Pulse Code Research Articles

Energy-based linear PCM audio recovery method of impaired MP4 file stored in dashboard camera memory

Sounds recorded by a dashboard camera presents essential evidence on a vehicle's driving condition for analyzing car accidents. In reality, when a direct impact of a vehicle equipped with a dashboard camera occurs, the power supply to the dashboard camera is abruptly terminated, and the digital video being recorded at that time may be incompletely stored; thus, there is no information such as the location or size to be read for the image and audio decoding. In this paper, we propose a method for recovering the linear pulse code modulation (PCM) audio from the impaired MP4 file partially stored in a video event data recorder (VEDR), known as a dashboard camera, during road vehicle accidents. This method is based on the principle of extracting an actual audio signal using the energy of each frame in the pseudo audio, which is generated from all data in the “mdat” area of the impaired MP4 file. For performance evaluation, the various video files with linear PCM audio format in the MP4 container were obtained through the different dashboard cameras. As the objective evaluation, we measured the audio recovery ratio, distance based on dynamic time warping (DTW) and comparison of the waveform and spectrogram. Moreover, the performance of the proposed method was evaluated using the multiple stimuli with a hidden reference and anchor (MUSHRA) test as the subjective assessment. As a result, it is shown from the evaluation that the proposed method can recover significantly better the linear PCM audio signal than the conventional methods from an impaired MP4 file.

Sequentially Adaptive Differential Pulse Code Modulation Using Adaptive LSP Filters.(Dept.E)

Sequentially Adaptive Differential Pulse Code Modulation Using Adaptive LSP Filters.(Dept.E)

A Comparative Analysis of Image Compression using PCM and DPCM

The quality of the signal is essential in digital communication and signal processing. The transmission channel is also important. Modulation is used for effectively transmission of signal. There exist several types of modulation techniques. One of such is the pulse code modulation (PCM). The performance of PCM is however affected by quantization error and noise in the transmission channel, which affects the quality of the output. Against this backdrop, this paper presents the use of differential pulse code modulation (DPCM) so as to address the limitation of pulse code modulation. The simulation environment is MATLAB 2018a. The MATLAB Simulink is used to design the PCM and DPCM systems using appropriate digital processing blocks. The DPCM system shows a significant improvement in terms of error reduction and quality of output.

Differential pulse code modulation and motion aligned optimal reconstruction for block-based compressive video sensing using conditional autoregressive-salp swarm algorithm

Compressive Video Sensing (CVS) is the video coding framework that uses compressive sensing (CS) theory into video codec to reduce the burden of encoding. Various existing methods are developed for CVS, but enhancing reconstruction speed and reconstruction quality remains a great challenge in video compression. Hence, an optimal method is developed using the proposed Differential Pulse Code Modulation+Conditional Autoregressive-Salp Swarm (DPCM+CA-SS) model for CVS system by considering the characteristics of CVS frame. The spatial redundancy of video frames is explored by partitioning the video frames into several blocks such that each block is measured using the measurement matrix. The measurement vectors are quantized using DPCM. The key frames are compressed into bits using Huffman coding, and the packet is transmitted to the decoder side. The temporal redundancy of CVS model is explored by encoding the frames using CA-SS-based prediction model to achieve Motion Aligned (MA) optimal reconstruction. However, the proposed CA-SS is designed by the integration of Conditional Autoregressive Value at Risk (CAViaR) and Salp Swarm Algorithm (SSA), respectively. However, the proposed DPCM+CA-SS obtained maximum PSNR and SSIM of 39.6649 db and 0.9120, and the minimum total time of 4.9523 s, respectively.

Stream-Based Visually Lossless Data Compression Applying Variable Bit-Length ADPCM Encoding.

Video applications have become one of the major services in the engineering field, which are implemented by server–client systems connected via the Internet, broadcasting services for mobile devices such as smartphones and surveillance cameras for security. Recently, the majority of video encoding mechanisms to reduce the data rate are mainly lossy compression methods such as the MPEG format. However, when we consider special needs for high-speed communication such as display applications and object detection ones with high accuracy from the video stream, we need to address the encoding mechanism without any loss of pixel information, called visually lossless compression. This paper focuses on the Adaptive Differential Pulse Code Modulation (ADPCM) that encodes a data stream into a constant bit length per data element. However, the conventional ADPCM does not have any mechanism to control dynamically the encoding bit length. We propose a novel ADPCM that provides a mechanism with a variable bit-length control, called ADPCM-VBL, for the encoding/decoding mechanism. Furthermore, since we expect that the encoded data from ADPCM maintains low entropy, we expect to reduce the amount of data by applying a lossless data compression. Applying ADPCM-VBL and a lossless data compression, this paper proposes a video transfer system that controls throughput autonomously in the communication data path. Through evaluations focusing on the aspects of the encoding performance and the image quality, we confirm that the proposed mechanisms effectively work on the applications that needs visually lossless compression by encoding video stream in low latency.

