Adaptive design of delta sigma modulators

In this thesis, a genetic algorithm based on differential evolution (DE) is used to generate delta sigma modulator (DSM) noise transfer functions (NTFs). These NTFs outperform those generated by an iterative approach described by Schreier and implemented in the delsig Matlab toolbox. Several lowpass and bandpass DSMs, as well as DSM\u27s designed specifically for and very low intermediate frequency (VLIF) receivers are designed using the algorithm developed in this thesis and compared to designs made using the delsig toolbox. The NTFs designed using the DE algorithm always have a higher dynamic range and signal to noise ratio than those designed using the delsig toolbox

Cascaded feedforward sigma-delta modulator for wide bandwidth applications

[[abstract]]A new sigma-delta modulator architecture for wide bandwidth application called cascaded feedforward sigma-delta modulator is proposed in this paper. This sigma-delta modulator is similar to the conventional feedforward summation sigma-delta modulator. The conventional feedforward summation sigma-delta modulator uses multi-bit feedback and therefore a multi-bit digital-to-analog converter (DAC) is needed. Due to the nonlinearity of the multi-bit DAC, it is difficult to be implemented. On the other hand the proposed approach uses 1.5-bit feedback, and thus the implementation of the analog part is much easier than the conventional one. Since the 1.5-bit feedback will cause coarse quantization errors, error cancellation must be done in the digital part. Here an adaptive filter with least mean square algorithm is used to reduce the nonlinear effect. The simulation results show that the signal to noise plus distortion ratio (SNDR) of this architecture is very close to that of the ideal feedforward summation sigma-delta modulator with multi-bit DAC and can be used for the wide bandwidth application.[[notice]]補正完

Single-bit adaptive channel equalization for narrowband signals

In this paper, a new design of a single-bit adaptive channel equalization is proposed using sigma delta modulation and a single-bit block Least Mean Square (LMS) algorithm. With correlated narrowband input signals, this model is capable to converge and provide equivalent equalization filter with improvement in the SNR and very low Symbol Error Rate (SER). The input, filter coefficients and output values are all in single-bit and ternary format that results in a reduction in hardware complexity compared to traditional multi-bit channel equalization. Additionally, the technique avoids the need for successive conversion from multi-bit to single bit and back at the receiver and transmitter stages

Design, analysis and evaluation of sigma-delta based beamformers for medical ultrasound imaging applications

The inherent analogue nature of medical ultrasound signals in conjunction with the abundant merits provided by digital image acquisition, together with the increasing use of relatively simple front-end circuitries, have created considerable demand for single-bit beamformers in digital ultrasound imaging systems. Furthermore, the increasing need to design lightweight ultrasound systems with low power consumption and low noise, provide ample justification for development and innovation in the use of single-bit beamformers in ultrasound imaging systems. The overall aim of this research program is to investigate, establish, develop and confirm through a combination of theoretical analysis and detailed simulations, that utilize raw phantom data sets, suitable techniques for the design of simple-to-implement hardware efficient digital ultrasound beamformers to address the requirements for 3D scanners with large channel counts, as well as portable and lightweight ultrasound scanners for point-of-care applications and intravascular imaging systems. In addition, the stability boundaries of higher-order High-Pass (HP) and Band-Pass (BP) Σ−Δ modulators for single- and dual- sinusoidal inputs are determined using quasi-linear modeling together with the describing-function method, to more accurately model the modulator quantizer. The theoretical results are shown to be in good agreement with the simulation results for a variety of input amplitudes, bandwidths, and modulator orders. The proposed mathematical models of the quantizer will immensely help speed up the design of higher order HP and BP Σ−Δ modulators to be applicable for digital ultrasound beamformers. Finally, a user friendly design and performance evaluation tool for LP, BP and HP modulators is developed. This toolbox, which uses various design methodologies and covers an assortment of modulators topologies, is intended to accelerate the design process and evaluation of modulators. This design tool is further developed to enable the design, analysis and evaluation of beamformer structures including the noise analyses of the final B-scan images. Thus, this tool will allow researchers and practitioners to design and verify different reconstruction filters and analyze the results directly on the B-scan ultrasound images thereby saving considerable time and effort

Reconfigurable switched-current ΣΔ power amplifier

Trabajo presentado al XIV Iberchip celebrado en Puebla (México) del 20 al 22 de febreo de 2008.In this paper a reconfigurable switched-current sigma delta modulator for power amplifier is shown, which is a suitable topology to save energy and integrated area in portable applications. Results have shown the correct operation of a 1.8V , 0.18¿ m CMOS reconfigurable switched-current sigma delta modulator with 11bits dynamic range within 1MHz and 7.8bits within 3.8MHz bandwidth.One of the authors Rosalino Rodríguez Calderon thanks the economic support received from CONACYT (México).Peer Reviewe

