Chivapreecha, Sorawat
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Chivapreecha, Sorawat
Alternative Name
Chivapreecha, S.
Chivapreecha, Sorwat
Main Affiliation
Email
sorawat.ch@kmitl.ac.th
16 results
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Item type:Publication, Continuously Tunable Frequency and Phase Biquad Oscillator(2026-05-01); Suwannawach, PiyapanThis paper presents an improvement to the performance of a biquad oscillator, which is a recursive oscillator known for its excellent long-term stability. However, a significant limitation is that the oscillation frequency cannot be changed while the system is operating. Directly changing the frequency during operation causes the amplitude of the generated signal to vary significantly, either increasing or decreasing. A zero-input response analysis is used to understand the cause of this problem and to develop a solution that allows the amplitude of the generated signal to remain constant even when the oscillation frequency is changed during operation. In addition, this paper presents a method for controlling the phase of the generated signal by using a 1st-order IIR phase shifter structure. The proposed structure is specifically designed to allow the phase of the signal at the oscillation frequency to be adjusted continuously and independently of the frequency parameter. This integrated oscillator enables real-time, independent control of both signal frequency and phase without amplitude drift, making it suitable for applications requiring precise and dynamic signal synthesis. - Some of the metrics are blocked by yourconsent settings
Item type:Publication, Deep Neural Networks for Sound Synthesis of Thai Duct F1ute, Khlui(2021-04-01) ;Sinjankhom, Tantep; ; Kato, TomonoriThis paper introduces Thai musical instrument sound synthesis with an intelligent technique which is a combination of deep neural networks and conventional signal processing techniques. The instrument focused on in this paper is the Khlui, which is a unique Thai reedless wind instrument. Khlui sound can be synthesized by performing a combination of additive and subtractive syntheses. The synthesis system takes the pitch and loudness of any monophonic input signal. Then, multilayer perceptron and recurrent neural networks are used together in order to generate 3 parameters which are harmonic distribution, filter magnitude response, and the signal envelope. Finally, the harmonic characteristic of the Khlui along with the blowing noises are highly synthesized. The results are natural-sounding and realistic when compared with the recorded Khlui audio. This research will contribute to the more convenient natural synthesis of the Khlui sounds. - Some of the metrics are blocked by yourconsent settings
Item type:Publication, Design and Implementation of Multiphase Digital Oscillator(2024-01-01) ;Jongsataporn, Thitaphan ;Yospanya, PoonnaThis paper presents a design and implementation of the multiphase digital oscillator, which can determine the initial phase of multi-output signals to desired initial phases. The proposed system consists of biquad digital oscillator to oscillate the reference signal, and the new phase shifters which are first order IIR filters that can shift the phase of signals at a specified frequency to give them the desired phases. Design example and simulation results will be shown. Moreover, real-time hardware implementation on STM32 digital signal processor can be demonstrated to confirm the proposed design can work with hardware experimental results. - Some of the metrics are blocked by yourconsent settings
Item type:Publication, Embedded Hardware Implementation of Noise Gate on STM32 Processor Using MATLAB/Simulink(2025-01-01) ;Jongsataporn, ThitaphanThis paper presents a model-based approach for developing a digital signal processing (DSP) system, specifically focusing on a Noise Gate application, using MATLAB/Simulink, and deploying it on the STM32F769I-Discovery board. The study thoroughly examines the advantages of model-based hardware implementation, particularly in facilitating rapid prototyping for real-time applications in educational contexts. The STM32F769I is powered by a high-performance Arm® Cortex®-M7 core, which can execute DSP instructions efficiently. Its ample memory and built-in debugging tools enhance its suitability for complex signal processing tasks. The implementation leverages MATLAB/Simulink and its robust compatibility with the STM32 hardware, allowing for seamless software and hardware component integration. The powerful model-based development features streamline the transition from simulation to physical hardware, reducing development time and enhancing reliability. The Noise Gate application is designed to control audio signals effectively, eliminating unwanted background noise through user-defined parameters such as threshold, attack time, and release time, which optimize performance based on specific audio environments. Additionally, the design applies a first-order IIR low-pass as a smoothing filter, which dynamically adjusts the gain according to the characteristics of the incoming audio signal. This ensures gradual opening and closing of the gate, resulting in smoother audio transitions. The results demonstrate significant noise suppression and overall audio quality enhancement, highlighting the practicality of employing model-based design methodologies for DSP applications. Furthermore, the insights gained from this study contribute to efficient hardware prototype development and offer implications for future projects in the field of digital signal processing. - Some of the metrics are blocked by yourconsent settings
Item type:Publication, Real-Time Zero-Phase Digital Filter Using Recurrent Neural Network(2023-01-01) ;Sinjanakhom, TantepThis paper proposes a method to design and implement a zero-phase digital filter that can run in a real-time system. Generally, zero-phase filters are designed for non-causal systems only as the time-reversal operations are required. Thus, the typical usage of these filters is for offline applications. For this reason, we propose a real-time zero-phase digital filter that is designed based on a recurrent neural network model, particularly the gated recurrent units. The model learns to perform zero-phase filtering by using training data made from the filtered signals that are generated by using the conventionally designed zero-phase filter. The original digital filter used to create the dataset is an IIR filter performing forward-backward filtering. The best trained model yields the mean absolute loss values at approximately 0.001 and can process at least 30 times faster than real-time. Furthermore, the trained model was implemented as a 3-band zero-phase graphic equalizer to exhibit one of its applications. - Some of the metrics are blocked by yourconsent settings