Processing of Digital Signals in (+1, -1) Binary System

Digital Signal Processing in (+1, -1) system has been studied in detail. Effects of number representation on quantization and truncation of rounding have been discussed. First order and second order filters have been investigated. Differential pulse code modulation method of coding is being studied. It is found that this system is well suited for VLSI design. The hardware realization of different components of DPCM is easy and straight. The position and negative signals can be processed in a unified way. It gives better performance in comparison of two’s complement representation of conventional binary system. The effect of limit cycles, stability and round off noise in first and second order filter have been discussed too.

Кодовое управление параметрическими моделями индукционных нагревателей

In this paper, we describe our approach to building controllable models of induction heaters. Parameters of equivalent circuits of aluminum heating inductors include certain components tied to secondary elements. As long as metal properties are subject to change during the heating process, resistive and reactive parts of equivalent circuits should be taken into account. Adjustable resistive components have rarely been used in circuit models. To control equivalent resistive two-terminal parts, we offer pulse code control of impedance-changing switches. Controlling switching time in models with series, parallel, and mixed connection provides required softness of resistive conductivity change. Control characteristics of current and voltage were obtained with use of a simulator. As the result of our numerical experiment, the set of operating parameters for an induction device was determined.

Music Rhythm Detection Algorithm Based on Multipath Search and Cluster Analysis

Music rhythm detection and tracking is an important part of the music comprehension system and visualization system. The music signal is subjected to a short-time Fourier transform to obtain the frequency spectrum. According to the perception characteristics of the human auditory system, the spectrum amplitude is logarithmically processed, and the endpoint intensity curve and the phase information of the peak value are output through half-wave rectification. The Pulse Code Modulation (PCM) characteristic value is extracted according to the autocorrelation characteristic of the endpoint intensity curve. This article proposes a rhythm detection algorithm based on multipath search and cluster analysis; that is, based on the clustering algorithm, it absorbs the idea of multipath tracking and proposes its own detection and tracking algorithm. It overcomes the weakness of the clustering algorithm that needs to use Musical Instrument Digital Interface (MIDI) auxiliary input to achieve the desired effect. This algorithm completely uses the PCM signal as the input, which is more robust than the clustering algorithm. The whole process is carried out in the time domain, and the amount of calculation is much smaller than the frequency domain calculation of multipath tracking, and the linear relationship with the rhythm of the music is much better than the filter bank algorithm. This algorithm can successfully detect the rhythm of the music with a strong sense of rhythm and can track the specific position of the rhythm point.

Optical switches based semiconductor optical amplifiers (SOAs) for performance characteristics enhancement by using various electrical pulse generators

Abstract This work has reported the optical switches based semiconductor optical amplifiers (SOAs) for the performance of the characteristics enhancement by using various electrical pulse generators. Max. Q, bit error rate (BER) after light detector, max. signal power (MSP), min. noise power (MNNP), and total optical power after power combiner based SOAs with various pulse generators are simulated and clarified. Return to zero (RZ) pulse code has clarified max Q than other pulse generators. The max output signal power is improved with Gaussian pulse in compared to other pulse generators. Triangle/RZ pulse generators have presented better total optical power than other proposed generators.

Image Data Compression Based on Two Hybrids Algorithms

Image compression is a way to reduce the storage space of images. Therefore, image compression process increases the performance of such as the data transmission process. This paper aims to present a new technique to compress digital image data. In this technique, two-hybrid algorithms were used to compress the image data. The first system consists of one-dimensional discrete cosine transform (DCT), differential pulse code modulation (DPCM), and Huffman code for difference signal. The second hybrid system utilizes an expert system called Learning Automata (LA) to code the difference signal obtained from the first system. A compression ratio of about (10.8:1) was obtained from the first system. The second system provides a (20.6:1) compression ratio with non-noticeable impairment. The information loss is caused by a hybrid (DCT/DPCM) system, not by the LA system. The conclusion drawn is that using two-hybrid systems to compress the image data provides a high compression ratio. Furthermore, learning automata is preferable since it removes all the redundancy in the row data. However, in learning automata, a Huffman code is determined pixel by pixel which takes a long time.

