Showing papers on "Upsampling published in 2002"

Context-driven fusion of high spatial and spectral resolution images based on oversampled multiresolution analysis

Abstract: This paper compares two general and formal solutions to the problem of fusion of multispectral images with high-resolution panchromatic observations. The former exploits the undecimated discrete wavelet transform, which is an octave bandpass representation achieved from a conventional discrete wavelet transform by omitting all decimators and upsampling the wavelet filter bank. The latter relies on the generalized Laplacian pyramid, which is another oversampled structure obtained by recursively subtracting from an image an expanded decimated lowpass version. Both the methods selectively perform spatial-frequencies spectrum substitution from an image to another. In both schemes, context dependency is exploited by thresholding the local correlation coefficient between the images to be merged, to avoid injection of spatial details that are not likely to occur in the target image. Unlike other multiscale fusion schemes, both the present decompositions are not critically subsampled, thus avoiding possible impairments in the fused images, due to missing cancellation of aliasing terms. Results are presented and discussed on SPOT data.

Context-Driven Fusion of High Spatial and Spectral Resolution Images Based on Oversampled

Abstract: This paper compares two general and formal solutions to the problem of fusion of multispectral images with high-resolution panchromatic observations. The former exploits the undecimated discrete wavelet transform, which is an octave bandpass representation achieved from a conventional discrete wavelet transform by omitting all decimators and upsampling the wavelet filter bank. The latter relies on the generalized Laplacian pyramid, which is another oversampled structure obtained by recursively subtracting from an image an expanded decimated lowpass version. Both the methods selectively perform spatial-fre- quencies spectrum substitution from an image to another. In both schemes, context dependency is exploited by thresholding the local correlation coefficient between the images to be merged, to avoid injection of spatial details that are not likely to occur in the target image. Unlike other multiscale fusion schemes, both the present decompositions are not critically subsampled, thus avoiding possible impairments in the fused images, due to missing cancellation of aliasing terms. Results are presented and discussed on SPOT data.

Underwater telemetry method using doppler compensation

Abstract: A method for Doppler compensation in a phase coherent underwater communications system includes the steps of receiving a communications signal from a plurality of underwater communications channels, where the signal consists of a sequence of raw data; then applying a Doppler estimation to the data; demodulating the raw data; low pass filtering the demodulated raw data; applying resampling and downsampling to the data at a fractional sampling rate, for n samples per symbol, and generating a synchronized, Doppler-phase-corrected output signal; then applying the synchronized, Doppler-phase-corrected fractionally sampled signal to a phase locked loop and equalizer; then comparing an estimated symbol with a decision symbol and updating a plurality of tap coefficients; and h) outputting decision symbols. The Doppler estimation is performed either by applying a concurrent sinusoidal signal in a different band than the band of the communications signal, and then applying a Fourier transform or adaptive spectral estimation over a duration larger that a packet length to thereby obtain the Doppler-shift frequency f cd ; or, by applying a beginning probe signal; applying an ending probe signal; applying a probe replica signal; cross-correlating the probe replica signal with the beginning probe signal and the ending probe signal using a matched filter processor; determining a difference of arrival time between the beginning and ending probe signals using the peaks of the cross-correlation outputs; taking a ratio of the difference of arrival time to a time difference of the beginning and ending probe signals to obtain a dilation/compression ratio due to the Doppler shift; and then obtaining the Doppler shift from the dilation/compression ratio.

Digital sample rate converter architecture

Abstract: A digital sample rate converter converts a digital input signal (Din) having a first sample rate (Fs_in) to a corresponding digital output signal Dout having a second sample rate (Fs_out), wherein an upsampling circuit (3) upsamples the digital input signal (Din) by a factor of N and a feedback algorithm circuit (23A) receives a corresponding digital signal of the same sample rate (Fs_in*N) to produce a digital signal (X6) having a sample rate which is a second predetermined factor (M) times the second sample rate (Fs_out). That signal is filtered by a decimation filter (17) and then downsampled by a predetermined factor to produce the digital output signal (Dout) with the second sample rate (Fs_out).

