This paper proposes a low-voltage syllabic companding log domain filter without state variable correction circuits, which is needed for externally linear and time-invariant operation of conventional filters. The proposed filter is simplified and has wide input range under low-supply voltage by varying a nodal voltage adaptively. The simulation results show 60-dB input range for over 40-dB signal to noise plus distortion ratio at a power supply of 0.6 V in a 0.18-μm CMOS process.

This paper presents a low-voltage CMOS syllabic-companding log domain filter with wide dynamic range. The proposed filter is synthesized based on the idea of varying a voltage which is fixed on a dynamically adjustable biasing (DAB) filter. A low-voltage envelope detector necessary for a DAB filter is also designed including quantization mechanism. Simulation results show the proposed filter is effective in widening dynamic range

Band connections employed in active cancellation circuits for effective reduction of digital substrate noise are proposed. An almost-odd-symmetrical noise characteristic is utilized for canceling out noises with different signs outside of a guard ring. Advancing this idea, interlaced connections of four bands are also proposed. Excess cancellation by those bands is more effective for noise reduction in a guard ring than a cancellation by the two bands. Use of L-shaped bands on the basis of the interlaced connection suppresses the noise more. Simulation results show that the proposed band connections reduce the noise

This paper proposes a new sound source location algorithm for one receiving point noticing a formant frequency. We measure the sound source location using the phase difference of a formant frequency through three microphones and the height of the receiving point. The formant frequency is tracking by the zero-crossing detection of the formant signal and the echo is avoided by the detection of the occurrence of the formant signal. From some experiments, the proposed system can detect the sound source location with the errors less than 0.6 m under the echo environment.

We have presented a unique pitch detection method for transcription using twelve cascaded comb filters by the elimination of the pitch and its harmonic frequencies against the extraction of the pitch frequencies in many previous methods. But this method is not suitable in the case where the played tones have some frequency deviations from the nominal frequencies and involve some noise. To overcome this problem, we propose a new method for the pitch detection using adaptive comb filters by controlling the number of delay elements. Using three adaptive comb filters, we can detect the pitches of triple tones.

In this paper, motion vector (MV) estimation methods with high-speed intermediate pictures for low-power video compression are proposed. The intermediate pictures are obtained by a special type of CMOS image sensor. An adaptive iterative-search block matching is proposed to obtain precise MVs of video-rate pictures from high-speed intermediate pictures with the reduced computational complexity. The sensor captures high-speed intermediate pictures without destructing signal charge and video-rate pictures with full signal accumulation time. The proposed active pixel sensor using bidirectional multiple charge transfer is useful for the nondestructive intermediate imaging with a reduced fixed pattern noise. The image sensor chip has been implemented by using 0.35-/spl mu/m CMOS technology. It operates with 3.3 V and captures 480 frames/s high-speed nondestructive intermediate pictures and 30 frame/s fully accumulated video-rate pictures. The proposed adaptive iterative-search block matching has a comparable precision to a full search block matching with reduction of computational complexity by a factor of about 1/13, on average. It also reduces the data-loading rate from the memory by a factor of about 1/4.

We have previously proposed a portable navigation system for the visually impaired (see Nakamura, K. et al., IEICE Trans. of Japan, vol.J79, p.1610-18, 1996; Takuno, S. et al., IEICE Trans. of Japan, vol.J83, p.293-302, 2000). However, there are some places where this system is unable to provide proper guidance. Since the traveling direction of the traveler in this system is measured by a terrestrial magnetism sensor, there is a possibility that the wrong direction is measured, due to the distortion of the terrestrial magnetism by vehicles, etc. Road markings, e.g., pedestrian crossing, white line of the road, are reliable information to measure the correct direction. Furthermore, street landmarks, e.g., electric pole, traffic-control sign and post box, are also useful for a traveler to confirm his position. Therefore, we have now adequately applied image processing to our portable navigation system to make it more reliable.

This paper describes CMOS image sensors integrating video compression circuits. The on-sensor compression is particularly useful for the low-power design of moving picture compression hardware, which is demanded especially in the mobile computing and telephony. Recent progress of the CMOS image sensor technology allows us to realise the integration of high-performance image sensor and computational functions on a chip. An example of the CMOS image sensor with compression is described. Prospects of the CMOS image sensors with moving picture compression are discussed.

A CMOS smart image sensor LSI with video compression is presented. The proposed on-sensor compression scheme using an analog 2-D DCT processor is particularly useful to achieve low-power one-chip digital camera. A prototype image sensor LSI has been developed using 0.35 /spl mu/m double polysilicon, triple metal CMOS technology. Image coding using the implemented LSI has been demonstrated.

The authors derive a discrete Fourier transform (DFT) algorithm for unevenly sampled data by generalizing the notch Fourier transform (NFT) algorithm. First, a second-order digital notch filter for unevenly sampled data (2-VNF) that eliminates one frequency component is proposed, and the frequency responses of the 2-VNF are examined. Next, a finite impulse response (FIR) filter that can separate only one frequency component from an input signal is obtained by connecting 2-VNFs in cascade. The Fourier coefficients of the frequency component are derived from the output values of the FIR filter. Some computational examples showing the characteristics of the new DFT algorithm are presented.>

This paper presents a simple and effective algorithm for an adaptive multifrequency receiver that can be used for several types of multifrequency signals. This algorithm uses the ability of frequency detection of signal processing using extreme and zero-crossing points in the multifrequency signals and an adaptive second-order notch filter.

This paper describes a new detection algorithm for a dualtone multifrequency (DTMF) signal. The new algorithm basically uses only additions and subtractions but no multiplications. A signaling tone can be detected by sampling an input signal at four times each frequency involved in a signaling tone and accumulating these sampled values synchronously by additions and subtractions. This algorithm is referred to as the synchronous additions and subtractions (SAS) method. Many conventional methods have used various types of filters to detect signaling tones. In the SAS method in which the design of filters is unnecessary, only the generation of sampling pulses is required, and main operations are additions and subtractions. These features are useful to implement flexible receivers and various types of receivers. The simplicity of the SAS method is demonstrated by the implementation of a DTMF receiver using a conventional 8-bit microcomputer without any special hardware. The minimum signal-to-noise ratio is 5.7 dB when the receiver operates for the signaling tones to be accepted. Four easily measurable parameters are used to distinguish signaling tones from noise inputs. Only two false detections occurred for severe noise inputs of thirty hours.

This paper describes another computational algorithm for the discrete Fourier transform(DFT) via the discrete Walsh transform(DWT). The number of multiplications required by this algorithm is approximately NL/9 where N is the number of data points and L is the number of Fourier coefficients desired. This number shows a 33 % decrease against NL/6 in the previous algorithm published by us. The proposed algorithm can be derived by using conventional sampling points in the DFT. The DFT computation via the DWT is superior to the fast Fourier transform(FFT) approach in applications where L is relatively small compared with N and where the Walsh and Fourier coefficients are both desired.