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Description
Time-interleaved analog to digital converters (ADCs) have become critical components in high-speed communication systems. Consumers demands for smaller size, more bandwidth and more features from their communication systems have driven the market to use modern complementary metal-oxide-semiconductor (CMOS) technologies with shorter channel-length transistors and hence a more compact design. Downscaling the supply voltage which is required in submicron technologies benefits digital circuits in terms of power and area. Designing accurate analog circuits, however becomes more challenging due to the less headroom. One way to overcome this problem is to use calibration to compensate for the loss of accuracy in analog circuits.
Time-interleaving increases the effective data conversion rate in ADCs while keeping the circuit requirements the same. However, this technique needs special considerations as other design issues associated with using parallel identical channels emerge. The first and the most important is the practical issue of timing mismatch between channels, also called sample-time error, which can directly affect the performance of the ADC. Many techniques have been developed to tackle this issue both in analog and digital domains. Most of these techniques have high complexities especially when the number of channels exceeds 2 and some of them are only valid when input signal is a single tone sinusoidal which limits the application.
This dissertation proposes a sample-time error calibration technique which bests the previous techniques in terms of simplicity, and also could be used with arbitrary input signals. A 12-bit 650 MSPS pipeline ADC with 1.5 GHz analog bandwidth for digital beam forming systems is designed in IBM 8HP BiCMOS 130 nm technology. A front-end sample-and-hold amplifier (SHA) was also designed to compare with an SHA-less design in terms of performance, power and area. Simulation results show that the proposed technique is able to improve the SNDR by 20 dB for a mismatch of 50% of the sampling period and up to 29 dB at 37% of the Nyquist frequency. The designed ADC consumes 122 mW in each channel and the clock generation circuit consumes 142 mW. The ADC achieves 68.4 dB SNDR for an input of 61 MHz.
Time-interleaving increases the effective data conversion rate in ADCs while keeping the circuit requirements the same. However, this technique needs special considerations as other design issues associated with using parallel identical channels emerge. The first and the most important is the practical issue of timing mismatch between channels, also called sample-time error, which can directly affect the performance of the ADC. Many techniques have been developed to tackle this issue both in analog and digital domains. Most of these techniques have high complexities especially when the number of channels exceeds 2 and some of them are only valid when input signal is a single tone sinusoidal which limits the application.
This dissertation proposes a sample-time error calibration technique which bests the previous techniques in terms of simplicity, and also could be used with arbitrary input signals. A 12-bit 650 MSPS pipeline ADC with 1.5 GHz analog bandwidth for digital beam forming systems is designed in IBM 8HP BiCMOS 130 nm technology. A front-end sample-and-hold amplifier (SHA) was also designed to compare with an SHA-less design in terms of performance, power and area. Simulation results show that the proposed technique is able to improve the SNDR by 20 dB for a mismatch of 50% of the sampling period and up to 29 dB at 37% of the Nyquist frequency. The designed ADC consumes 122 mW in each channel and the clock generation circuit consumes 142 mW. The ADC achieves 68.4 dB SNDR for an input of 61 MHz.
ContributorsNazari, Ali (Author) / Barnaby, Hugh James (Thesis advisor) / Jalali-Farahani, Bahar (Committee member) / Bakkaloglu, Bertan (Committee member) / Kitchen, Jennifer (Committee member) / Arizona State University (Publisher)
Created2017
Description
A dual-channel directional digital hearing aid (DHA) front end using Micro Electro Mechanical System (MEMS) microphones and an adaptive-power analog processing signal chain is presented. The analog front end consists of a double differential amplifier (DDA) based capacitance to voltage conversion circuit, 40dB variable gain amplifier (VGA) and a continuous time sigma delta analog to digital converter (CT - ΣΔ ADC). Adaptive power scaling of the 4th order CT - ΣΔ achieves 68dB SNR at 120μW, which can be scaled down to 61dB SNR at 67μW. This power saving will increse the battery life of the DHA.
ContributorsDeligoz, Ilker (Author) / Kiaei, Sayfe (Thesis advisor) / Bakkaloglu, Bertan (Committee member) / Jalali-Farahani, Bahar (Committee member) / Aberle, James T., 1961- (Committee member) / Chae, Junseok (Committee member) / Arizona State University (Publisher)
Created2010