Item type:Publication, Linear Phase FIR Filter Design for Digital Hearing Aids Using a Neural Network-Based Optimization(2024-01-01); Yospanya, PoonnaA neural network-based optimization for the design of linear phase digital filters used in digital hearing aid applications is presented in this paper. To achieve hearing loss compensation, a 53-tap finite impulse response (FIR) filter is utilized, and the weights of the trained network can be used to obtain the impulse response of the FIR filter. The target frequency response or label comes from the audiogram, the training data is created, and a linear perceptron supervised learning model is used. Audiogram matchings are shown in design examples. The proposed neural network-based design approach can give a very high accuracy, high processing speed when compared with the existing methods, and also low complexity in filter structure realization. - Some of the metrics are blocked by yourconsent settings
Item type:Publication, Evaluation of Multi-Bit Input Logic Blocks in RTL-Designed FPGA Architecture: A Framework for FPGA and ASIC Integration(2025-01-01) ;Sato, Tomoaki ;Murakami, Anyu; This paper explores the evaluation and optimization of multi-bit input logic blocks (LBs) within RTL-designed FPGA architectures. Traditional FPGA designs face limitations in power consumption, delay, and area due to the constraints of reconfigurable circuits. The proposed architecture leverages RTL-level design capabilities to address these challenges and enables the co-design of FPGAs and ASICs. The authors evaluate the performance of 8-bit, 16-bit, and 32-bit input LBs in terms of delay, area, and synthesis feasibility. The results demonstrate that 8-bit input LBs achieve a delay of 0.68 ns with an area of 2202.48 μm<sup>2</sup>, outperforming multi-stage smaller LBs. Although 16-bit input LBs show potential for delay reduction, their synthesis demands significant time and results in a large area footprint, rendering them impractical. Synthesis of 32-bit input LBs was not feasible due to current tool limitations. These findings highlight the effectiveness of 8-bit input LBs for pattern matching tasks and emphasize the importance of application-specific optimization. The fixed routing feature of RTL-designed FPGAs facilitates the development of efficient, customizable designs tailored to specific workloads. This work contributes to the advancement of FPGA architectures, offering insights for future research on larger input LBs and their integration into high-performance applications. - Some of the metrics are blocked by yourconsent settings
Item type:Publication, Application-Optimized FPGAs Design Using RTL-Designed FPGAs Architectures(2025-01-01) ;Sato, Tomoaki ;Murakami, Anyu; RTL-Designed Field-Programmable Gate Arrays (FPGAs) can describe FPGA functionality using Hardware Description Languages (HDLs), which means they can be easily customized to configure the FPGA. In conventional FPGAs, switches are used for routing control, making it impossible to design them using HDLs. This study leverages the customizable nature of RTL-Designed FPGAs to explore the optimal configuration of FPGAs for packet processing in computer networks. It demonstrates that a 4-input Look-Up Table (LUT) is superior to a 3-input LUT in terms of throughput and reveals that, as the number of LUT inputs increases, having 5 routing paths is more optimal than 4. - Some of the metrics are blocked by yourconsent settings
Item type:Publication, A Deep Learning Approach to Digital Filter Parameter Estimation Based on Amplitude Responses(2021-01-21) ;Yospanya, Poonna; Jongsataporn, ThitaphanThis paper presents our attempt to tackle the problem of digital filter type and parameter estimation given a set of points sampled from a filter frequency response. We compared results from various multilayer perceptron and convolutional neural network configurations. The results suggest that a convolutional neural network generally produces faster convergence with a lower loss at the same number of epochs than a multilayer perceptron network. However, the maximum amplitude response error, which is the true performance metrics, can be comparable in some cases. A combination of multiple best-performing configurations for different tasks is used to assemble the final model. - Some of the metrics are blocked by yourconsent settings
Item type:Publication, Neural Networks for Real-Time Digital Emulation of Guitar Speaker Cabinet Impulse Response(2022-01-01) ;Sinjanakhom, TantepThis This paper presents a real-time signal processing system in which a neural network generates the impulse response (IR) of a Marshall 1960A guitar cabinet with 25W Celestion speakers based on user-specified parameters. The parameters include the microphone type, position of the speaker on which the microphone is mounted, distance between the microphone and the cabinet, and off-axis tilting angle. The trained model of neural network can generate the impulse response for a speaker cabinet, as well as the sound of settings not included in training set. Cross-correlation, error-to-signal ratio, power spectral density error, and magnitude-squared coherence were all utilized to assess the model's output. Mean Opinion Score listening tests were performed to determine the similarity of the convolved guitar signals. According to the results, the emulated cabinet sounds were perceived to be nearly identical to the original sounds. The performance of the real-time audio plugin implementation is proved to be computationally efficient. Because raw IR data for each microphone configuration does not need to be saved directly to the PC's memory, utilizing it in music production work can be more convenient, allowing the user to modify the parameters while hearing the differences without having to repeat the IR file loading procedure.