Error Correction For Automotive Telematics Systems

One benefit of data communication over the voice channel of the cellular network is to reliably transmit real-time high priority data in case of life critical situations. An important implementation of this use-case is the pan-European eCall automotive standard, which has already been deployed since 2018. This is the first international standard for mobile emergency call that was adopted by multiple regions in Europe and the world. Other countries in the world are currently working on deploying a similar emergency communication system, such as in Russia and China. Moreover, many experiments and road tests are conducted yearly to validate and improve the requirements of the system. The results have proven that the requirements are unachievable thus far, with a success rate of emergency data delivery of only 70%. The eCall in-band modem transmits emergency information from the in-vehicle system (IVS) over the voice channel of the circuit switch real time communication system to the public safety answering point (PSAP) in case of a collision. The voice channel is characterized by the non-linear vocoder which is designed to compress speech waveforms. In addition, multipath fading, caused by the surrounding buildings and hills, results in severe signal distortion and causes delays in the transmission of the emergency information. Therefore, to reliably transmit data over the voice channels, the in-band modem modulates the data into speech-like (SL) waveforms, and employs a powerful forward error correcting (FEC) code to secure the real-time transmission. In this dissertation, the Turbo coded performance of the eCall in-band modem is first evaluated through the adaptive white Gaussian noise (AWGN) channel and the adaptive multi-rate (AMR) voice channel. The modulation used is biorthogonal pulse position modulation (BPPM). Simulations are conducted for both the fast and robust eCall modem. The results show that the distortion added by the vocoder is significantly large and degrades the system performance. In addition, the robust modem performs better than the fast modem. For instance, to achieve a bit error rate (BER) of 10^{-6} using the AMR compression rate of 7.4 kbps, the signal-to-noise ratio (SNR) required is 5.5 dB for the robust modem while a SNR of 7.5 dB is required for the fast modem. On the other hand, the fading effect is studied in the eCall channel. It was shown that the fading distribution does not follow a Rayleigh distribution. The performance of the in-band modem is evaluated through the AWGN, AMR and fading channel. The results are compared with a Rayleigh fading channel. The analysis shows that strong fading still exists in the voice channel after power control. The results explain the large delays and failure of the emergency data transmission to the PSAP. Thus, the eCall standard needs to re-evaluate their requirements in order to consider the impact of fading on the transmission of the modulated signals. The results can be directly applied to design real-time emergency communication systems, including modulation and coding

A CMOS Digital Beamforming Receiver

As the demand for high speed communication is increasing, emerging wireless techniques seek to utilize unoccupied frequency ranges, such as the mm-wave range. Due to high path loss for higher carrier frequencies, beamforming is an essential technology for mm-wave communication. Compared to analog beamforming, digital beamforming provides multiple simultaneous beams without an SNR penalty, is more accurate, enables faster steering, and provides full access to each element. Despite these advantages, digital beamforming has been limited by high power consumption, large die area, and the need for large numbers of analog-to-digital converters. Furthermore, beam squinting errors and ADC non-linearity limit the use of large digital beamforming arrays. We address these limitations. First, we address the power and area challenge by combining Interleaved Bit Stream Processing (IL-BSP) with power and area efficient Continuous-Time Band-Pass Delta-Sigma Modulators (CTBPDSMs). Compared to conventional DSP, IL-BSP reduces both power and area by 80%. Furthermore, the new CTBPDSM architecture reduces ADC area by 67% and the energy per conversion by 43% compared to previous work. Second, we introduce the first integrated digital true-time-delay digital beamforming receiver to resolve the beam squinting. True-time-delay beamforming eliminates squinting, making it an ideal choice for large-array wide-bandwidth applications. Third, we present a new current-steering DAC architecture that provides a constant output impedance to improve ADC linearity. This significantly reduces distortion, leading to an SFDR improvement of 13.7 dB from the array. Finally, we provide analysis to show that the ADC power consumption of a digital beamformer is comparable to that of the ADC power for an analog beamformer. To summarize, we present a prototype phased array and a prototype timed array, both with 16 elements, 4 independent beams, a 1 GHz center frequency, and a 100 MHz bandwidth. Both the phased array and timed array achieve nearly ideal conventional and adaptive beam patterns, including beam tapering and adaptive nulling. With an 11.2 dB array gain, the phased array achieves a 58.5 dB SNDR over a 100 MHz bandwidth, while consuming 312 mW and occupying 0.22 mm2. The timed array achieves an EVM better than -37 dB for 5 MBd QAM-256 and QAM-512, occupies only 0.29 mm2, and consumes 453 mW.PHDElectrical EngineeringUniversity of Michigan, Horace H. Rackham School of Graduate Studieshttps://deepblue.lib.umich.edu/bitstream/2027.42/147716/1/smjang_1.pd