Hasan M Kadhim
@uomustansiriyah.edu.iq
Electrical Engineering Department/ College of Engineering
Mustansiriyah University
EDUCATION
PhD, DSP, Newcastle University, UK.
MSc, Electronic and Communications, Mustansiriyah University, Iraq.
BSc, Electrical Engineering, Baghdad University, Iraq.
RESEARCH, TEACHING, or OTHER INTERESTS
Signal Processing, Artificial Intelligence, Electrical and Electronic Engineering, Statistics and Probability
Scopus Publications
Scholar Citations
Scholar h-index
Scholar i10-index
Scopus Publications
Ruaa N. Ismael and Hasan M. Kadhim
University of Basrah - College of Engineering
This study proposes a blind speech separation algorithm that employs a single-channel technique. The algorithm’s input signal is a segment of a mixture of speech for two speakers. At first, filter bank analysis transforms the input from time to time-frequency domain (spectrogram). Number of sub-bands for the filter is 257. Non-Negative Matrix Factorization (NNMF) factorizes each sub-band output into 28 sub-signals. A binary mask separates each sub-signal into two groups; one group belongs to the first speaker and the other to the second speaker. The binary mask separates each sub-signal of the (257×28) 7196 sub-speech signals. That separation cannot identify the speaker. Identification of the sub-signal speaker for each sub-signal is achieved by speaker clustering algorithms. Since speaker clustering cannot process without speaker segmentation, the standard windowed-overlap frames have been used to partition the speech. The speaker clustering process fetches the extracted phase angle from the spectrogram (of the mixture speech) and merges it into the spectrogram (of the recovered speech). Filter bank synthesizes these signals to produce a full-band speech signal for each speaker. Subjective tests denote that the algorithm results are accepted. Objectively, the researchers experimented with 66 mixture chats (6 females and 6 males) to test the algorithm. The average of the SIR test is 11.1 dB, SDR is 1.7 dB, and SAR is 2.8 dB.
Ansam Q. Kamil, Hasan M. Kadhim, and Riyam M. Alsammarraie
IEEE
The most important challenge in antenna design is increasing its gain because antenna gain is considered one of the main factors in determining antenna performance for applications in communication engineering. The paper tackles antenna enhancement by optimum antenna design for geometrical parameters of Rectangular-Shaped Slotted Single-Band Microstrip Patch Antenna. To increase the gain, the researchers used an optimization process of slot insertion in the antenna patch plane. The sizes of slots in the patch have been effectively changed to increase the gain. The design improved the directivity of the antenna as well. The dimensions of the patch for the array-shaped antenna are (21×12×1.6) mm. Its substrate dimensions are (30x30x1.6) mm. The design is tested by the frequencies 3.1 GHz-10.6 GHz Ultra-Wideband. For single-band 10.4 GHz, the proposed antenna gain is 5.06 dB, and VSWR is less than 2. The design is simulated successfully by computer simulation technology. The research technique causes gain enhancement with the range of 1.2 dB to 5.06 dB. The technique provides high gain at a single band for wireless communication applications.
Amal Ibrahim Nasser, Ammar Ali Sahrab, and Hasan M. Kadhim
Institute of Advanced Engineering and Science
<p><span lang="EN-US">In this research, we will introduce implementation requirements of a remote wireless control and monitoring unit of industrial production lines automatically controlled using programmable logic controller (PLC). PLC is capable of collecting different types of data and converting them into electrical signals that can be controlled by the industrial network using supervisory control and data acquisition (SCADA) systems. SCADA will be installed in the main server inside the control unit. The PLC will be used as a decision maker of the received signals for the industrial lines that comes from a group of detectors (sensors/transducers). The output of the PLC processor will trigger the engines, according to a specific industrial process management program. The processed data could be transferred through wireless or wired method. The wireless approach will be shown in this study, along with two other ways to implement it.</span></p>
Ruaa N. Ismael, Hasan M. Kadhim, and Susanto B. Sulistyo
IEEE
The paper summarizes two algorithms to process spontaneous conversation, which contain two speech formats, a mixture, and dialogue of two speakers. Mixture format (called overlapped speech), when they are speaking simultaneously. Dialogue is when only one speaker is speaking and the other is silent. The first algorithm is overlapped speech detection. The algorithm processes the spontaneous conversation to produce two segregated mixture and dialog speech segments. The second algorithm supposes that the speaker diarization corpus processes the dialogue speech segments. The corpus outputs two sets of speech segments, first for the first speaker and second for the second speaker. Inputs of the second algorithm are two. The first is mixture speech (output of the first algorithm). Since the corpus output segments belong to the mixture format speakers, these segments assist the separation process (informed speech separation). The two algorithms are tested for 66 spontaneous conversations. The average EMISS of the overlapped speech algorithm is 0.4%. For the speech separation, the average SDR is 3.05 dB, SIR 9.8dB, and SAR 4.65 dB. On a uniform filter bank domain, the mathematical model of this paper’s informed speech separation algorithm exploited Non-Negative Matrix Factorization NNMF.
