient entropy coder is designed which reduces the overall bit rate significantly. It is shown that high-quality HDTV images can be obtained at as low a bit rate as 45 Mb/s or less with a very low-complexity encoder. For dividing the image into subbands, a new class of quadrature mirror filters (QMFs) called generalized quadrature mirror filters (GQMFs) is used for filtering. Performance is also evaluated by using short kernel filters (SKFs), which are easy to implement and require very few computations     

Embedded DPCM for Variable Bit Rate Transmission

Within the bit stream of an embedded digital code is a stream that can be decoded to produce a reasonable replica of the analog source signal. Unlike pulse code modulation (PCM), differential PCM (DPCM), is not an embedded code. IfCbits/sample are delected from the bit stream of a DPCM encoder withEbits/sample, the decoded analog signal is substantially noisier than the output of a DPCM codec withD = E - Cbits/sample. The penalty is 4-10 dB in signal-to-noise ratio (snr). However, with minor modifications to the encoder and decoder, DPCM becomes an embedded code. Embedded DPCM withEbits/ sample at the encoder andDbits/sample transmitted produces exactly the same output as embedded DPCM withDbits/sample encoding and perfect transmission. The snr of embedded DPCM is slightly lower than the Snr of DPCM. The penalty is 0.5-0.8 dB if the minimum transmitted bit rate is 2 bits/sample. It is less than 0.3 dB ifDis at least 3 bits/sample. Combined with an appropriate adaptive quantizer the embedded DPCM codec produces embedded ADPCM (adaptive DPCM) for variable rate transmission ranging from 2 bits/sample up to any desired maximum. Applications exist in speech interpolation, packet switching, and hardware architecture.     

A Voiceband Codec with Digital Filtering

Oversampling and digital filtering have been used to design a per-channel voiceband codec with resolution that exceeds the typical transmission system requirement by more than 15 dB. This extended dynamic range will allow for the use of digital processing in the management of signal levels and system characteristics in many telecommunication applications. Digital filtering contained in the codec provides rejection of out-of-band inputs and smoothing of the analog output that is sufficient to eliminate the need for analog filtering in most telephone applications. Some analog filtering may be required only to maintain the expanded dynamic range in cases where there is a danger of large amounts of out-of-band energy on the analog input impairing the dynamic range of the modulator. The encoder portion of the oversampled codec comprises an interpolating modulator that samples at 256 kHz followed by digital filtering that produces a 16-bit PCM code at a sample rate of 8 kHz. In the decoder, digital processing is used to raise the sampling rate to 1 MHz prior to demodulation in a 17-level interpolating demodulator. The circuits in the codec are designed to be suitable for large-scale integration. Component matching tolerances required in the analog circuits are of the order of only ± 1 percent, While the digital circuits can be implemented with fewer than 5000 gates with delays on the order of 0.1 μs. In this paper the response of the codec is described mathematically and the results are confirmed by measurements of experimental breadboard models.     

A single-chip CMOS analog/digital mixed NTSC decoder

A single-chip CMOS LSI that integrates all analog-to-digital (A/D), digital-to-analog (D/A), peripheral, and digital signal processing circuits necessary for a digital National Television System Committee (NTSC) signal decoder is described. The LSI chip accepts composite NTSC video signals in analog form, digitizes them using the on-chip A/D converter, converts them to component RGB signals, and then converts the signals to analog form by using the on-chip D/A converters. The development of circuits that maximize use of the input digital data is discussed. A 6-b A/D circuit is used to reduce the circuit size. Circuits that help maintain acceptable picture quality despite 6-b resolution were developed. Besides analog NTSC signal input and RGB signal output, the IC can also input and output digital NTSC signals, Y/C (luminance, chrominance) signals, and RGB signals. Applications of the LSI are presented