In this paper, we introduce an image coding method which unifies two image coding techniques: variable-length transform coding (VLTC) [1], [2] and image-adaptive vector quantization (IAVQ) [3]. In both VLTC and IAVQ, ...
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In this paper, we introduce an image coding method which unifies two image coding techniques: variable-length transform coding (VLTC) [1], [2] and image-adaptive vector quantization (IAVQ) [3]. In both VLTC and IAVQ, the image is first decomposed into a set of blocks. VLTC encodes each block in the transform domain very efficiently;however, it ignores the interblock correlation completely. IAVQ addresses the interblock correlation by using a codebook generated from a subset of the blocks to vector-quantize all blocks. Although the resulting codebook represents the input image better than a universal codebook generated from a large number of training images, it has to be transmitted separately as an overhead, therefore degrading the coding performance at high bit rates. In the unified IAVQ-VLTC, we perform IAVQ for each input image in the block transform domain. For each transform block, we transmit either the closest codeword or the corresponding label. The number of codewords and labels transmitted is exactly equal to the number of transform blocks coded, implying that the overhead problem introduced by IAVQ disappears. VLTC is used for codeword encoding. A further reduction in the bit rate required for transmitting the labels is accomplished by using a technique called codebook self-organization and by exploiting the similarity between the neighboring blocks. For a special case, it is shown theoretically that IAVQ-VLTC always performs as well as, or better than, VLTC. The simulation results demonstrate that IAVQ-VLTC outperforms VLTC for bit rates up to 0.6 bits/pixel.
An exact derivation of an optimal lapped orthogonal transform (LOT) is presented. The optimal LOT is related to the discrete cosine transform (DCT) in such a way that a fast algorithm for a nearly optimal LOT is deriv...
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An exact derivation of an optimal lapped orthogonal transform (LOT) is presented. The optimal LOT is related to the discrete cosine transform (DCT) in such a way that a fast algorithm for a nearly optimal LOT is derived. Compared to the DCT, the fast LOT requires about 20-30% more computations, mostly additions. An image coding example demonstrates the effectiveness of the LOT in reducing blocking effects; the LOT actually leads to slightly smaller signal reconstruction errors than does the DCT.< >
Image transform coding is first briefly reviewed using conventional viewpoints. Then a new spatial domain interpretation is given to image transform coding. An improvement based on this viewpoint for the Fourier trans...
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Image transform coding is first briefly reviewed using conventional viewpoints. Then a new spatial domain interpretation is given to image transform coding. An improvement based on this viewpoint for the Fourier transform coding, which possesses simple spatial domain relations, is presented.
An adaptive cosine transform coding scheme for color images which incorporates human visual properties into the coding scheme is described. It employs adaptive quantization to exploit the statistical nature of the coe...
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An adaptive cosine transform coding scheme for color images which incorporates human visual properties into the coding scheme is described. It employs adaptive quantization to exploit the statistical nature of the coefficients and adaptive block distortion equalization to reduce the block edge structures inherent in block transform coding schemes. Results show that the subjective quality of the reconstructed images at a bit rate of 0.4 bit/pixel or a compression ratio of 60:1 is very good.< >
A new hybrid coding method for transmitting videoconferencing images at a bit rate 384 kbits/s is proposed. Considering the characteristics of motion-compensated interframe prediction errors for typical videoconferenc...
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A new hybrid coding method for transmitting videoconferencing images at a bit rate 384 kbits/s is proposed. Considering the characteristics of motion-compensated interframe prediction errors for typical videoconferencing scenes, a filter is introduced to separate pulsive components on which conventional discrete cosine transform (DCT) coding method does not work well. These separated pulsive components are coded by using scalar quantization (SQ). The remainder are DCT coded. For DCT coefficients, an adaptive coding method based on the classification of DCT coefficients is applied in order to improve the coding performance. Since the proposed method employs both DCT coding and SQ of prediction errors, it is named "DCT/SQ coding method." Experimental results show that the DCT/SQ coding method is effective to reduce so-called mosquito effects, and thus it can improve the quality of decoded images.
A two-dimensional, 32nd-order orthogonal transform coding system which performs three types of transformations in realtime for NTSC color television signals has been developed. This paper reports on a hardware realiza...
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A two-dimensional, 32nd-order orthogonal transform coding system which performs three types of transformations in realtime for NTSC color television signals has been developed. This paper reports on a hardware realization of this system, the optimum bitassignments for 32 and 22 Mbit/s transmission rates and the effects of bit errors during transmission. Through the evaluation of picture quality, the system gave a fine quality of pictures after compressing an 8-bit digitized television signal into 3 bits/pel corresponding to 32 Mbit/s transmission, and a passable quality of pictures by 2.25 bits/ pel corresponding to 22 Mbit/s transmission.