Phased subarray processing for underwater 3D acoustic imaging

Abstract: 3D sonar imaging using a fully-populated rectangular 2D array has many promising applications for underwater imaging. A primary limitation of such systems is the large number of parallel front-end hardware channels needed to process the signals in transmit and receive when using conventional full phased array imaging. A subaperture beam acquisition and image formation process is presented that significantly reduces the number of front-end hardware channels while achieving image quality approaching that of full phased array imaging. Rather than transmitting and receiving on all N/spl times/N transducer elements to form each beam, an M/spl times/M subset of elements - called a subarray - is used for each firing. The limited number of front-end processing channels are used to acquire data from each subarray. Switching hardware allows the subarray to be multiplexed across the full array. Due to the Nyquist sampling criteria in beamspace, the number of beams acquired by each subarray can be significantly reduced compared to the number required for the full array. The phased subarray processing includes beam upsampling, lateral interpolation with a subarray-dependent filter, and coherent weighting and summation of all subarray images to form a high resolution image. The phased array method achieves an image quality nearing that of full phased array imaging with significantly fewer processing channels, slightly reduced SNR, and roughly three times the number of firings for reasonable configurations.

A unified approach to restoration, deinterlacing and resolution enhancement in decoding MPEG-2 video

Abstract: The quality and spatial resolution of video can be improved by combining multiple pictures to form a single superresolution picture. We address the special problems associated with pictures of variable but somehow parameterized quality such as MPEG-decoded video. Our algorithm provides a unified approach to restoration, chrominance upsampling, deinterlacing, and resolution enhancement. A decoded MPEG-2 sequence for interlaced standard definition television (SDTV) in 4:2:0 is converted to: (1) improved quality interlaced SDTV in 4:2:0; (2) interlaced SDTV in 4:4:4; (3) progressive SDTV in 4:4:4; (4) interlaced high-definition TV (HDTV) in 4:2:0; (5) progressive HDTV in 4:2:0. These conversions also provide features such as freeze frame and zoom. The algorithm is mainly targeted at bit rates of 4-8 Mb/s. The algorithm is based on motion-compensated spatial upsampling from multiple images and decimation to the desired format. The processing involves an estimated quality of individual pixels based on MPEG image type and local quantization value. The mean-squared error (MSE) is reduced, compared to the directly decoded sequence, and annoying ringing artifacts, including mosquito noise, are effectively suppressed. The superresolution pictures obtained by the algorithm are of much higher visual quality and have lower MSE than superresolution pictures obtained by simple spatial interpolation.

On the design and multiplier-less realization of digital IF for software radio receivers with prescribed output accuracy

Abstract: This paper studies the design and multiplier-less realization of the digital IF in software radio receivers The new architecture consists of a compensator for compensating the passband droop of the conventional cascaded integrator and comb (CIC) filter The passband droop is improved by a factor of four and it can be implemented with four additions using the sum-of-powers-of-two (SOPOT) coefficients The decimation factor of the multistage decimator is also reduced so that its output can be fed directly to the Farrow structure for sample rate conversion (SRC), eliminating the need for another L-band filter for upsampling By so doing, the programmable FIR filter can be replaced by a half-band filter placed immediately after the Farrow structure As the coefficients of this half-band filter, the multistage decimators and the subfilters in the Farrow structure are constants, they can be implemented without multiplication using SOPOT coefficients As a result, apart from the limited number of multipliers required in the Farrow structure, the entire digital IF can be implemented without any multiplications A random search algorithm is employed to minimize the hardware complexities of the proposed IF subject to a given specification in the frequency domain and prescribed output accuracy, taking into account signal overflow and round-off noise Design results are given to demonstrate the effectiveness of the proposed method

A low peak-to-average ratio quiescent mode architecture and signal design method for DMT modems

Abstract: A physical layer is presented which maps an all zero bit stream to a DMT symbol with low PAR. This allows the AFE to save power while the physical layer of the modem operates identically for both Showtime and Idle data. A minimum amount of changes relative to the existing DMT transmitter architecture are required, most notably the removal of a scrambler from the physical layer and the addition of an XOR mapping ( 17 ) and symbol inverter ( 25 ). A method is presented for designing the XOR mapping which takes into account upsampling and filtering in the AFE.