Hasan M. Kadhim, Alaa H. Ahmed, and Saif A. Abdulhussien
Universitas Ahmad Dahlan
According to Fourier analysis, any periodic function can be analyzed as an infinite series of trigonometric functions (sets of sines and cosines). The kernel of decay cosine yields an extension for the previous frequency-based, sieve-type detection algorithm by giving smooth peaks for decaying amplitudes with the harmonics of the signal correlation. The sequential outline of the RAPT algorithm is: 1) Providing speech samples with their sampling rate and with a reduced sampling rate. 2) Periodically, computing normalized cross-correlation function (NCCF) of the reduced sampling rate speech signal with lags in the F0 range. 3) Indicating the locations of maximum at the 1st pass of NCCF. 4) For the vicinity of the peaks in that 1st pass, calculate the NCCF for the original sampling rate. 5) Again, finding the maximum in that NCCF. Obtaining the location and amplitude of the modified peak. 6) For each peak obtained from the NCCF (high resolution), estimate the F0 of the processed frame. 7) The hypothesis of the frame for unvoiced/voiced is advanced for each frame. 8) Finding the group of the NCCF peaks via optimization process for the unvoiced/voiced hypotheses for all the frames which have the best match with the above characteristics. 9) Using the well-known speech pitch tracking algorithm (PTA), RAPT has the following differences: - PTA computes the NCCF in the linear prediction coding (LPC).
Musa H. Wali, Ali Khalid Jassim, and Hasan Ma Almgotir
Institute of Advanced Engineering and Science
To obtain a high data rate that is commensurate with the growing demand for internet services, the fifth generation (5G) cellular networks will use the bandwidth beyond 6 GHz, called millimeters waves (mm-waves), to obtain a higher. The first phase (phase I) of the 5G network design for high user density, where the optimized microcells are deployed at carrier frequency 700 MHz with 20 MHz bandwidth. The second phase (phase II) of the design consists of the deployment of microcells which are operating at 3.6 GHz with 100 MHz bandwidth; this phase is planned to cover 200000 users within the province. The third phase (phase III) of the design is represented by the deployment of picocells, which are planned to operate at 26 GHz frequency and bandwidth 500 MHz; this phase is planned to cover 3,500,000 users within the province. Two types of modulation are adopted for the network (orthogonal frequency division multiplexing (OFDM) and 256 quadrature amplitude modulation (QAM)); the overall performance of the network is studied with regards to the percentage of coverage, power overlapping ratio, frequency interference, and quality of service (QoS).
Hasan Ma Almgotir, Enaam A. Khaliq Ali, Wedian H. Abd al ameer, and Mustafa A. Fadel Al-Qaisi
Institute of Advanced Engineering and Science
This research presents a study, modelling and simulation of the piezoelectric material work as filters (piezoelectric filter) used to eliminate the harmonics in power electronic circuits, high order harmonics are generating due to the high switching frequencies and circuit equipment, detailed simulation is achieved for the piezoelectric filter tested in full-bridge DC/DC converter circuit with resistive load works as dc power supply (12 to 48 volt). As a result, the uses of piezoelectric filters have a great impact on harmonics elimination, which leads to reduce the overall total harmonic distortion leads to increase the efficiency, as well as the output voltage from the dc power supply remain constant by varying the load resistance over a wide range. The dc power supply circuit including the piezoelectric filter has been simulated using PSIM (V9.1) power electronic circuit simulation software.
