Nikil S. Jayant

Three-dimensional subband coding of video

We describe and show the results of video coding based on a three-dimensional (3-D) spatio-temporal subband decomposition. The results include a 1-Mbps coder based on a new adaptive differential pulse code modulation scheme (ADPCM) and adaptive bit allocation. This rate is useful for video storage on CD-ROM. Coding results are also shown for a 384-kbps rate that are based on ADPCM for the lowest frequency band and a new form of vector quantization (geometric vector quantization (GVQ)) for the data in the higher frequency bands. GVQ takes advantage of the inherent structure and sparseness of the data in the higher bands. Results are also shown for a 128-kbps coder that is based on an unbalanced tree-structured vector quantizer (UTSVQ) for the lowest frequency band and GVQ for the higher frequency bands. The results are competitive with traditional video coding techniques and provide the motivation for investigating the 3-D subband framework for different coding schemes and various applications.

A low-delay CELP coder for the CCITT 16 kb/s speech coding standard

A low-delay code-excited linear prediction (LD-CELP) speech coder which is expected to be standardized in 1992 as a CCITT G Series Recommendation for universal applications of speech coding at 16 kb/s is presented. The coder achieves a one-way coding delay of less than 2 ms by making both the LPC predictor and the excitation gain backward-adaptive and by using a small excitation vector size of five samples. The official CCITT laboratory tests revealed that the speech quality of this 16 kb/s LD-CELP coder is either equivalent to or better than that of the CCITT G.721 standard 32-kb/s ADPCM coder for almost all conditions tested. A description of the LD-CELP algorithm, its implementation on the DSP32C for CCITT testing, and performance results from these tests are presented. >

Signal compression: technology targets and research directions

A description of technology targets in signal compression and a nonexhaustive account of research directions that may lead toward these targets are presented. Opportunities for integrating source coding and channel coding technologies are also pointed out. Such integration, which has hitherto been an informal exercise, will become increasingly essential as communication capabilities are stretched with capacity-limited channels such as wireless media. In parallel, as greater sophistication is sought in the integration of speech and data with broadband signals such as CD-audio and high-resolution video, there will be increased interaction of signal compression technology with the field of communication networking. >

Postprocessing for very low bit-rate video compression

This paper presents a novel postprocessing algorithm developed specifically for very low bit-rate MC-DCT video coders operating at low spatial resolution, postprocessing is intricate in this situation because the low sampling rate (as compared to the image feature size) makes it very easy to overfilter, producing excessive blurring. The proposed algorithm uses pixel-by-pixel processing to identify and reduce both blocking artifacts and mosquito noise while attempting to preserve the sharpness and naturalness of the reconstructed video signal and minimize the system complexity. Experimental results show that the algorithm successfully reduces artifacts in a 16 kb/s scene-adaptive coder for video signals sampled at 80 x 112 pixels per frame and 5-10 frames/s. Furthermore, the portability of the proposed algorithm to other block-DCT based compression systems is shown by applying it, without modification, to successfully post-process a JPEG-compressed image.

Enhancement of ADPCM speech coding with backward-adaptive algorithms for postfiltering and noise feedback

It is shown that postfiltering circuits based on higher order LPC (linear predictive coding) models can provide very low distortion in terms of special tilt. Thus, they can provide better speech enhancement than circuits based on the backward-adaptive pole-zero predictor in ADPCM (adaptive digital pulse code modulation). Quantitative criteria for designing postfiltering circuits based on higher-order LPC models are discussed. These postfilters are particularly attractive for systems where high-order LPC analysis is an integral part of the coding algorithm. In a subjective test that used a computer-simulated version of these circuits, enhanced ADPCM obtained a mean opinion score of 3.6 at 16 kb/s. >

New directions in subband coding

Two very different subband coders are described. The first is a modified dynamic bit-allocation-subband coder (D-SBC) designed for variable rate coding situations and easily adaptable to noisy channel environments. It can operate at rates as low as 12 kb/s and still give good quality speech. The second coder is a 16-kb/s waveform coder, based on a combination of subband coding and vector quantization (VQ-SBC). The key feature of this coder is its short coding delay, which makes it suitable for real-time communication networks. The speech quality of both coders has been enhanced by adaptive postfiltering. The coders have been implemented on a single AT&T DSP32 signal processor. >