Time discrete filter comprising upsampling, sampling rate conversion and downsampling stages

Abstract: A time discrete filter comprises a sampling rate converter provided with an input and an output, and a down-sampler having a down-sampling factor nd. The time discrete filter further comprises an up-sampler having an up-sampling factor nu, whereby the up-sampler is coupled to the converter input, and the converter output is coupled to the down-sampler. It has been found that if a sampling rate conversion operation is preceded by an up-sampling operation and only after the conversion is followed by a down-sampling operation to a wanted sampling frequency, that then the complexity in terms of the ultimate number of calculations, in particular multiplications and additions, is reduced. This leads to a decrease of the number of instructions per second which is a measure for the complexity of a Digital Signal Processing (DSP) algorithm. In addition this leads to an associated decrease of power consumed by a DSP, such as applied in for example audio, video, and (tele)communication devices, as well as radio and television apparatus.

On the design and multiplier-less realization of digital IF for software radio receivers

Abstract: This paper proposes to reduce the decimation factor of the multistage decimator so that its output can be fed directly to the Farrow structure for sample rate conversion, eliminating the need for another L-band filter for upsampling. Furthermore, it was found out that the programmable FIR filter can be replaced by a half-band filter placed immediately after the Farrow structure, i.e. after sample rate conversion. This significantly reduces the complexity of the proposed software radio receiver because this half-band filter, which consists of fixed filter coefficients, can be implemented efficiently without multiplication using SOPOT coefficients. As the coefficients of the multistage decimators and the subfilters in the Farrow structure are also fixed, they can also be implemented efficiently using the SOPOT coefficients. As a result, apart from the limited number of multipliers required in the Farrow structure, the entire digital IF can be implemented without any multiplication. Design example is given to demonstrate the effectiveness and feasibility of the proposed approach.

PAR reduction for DMT systems with unloaded subchannels

Abstract: The inverse discrete Fourier transform used by discrete multitone modulation produces transmitted signals with a large peak-to-average ratio. Peak-to-average ratio reduction methods allow the transmitter to reduce the probability of clipping, or maintain the same probability of clipping while reducing the dynamic range requirements of the analog front end. In this paper we propose a method of peak-to-average ratio reduction that uses unloaded subchannels to prevent clipping in the transmitted signal. Considerations such as efficient adaptation methods and extensions to compensate for upsampling and filtering are addressed.

Look up table based upsampling and digital filtering

Abstract: A system and method are disclosed for processing a signal that includes receiving encoded data including a plurality of input symbols; generating an input symbol set from the plurality of input symbols; looking up a plurality of output samples for the input symbol set; and outputting the plurality of output samples.

Comparison of morphological pyramids for multiresolution MIP volume rendering

Abstract: We recently proposed a multiresolution representation for maximum intensity projection (MIP) volume rendering based on morphological adjunction pyramids which allow progressive refinement and have the property of perfect reconstruction. In this algorithm the pyramidal analysis and synthesis operators are composed of morphological erosion and dilation, combined with dyadic downsampling for analysis and dyadic upsampling for synthesis. Here we introduce an alternative pyramid scheme in which a morphological opening instead of an erosion is used for pyramidal analysis. As a result, the approximation accuracy when rendering from higher levels of the pyramid is improved.

The CA-CMAC for downsampling image data size in the compressive domain

Abstract: The CA-CMAC for downsampling image data size in the compressive domain is proposed in this paper. When the transmitting data is limited, it can reduce the bit rate during transmitting image data and decrease computations per pixel during the reconstructive process. The proposed method maps the image data into the CMAC lookup table, which can learn the characteristics of original image and can change image data size during downsampling and upsampling processes. It is unlike the conventional linear interpolation method, which gets lower SNR and costs more computation in the compression and reconstructive processes. The CA-CMAC method uses only a few hypercubes to learn the characteristics of original image, and transmits the learned characteristics to the receiver for reconstruction. Finally, the proposed method is applied to downsample JPEG data size in this paper, and it is shown that it gets high SNR after reconstruction.