Salah M. Swadi, Mauwafak A. Tawfik, Emad N. Abdulwahab, and Hasan Almgotir Kadhim
IEEE
This work focus on the application of Fuzzy-Backstepping controller, based on optimization method, for trajectory tracking problem. The control algorithm based on the errors in postures of mobile robot which feed to the Fuzzy-Backstepping controller to generate correction signals that transport to torque for each driven wheel. By means of dynamic model of mobile robot, these torques are used to compute the linear and angular speed to reach the desired pose. In this work, a dynamics model of mobile robot was driven for the case where the centroid, of mobile robot platform, is not coincide with reference frame of mobile robot, while the inertia is counted for. The Evolutionary Algorithm has been used to modify the parameters of the proposed controller for wheeled mobile robot. Simulation results show the effectiveness of the proposed control algorithm from tracking S-shape trajectory. The results show a good matching between the desired trajectory and the simulation one, while the error in posture goes to zero rapidly.
Sadiq. K. Ahmed, Madhukar Chandra, and Hasan Almgotir Kadhim
IEEE
This paper presents a novel approach to enhance the cross polarization discrimination in a dual linear polarization microstrip patch antenna at the frequency of 5.5 GHz. Two different designs of a dual linear polarization antenna using metamaterials are considered. In the first design, microstrip patch antenna is loaded with two pairs of spiral ring resonators, and in the second design, two orthogonal microstrip feed lines are loaded with pairs of split ring resonators. The addition of metamaterial inclusions to the antenna structure allow compensation for an asymmetric current distribution flow on the patch antenna and thus result in symmetrical current distribution on it. This compensation leads to a significant improvement in the cross polarization discrimination in comparison to the conventional dual linear polarized antenna. The simulation shows an improvement of 12 dB for the first design, and 6.6 dB for the second design in cross polarization discrimination as compared to the conventional antenna.
Hasan Almgotir Kadhim, Lok Woo, and Satnam Dlay
IEEE
The paper describes a novel method that improvises the procedure for supervised speaker diarization. The procedure supposes that the database of the speakers is available. Initially, the database and observation signal of the speakers, are prepared. The audio features has been extracted from the database and the observation signal. Instead of the using of one of Mel Frequency Cepstral Coefficient, Perceptual Linear Prediction, or Power Normalized Cepstral Coefficients, a combination of all of them have been used. The combination form of these features is independent, i.e. They are concatenated in the feature matrix. The comparison between features of observation signal and statistical properties of database features, has been made. The comparing procedure is used to make the decision of the logical mask of the comparison. Both of bottom-up and top-down scenarios collaborate to complete the last decisions successfully. Diarization Error Rate test denotes that combination of features has less than errors than any one alone.
Hasan Almgotir Kadhim, Lok Woo, and Satnam Dlay
IEEE
To simplify the jobs of speaker diarization and speech separation, at first, speech signal should be segregated to two speech formats, dialog and mixture. This paper describes a new algorithm which achieves that first step efficiently. The algorithm is based on Perceptual Linear Predictive feature extraction, optimized k-means and both top-down & bottom-up scenarios. After extracting features of the observation signal, k-means clusters the statistical properties such as variances of the PDF (histogram) of clustered extracted features. k-means is optimized by discounting the worst pattern of clustering step through doing the k-means procedure twice. The feedback loop is necessary for the guiding of the optimized k-means by exploiting the attributes of ordinary k-means. The results of segregation are excellent. The calculated diarization error rate of outputs is very limited.
Sadiq Kadhim Ahmed Aqbi and Madhukar Chandra
UK Simulation Society
Hasan Almgotir Kadhim, Wai Lok Woo, and Satnam Dlay
UK Simulation Society