A video compression scheme based on the wavelet representation and multiresolution motion compensation (MRMC) is presented in this paper. The multiresolution/multifrequency nature of the discrete wavelet transform is ...
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A video compression scheme based on the wavelet representation and multiresolution motion compensation (MRMC) is presented in this paper. The multiresolution/multifrequency nature of the discrete wavelet transform is an ideal tool for representing images and video signals. Wavelet transform decomposes a video frame into a set of subframes with different resolutions corresponding to different frequency bands. These multiresolution frames also provide a representation of the global motion structure of the video signal at different scales. The motion activities for a particular subframe at different resolutions are different but highly correlated since they actually specify the same motion structure at different scales. In the multiresolution motion compensation approach, motion vectors in higher resolution are predicted by the motion vectors in the lower resolution and are refined at each step. In this paper, we propose a variable block-size MRMC scheme in which the size of a block is adapted to its level in the wavelet pyramid. This scheme not only considerably reduces the searching and matching time but also provides a meaningful characterization of the intrinsic motion structure. The variable block-size MRMC approach also avoids the drawback of the constant-size MRMC in describing small object motion activities. After wavelet decomposition, each scaled subframe tends to have different statistical properties. An adaptive truncation process was implemented and a bit allocation scheme similar to that In the transform coding is examined by adapting to the local variance distribution in each scaled subframe. Based on the wavelet representation, variable block-size MRMC approach and a uniform quantization scheme, four variations of the proposed motion-compensated wavelet video compression system are identified. It is shown that the motion-compensated wavelet transform coding approach has a superior performance in terms of the peak-to-peak signal-to-noise ratio as well as
A hi-fi audio CODEC with an improved adaptive transform coding (ATC) algorithm is developed using digital signal processors (DSP's). An audio signal with a 20 kHz bandwidth sampled at 48 kHz is coded at a rate of ...
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A hi-fi audio CODEC with an improved adaptive transform coding (ATC) algorithm is developed using digital signal processors (DSP's). An audio signal with a 20 kHz bandwidth sampled at 48 kHz is coded at a rate of 128 kb/s. The algorithm utilizes adaptive block size selection, which is effective for pre-echo suppression. A modified discrete cosine transform (MDCT) with a simple window set is employed to reduce block boundary noise without decreasing the performance of transform coding. In addition, a fast MDCT calculation algorithm, based on a fast Fourier transform, is adopted. Weighted bit allocation is employed to quantize the transformed coefficients. The CODEC is realized by a multiprocessor system composed of newly developed DSP boards. Subjective tests with the CODEC show that the coding quality is comparable to that of compact disc signals.
Since differential-pulse-code modulation (DPCM) and orthogonal transform coding (OTC) are the most fundamental methods of high-efficiency coding (bit-reduction method), it is important to clarify the basic coding char...
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Since differential-pulse-code modulation (DPCM) and orthogonal transform coding (OTC) are the most fundamental methods of high-efficiency coding (bit-reduction method), it is important to clarify the basic coding characteristics of these methods and the difference between them in order to utilize the bit-reduction method effectively. This paper theoretically as well as experimentally compares the coding efficiency of a DPCM having a two-dimensional predictor with that of a two-dimensional Hadamard transform coding method (HTC) in the intrafield coding of the NTSC composite signal. The comparison evidenced that the distinctive difference in coding characteristics between DPCM and HTC depends greatly on the power level of carrier chrominance signals. That is, it is confirmed theoretically and experimentally that the coding efficiency of the HTC is far lower than that of the DPCM in the case of a signal having a high power level carrier chrominance signal such as a color-bar signal.
This paper discusses speech coding systems based upon transform coding (TC). It compares several transforms and shows that the cosine transform leads to a nearly optimum performance for almost all speech sounds. Vario...
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This paper discusses speech coding systems based upon transform coding (TC). It compares several transforms and shows that the cosine transform leads to a nearly optimum performance for almost all speech sounds. Various adaptive coding strategies are then investigated, and a coding scheme is proposed that is based on a nonadaptive discrete cosine transform (DCT), on an adaptive bit assignment, and on adaptive quantization. The adaptation is controlled by a short-term basis spectrum that is derived from the transform coefficients prior to coding and transmission and that is transmitted as side information to the receiver. The main result is that this adaptive transform coder performs better than all known nonpitch-tracking coding schemes; it extends the range of speech waveform coding to lower bit rates and closes the gap between vocoders and predictive waveform coders.
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