Theoretical and experimental study on reducing coherent Rayleigh noise of Rayleigh scattering distributed optical fiber sensing system using PSK pulse coding

A new method of reducing coherent Rayleigh noise (CRN) by using Phase Shift Keying (PSK) pulse coding is characterized and demonstrated theoretically and experimentally. A phase modulator driven by an arbitrary function generator is employed to encode pulses with PSK modulation, the traces obtained by coded pulses with different coding orders are superposed and averaged to reduce CRN. The experimental results validate that our method can realize about 0.49 dB amplitude fluctuation in 10 km optical fiber with 10 m spatial resolution when the coding length is 6bit. Compared with the method of FSAV, the method presented in this paper has almost the same CRN reduction effect at different spatial resolutions. From a practical point of view, the proposed method may be attractive in reducing the CRN for the Rayleigh scattering distributed optical fiber sensing system especially when the high spatial resolution and high accuracy are the main concerns.

Simulation Analysis on the Simultaneous Deployment of Brillouin Gain and Loss in Coded Brillouin Optical Time Domain Analysis (BOTDA) Fiber Sensor

We report in this paper the simulation analysis on the simultaneous deployment of Brillouin gain and Brillouin loss in the Golay coded phase-shift pulse Brillouin optical time domain analysis (PSP-BOTDA) fiber optic sensor for improving the sensing distance and at the same time obtaining high spatial resolution measurement. In this technique, the Golay coded pump pulse is alternately frequency modulated for the purpose of generating Stokes and Anti-Stokes scattering. The technique also modulates the coded pump with the return-to-zero (RZ) formats. Simulation analysis has revealed that the simultaneous use of Brillouin gain- and loss and the pulse coding technique in the PSP-BOTDA has improved the dynamic sensing range and the signal-to-noise improvement ratio (SNIR) of the sensor. By performing simulation over 500 m of fiber, we have also successfully demonstrated 10-cm of high spatial resolution measurement with the use of 1ns of coded pulse duration.

High compression efficiency image compression algorithm based on subsampling for capsule endoscopy

In this paper, a simple image compression algorithm is proposed for wireless capsule endoscopy. The proposed algorithm consists of new simplified YUV colour space, corner clipping, uniform quantization, subsampling, differential pulse code modulation and Golomb Rice code. Simplified YUV colour space is proposed based on special nature of endoscopic images and provide good results. The quantization and subsampling are used as lossy compression techniques and fixed Golomb-Rice code is used to encode residual value obtained after differential pulse code modulation operation. Here performance of different combination of quantization and subsampling techniques are analyzed based combination along with the proposed compression algorithm provides compression ratio of 89.3% and peak signal noise ratio of 45.1. the proposed algorithm provided better results as compared to various reported algorithms in literature in term of CR and PSNR.

Dynamically tuneable pre‐modulation filter for an airborne PCM/FM telemetry system

In the conventional airborne telemetry system, the pre-modulation filter with a multi-pole active Bessel filter is preferred to use in Pulse Code Modulator (PCM)/FM transmission. In the existing system, it is not possible to change the cut-off frequency for different data rates dynamically as required in the launch scenario of a long-range aerospace vehicle. In general, aerospace vehicle has multiple propellant stages to travel a desired trajectory path. Each stage gets separated from the vehicle at different instances. Each stage measurement plan is defined and correspondingly the PCM format is generated with an optimum data rate. Hence, the telemetry system is required to transmit variable data rates at various instances of a long-range aerospace vehicle from launch point to end point of a vehicle. This can be addressed by designing a dynamically tuneable pre-modulation (DTPM) filter. Here, a suitable DTPM filter scheme is proposed to mitigate variable data rate transmission in the telemetry system. The scheme is analysed as per IRIG-106 standard and simulated using MATLAB. The same has been modelled using VHDL and implemented targeting 28 nm technology Xilinx Zynq FPGA device.

A Novel All Digital Transmitter With Three-level Quadrature Differential RF-PWM

To solve spectral aliasing of digital pulse width modulation (PWM) signal during digital up-conversion, a novel all-digital transmitter architecture using three-level quadrature differential RF-PWM is presented. By using triangle wave with 90° phase shift as the reference signal of quadrature differential PWM, the pulse coding of complex modulation signal and digital up-conversion will be realized. The feasibility of the transmitter is demonstrated by deriving the output formula of 3-level quadrature differential PWM, the design requirement is also given. Simulation and experiment results with QPSK and 16QAM signals show that the proposed scheme can significantly suppress the transmitter in-band noise, eliminate even harmonics, and effectively reduce the system operating frequency.

Timing pulse code modulation as a tool for quantization noise reduction in special-purpose IT systems

The aim of the article is to examine the issue of eliminating quantization noise in the process of analogue-to-digital conversion by means of timing pulse code modulation. According to the results of the method proposed, it has established that the use of timing modulation significantly increases the number of sampling levels at a given interval without reducing the duration of individual code pulses, that, in its turn, significantly improves signal quality (readability) after the digital-to-analogue conversion on the receiving side.