Adaptive postprocessing algorithms for low bit rate video signals

A new adaptive postprocessing algorithm to enhance the quality of a noisy video sequence is presented. The algorithm recognizes that the visibility of noise depends on local signal characteristics. It therefore classifies the video signal into different classes and uses separate nonlinear filters matched to each class. The most general version of the algorithm employs motion-compensated frame averaging to improve picture quality in a first stage. A classification algorithm subsequently divides subblocks of pixels in the averaged frame into four classes: edge, smooth, nonsmooth with motion and nonsmooth without motion. Spatial algorithms that perform multilevel median filtering, double median filtering, and median filtering are used for pixels belonging to edge, smooth, and nonsmooth with motion categories. Pixels in the nonsmooth, unmoving category are left unfiltered to preserve corresponding image texture. In a simpler version of this four-class system, the motion cues and motion-compensated frame averaging are eliminated, and the purely spatial filtering is based on a three-class algorithm. When used at the output of a 3-D subband coder at 384 kbps, the spatial postfilter was shown to provide a consistent gain in subjectively evaluated picture quality. Twenty-five viewers participated in an experiment involving three coded sequences. In a pairwise comparison of postfiltered and unfiltered sequences, the postfiltered version was judged to be better in 63 out of 75 instances.

Coding of wideband speech

Abstract The technologies of ISDN teleconferencing, CD-ROM multimedia services, and High Definition Television are creating new opportunities and challenges for the digital coding of wideband audio signals, wideband speech in particular. In the coding of wideband speech, an important point of reference is the CCITT standard for 7 kHz speech at a rate of 64 kbit/s. Results of recent research are pointing to better capabilities — higher signal bandwidth at 64 kbit/s, and 7 kHz bandwidth at lower bit-rates such as 32 and 16 kbit/s. The coding of audio with a signal bandwidth of 20 kHz is receiving significant attention due to recent activity in the ISO (International Standards Organization), with a goal of storing a CD-grade monophonic audio channel at a bit-rate not exceeding 128 kbit/s. Prospects for accomplishing this are very good. As a side result, emerging algorithms will offer very attractive options at lower rates such as 96 and 64 kbit/s. As we address new challenges in wideband speech technology, several strides in coding research are likely to occur. Among these are refinements of existing models for auditory noise-masking, and a unification of linear prediction and frequency-domain coding.

Speech coding with time-varying bit allocations to excitation and LPC parameters

The authors explore the benefits of time-varying bit allocation to excitation and LPC (linear predictive coding) parameters for the case of codebook-excited LPC. The overall bit rate in the experiment was 4.8, 6.4, or 8.0 kb/s. In each case, permissible bit rates for the LPC component were 0, 24, 36, or 48 bits per frame, one of which was selected for each speech frame using a brute-force search maximum performance. Average SNR gains over conventional time-invariant methods were modest, on the order of 1 to 2 dB, but gains for certain speech segments were as high as 3 to 5 dB. Perceptually, gains due to variable bit allocation were most noticeable in the 6.4 kb/s system, especially with female speakers. However, even in this case, the benefits of flexible bit allocation were somewhat offset by distortions due to other inadequacies in the coding algorithm.<<ETX>>

Adaptive postfiltering of 16 kb/s-ADPCM speech

Adaptive differential PCM (ADPCM) is known to provide high quality digitization of telephone bandwidth speech at 32 kb/s. For ADPCM systems operating at sub-standard rates such as 24 or 16 kb/s, a postfiltering technique has been shown to provide a simple means for enhancing speech quality [1]. This paper proposes an algorithm which adapts the degree of postfiltering to the local performance of the ADPCM coder. The adaptation requires no extra information from the transmitter. As a result of the adaptation, segments of speech which are reproduced relatively well by the coder are only mildly postfiltered. At 16 kb/s, the adaptively postfiltered system provides good communications quality with most telephone speech inputs.