The characteristics of the signalprocessing tasks associated with an advanced wireless receiver are well matched to the capabilities offered by CCM (custom computing machine) technology. Collectively, digital receive...
详细信息
The characteristics of the signalprocessing tasks associated with an advanced wireless receiver are well matched to the capabilities offered by CCM (custom computing machine) technology. Collectively, digital receiver algorithms seem to share the following properties: (a) repetitive operations are performed on huge data sets, (b) the dominant computations are conducive to very deep computational pipelines, (c) a moderate amount of latency can be tolerated, and (d) different environmental conditions require different signalprocessing methods, which in turn require distinct computational structures (time-varying computation). In addition, when coupled with run-time reconfiguration, the amalgamation of signalprocessing tasks may be compactly combined in a power-efficient computing module. This paper focuses on the design methodology for implementing large and intricate stream-oriented signalprocessing tasks.
Speech recognition is a major topic in speech signalprocessing. Many algorithms based on results of speech analysis, among which dynamic time warping) and hidden Markov models are the most important, have been advanc...
详细信息
Speech recognition is a major topic in speech signalprocessing. Many algorithms based on results of speech analysis, among which dynamic time warping) and hidden Markov models are the most important, have been advanced. However, these algorithms generally turn out to be too complicated to be implemented in real time systems. The proposed algorithm in this paper, which uses a fuzzy logic recognition approach based on the power distribution pattern of a segment of a speech, allows the implementation of real-time speech recognition.
An elective course in electronic music synthesis has been developed for electrical and computer engineering students. The course provides an interesting way to integrate and apply DSP and computer manipulation concept...
详细信息
An elective course in electronic music synthesis has been developed for electrical and computer engineering students. The course provides an interesting way to integrate and apply DSP and computer manipulation concepts studied in previous courses, and extends student understanding of more advanced concepts such as time-evolving spectra. MATLAB is a standard platform used in the signals and DSP courses, so MATLAB forms the primary tool for converting algorithmic descriptions of waveforms into sound. The paper outlines course topics and methods, includes a detailed example of pedagogy, and presents assessment results.
Over the last decade, digital signalprocessing (DSP) has evolved from being a term known by only a few specialists, to a household term. The growth in DSP applied in e.g., consumer, medical, communications, networkin...
详细信息
Over the last decade, digital signalprocessing (DSP) has evolved from being a term known by only a few specialists, to a household term. The growth in DSP applied in e.g., consumer, medical, communications, networking and computing devices has been spectacular. In fact, the digital signal processor market has grown 40% per year since 1988 and this figure is expected to continue over the neat 10 years. At the same time, the extreme improvement in hardware technologies has been paving the way for designing dedicated architectures for real time execution of still more complex DSP algorithms, continuously decreasing the power and silicon consumption required to perform certain functionalities. Consequently, the authors consider advanced DSP topics as being essential in the curriculum for a still growing number of electrical engineering students. These topics are basically related to: (1) the design of highly complex DSP algorithms according to given specifications; and (2) real time implementation of these algorithms using various and conceptually different hardware architectures. Aalborg University, Denmark, which has a longstanding tradition for project-oriented teaching in traditional DSP theory at the Master level, therefore in 1994 launched a new Master programme in "DSP algorithms and ASIC Architectures" Now, after five years of very successful execution of this programme, the authors report on their experiences.
We present an efficient and numerically stable implementation of a class of adaptive beamforming algorithms. Our method is based on a variant of the URV factorization, and is particularly useful when the effective ran...
详细信息
We present an efficient and numerically stable implementation of a class of adaptive beamforming algorithms. Our method is based on a variant of the URV factorization, and is particularly useful when the effective rank of the data matrix is smaller than the number of antenna elements, which is usually the case when the number of signal sources is smaller than the number of antenna elements in a high SNR environment.
This paper presents the implementation of an MPEG-2 advanced audio coding (AAC) coder on a fixed-point DSP. AAC is a powerful coding algorithm and offers high-quality multi-channel surround audio. We also discuss the ...
详细信息
This paper presents the implementation of an MPEG-2 advanced audio coding (AAC) coder on a fixed-point DSP. AAC is a powerful coding algorithm and offers high-quality multi-channel surround audio. We also discuss the implementation issue and effort in porting AAC algorithms to the DSP.
Image indexing is the process of image retrieval from databases of images or videos based on their contents. Specifically histogram-based algorithms are considered to be effective for color image indexing. We suggest ...
详细信息
Image indexing is the process of image retrieval from databases of images or videos based on their contents. Specifically histogram-based algorithms are considered to be effective for color image indexing. We suggest a new method of color space quantization in the CIELUV color space, named weighted LUV quantization. With this method, each bin in the LUV space has a different weighting factor, which is applied to the histogram intersection. The weighted LUV histogram intersection provides the advantage of perceptual uniformity of the CIELUV color space. An additional advantage is the consideration of perceptual sensitivity to more saturated colors by the use of a weighting factor.
Third generation CDMA systems will simultaneously support traffic with different data rates, such as data and voice traffic. advanced space-time processing can be used to achieve high system capacity in such systems. ...
详细信息
Third generation CDMA systems will simultaneously support traffic with different data rates, such as data and voice traffic. advanced space-time processing can be used to achieve high system capacity in such systems. In this paper, we extend the previously developed multiple antenna successive interference canceller of Mailaender and Huang (see UCSD conference on Wireless Communications, p.47-53, San Diego, CA, 1999) to the case of mixed data rate operation. The gain in this case relative to a simple space-time RAKE receiver is even greater than in the single rate case.
This paper reports a new simulation method using software based millimeter-wave test bed that has been developed to emulate a wide range of millimeter-wave communication concepts in the hardware-in-the-loop-simulation...
详细信息
This paper reports a new simulation method using software based millimeter-wave test bed that has been developed to emulate a wide range of millimeter-wave communication concepts in the hardware-in-the-loop-simulations and to evaluate an advancedsignalprocessing algorithm for the high speed communication. The system description of the test bed architecture and the implementation of a millimeter-wave transceiver of 38 and 60 GHz are presented. To evaluate the test bed performance, the BERs and the constellations are measured.
This paper describes a smart recognition system which performs character matching by replacing speech parameters with the five Japanese vowels and a few consonant categories. The proposed algorithm can make speaker-in...
详细信息
This paper describes a smart recognition system which performs character matching by replacing speech parameters with the five Japanese vowels and a few consonant categories. The proposed algorithm can make speaker-independent voice recognition. The algorithm has an advantage over the conventional speaker-independent word recognition system because it can reduce the required memory to about 0.5% of the conventional algorithm for storing the reference templates and for the instruction set, and can be performed even in a low-speed processor. We implemented this recognition algorithm in a fixed-point, 20 MIPS digital signal processor board with a 9-k/spl times/16-bit on-chip RAM. Recognition experiments using 20 Japanese city names had a 90.3% accuracy. Such an accuracy is good enough for a voice control system.
暂无评论