Analysis of Speech Fundamental Frequencies for Different Tasks in Japanese

Speech fundamental frequency (SFF) assessment is essential for all dysphonia patients to effectively evaluate the therapeutic effects of voice therapy, especially in patients with disturbances in their voice pitch due to mutational dysphonia, Reinke's edema, or as side effects of hormone therapy. A standard method of SFF measurement remains unknown. Speech tasks such as sustained vowel phonation, counting, reading passage, and spontaneous speech have generally been used for SFF measurements. Ideally, spontaneous speech best reflects SFF; however, this task has not yet been clearly defined and is limited with regard to its adaptation to a clinical setting. A reliable task for SFF measurement in Japanese, which corresponds to a speech task that most closely reflects the value that would be observed with typical spontaneous speech, has not been investigated. This study aimed to identify a reliable speech task by measuring the SFF values elicited by different widely used speech tasks in Japanese, and assess its reliability and coefficient of determination (R2). Sixty healthy volunteers (30 men and 30 women; aged 19-30 years; mean age 22.5 years) were enrolled. All experimental procedures were performed in Japanese. The SFF values for the speech tasks were determined through the voice samples recorded using a Pulse Code Modulation (PCM) recorder. Each task, except spontaneous speech, was repeated five times, and the average fundamental frequency in each task was determined as the SFF. To assess the reliability of the SFF values across daily variations within individual speakers, the SFF measurements were repeated on two different days, separated by at least 1 week. The SFF values of sustained /a/ phonation, sustained vowel-average, counting, reading passage, and spontaneous speech had excellent reliability, in terms of their reproduction based on intraclass correlation. Significantly high SFF values were observed, in decreasing order, for sustained vowels-average, counting, reading passage, and spontaneous speech in both males and females. The highest R2 for spontaneous speech was that of reading passage in both males (R2=0.771) and females (R2=0.806) (P < 0.01). When spontaneous speech was presented as a task most reflective of daily conversation, reading passage was determined to be the reliable task to assess the therapeutic effect of voice therapy in Japanese.

A 94.2-dB SNDR 142.6-μW VCO-Based Audio ADC With a Split-ADC Differential Pulse Code Modulation Architecture

This letter presents a voltage-controlled oscillator (VCO)-based analog-to-digital converter (ADC) for audio applications using differential pulse code modulation (DPCM) to mitigate VCO voltage-to-frequency nonlinearity. To overcome VCO gain variation, a split-ADC architecture is employed where two sub-ADCs share one calibration unit, thereby enabling a quasideterministic background gain calibration with small delay and fast convergence speed. In addition, a hybrid dynamic element matching (DEM) is employed, which offers a lower delay and a smaller in-band mismatch noise compared to the segmented-tree DEM. Fabricated in 65 nm, the ADC achieves 94.2-dB SNDR, 97.3-dB SNR, and 100.3-dB DR in 20-kHz bandwidth while dissipating 142.6-μW power, yielding a DR-based FoM of 181.8 dB.

Color Image Compression System by using Block Categorization Based on Spatial Details and DCT Followed by Improved Entropy Encoder

In this paper, a new high-performance lossy compression technique based on DCT is proposed. The image is partitioned into blocks of a size of NxN (where N is multiple of 2), each block is categorized whether it is high frequency (uncorrelated block) or low frequency (correlated block) according to its spatial details, this done by calculating the energy of block by taking the absolute sum of differential pulse code modulation (DPCM) differences between pixels to determine the level of correlation by using a specified threshold value. The image blocks will be scanned and converted into 1D vectors using horizontal scan order. Then, 1D-DCT is applied for each vector to produce transform coefficients. The transformed coefficients will be quantized with different quantization values according to the energy of the block. Finally, an enhanced entropy encoder technique is applied to store the quantized coefficients. To test the level of compression, the quantitative measures of the peak signal-to-noise ratio (PSNR) and compression ratio (CR) is used to ensure the effectiveness of the suggested system. The PSNR values of the reconstructed images are taken between the intermediate range from 28dB to 40dB, the best attained compression gain on standard Lena image has been increased to be around (96.60 %). Also, the results were compared to those of the standard JPEG system utilized in the “ACDSee Ultimate 2020” software to evaluate the performance of the proposed system.

Pulse Coding Controlled Switching Converters with Notch Frequency Generation in Noise Spectrum

This paper proposes a new technology about generating notch characteristics in noise spectrum of DC-DC switching converters for the communication devices. There are very few reports about the pulse coding controlled switching converter, though it is sometimes used in digital-to-analog converter (DAC) field, to our knowledge. We have investigated the pulse coding technology to switching converters to generate automatically the notch characteristics in the noise spectrum. The notch frequency Fn in the spectrum of the switching pulses is created by two coding pulses to control the main power switch. The clock and coding pulses are automatically generated from the input frequency Fin to create the notch frequency Fn=Fin. In addition, the notch frequency is experimentally confirmed with the prototype circuit.