Comparison of Morphological Pyramids for Multiresolution MIP Volume Rendering

Abstract: We recently proposed a multiresolution representation for maximum intensity projection (MIP) volume rendering based on morphological adjunction pyramids which allow progressive refinement and have the property of perfect reconstruction. In this algorithm the pyramidal analysis and synthesis operators are composed of morphological erosion and dilation, combined with dyadic downsampling for analysis and dyadic upsampling for synthesis. Here we introduce an alternative pyramid scheme in which a morphological opening instead of an erosion is used for pyramidal analysis. As a result, the approximation accuracy when rendering from higher levels of the pyramid is improved.

Theory of Subband Decomposition

Abstract: One of the most practical and successful applications of multirate filters is in video or audio compression using subband coding. Consider an example of audio subband coding shown in Figure 5.1.1. Let us say that one needs to sample the signal at a 10-kHz rate which corresponds to a bandwidth of 5 kHz.* If we need an accuracy of 16 bits, then we need to transmit or store a total of 160 Kb/s. As shown in Figure 5.1.1, in practice, the energy near the cutoff frequency is very small. Hence, we can take the time signal and, using multirate filters, split the signal into two bands, one containing the signal components below 5 kHz and the other band containing the components in the 5 kHz to 10 kHz change. Note that due to downsampling by 2, the total number of samples still remains the same. We can now use 16-bit accuracy for the lower subband with more energy and 8-bit accuracy for the higher band without losing any fidelity, as the high pass band has very little energy. Thus, for this case, we need only a total of 80 Kb/s + 40 Kb/s = 120 Kb/s. Thus, we have achieved, for this example, a compression ratio of 4/3. Of course, one need not be limited to only two subbands. Actually, for speech signals, which need 8 kHz sampling, 16 Kb/s is enough or, on the average 2 bit/sample.

A method for transmitting data through multi-carrier modulation

Abstract: A method for transmitting data by multi-carrier modulation, the data being at the transmitter end into blocks and modulated by inverse discrete Fourier transform in blocks, and wherein a plurality of carriers, that is, components of the DFT blocks are allocated that are not filled with data, and these carriers are occupied, so that a lot of time functions is generated which, after a filtering Dirac-like properties, characterized in that - that the time functions are generated so that the obtained after filtering Dirac-like functions have time at neighboring locations maxima in an oversampled temporal resolution, - that both the oversampled Dirac-like time functions (x ~ - that a pair selected from the pairs of the stored oversampled Dirac-like time functions and non-oversampled filter input functions, which has a maximum in the oversampled time functions at a predetermined point (m) of the over-sampling sampling grid, wherein in an iterative calculation in each step moved by applying the shift rate in each case a pair of oversampled time function and associated filter input function at the site of a caused by a data occupancy of the remaining carrier peak value and then weighted with a factor (α) is subtracted from the respective instantaneous time or input function, wherein said factor (α) is formed by a step size and by the crossing of a predetermined threshold (x be taken into account, in which in a preceding iterative step, the iterative modifications have been applied to the timing signal in the oversampled grid by filtering and the corresponding time signal without upsampling and filtering without parallel, and -

Performance analysis of a Fourier coefficient estimation method using IIR-BPFs and an LMS algorithm and improvement of its performance

Abstract: In many areas, it is extremely important to accurately estimate the amplitude and phase of a signal even if the frequency contained in the signal to be analyzed is known. For cases in which the signal frequencies to be analyzed are in a nonharmonic relation and white noise and interference are superposed on the signal, a method (BPLMS method) combining the IIR type BPF and the LMS algorithm has been proposed to derive the Fourier coefficients of the signal components accurately. In the present paper, in order to investigate a method of improving the BPLMS method, an approximate equation for the estimated accuracy is derived. Also, the validity of the approximate equation is confirmed by computer simulations under various conditions. As a result of the approximate analysis, it is shown that the normalized frequency difference between the signal and the interference can be increased by combining a downsampling process so that the estimation accuracy can be improved. Finally, by simulation, the proposed method and the BPLMS method are compared in terms of estimation accuracy. An example of application to the pitch estimation of musical sounds is given. © 2002 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 85(12): 19-28, 2002; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.10010