In this paper, the possibility of using the ECG signal as an unequivocal biometric marker for authentication and identification purposes has been presented. Furthermore, since the ECG signal was acquired from 4 sources using different measurement equipment, electrodes positioning and number of patients as well as the duration of the ECG record acquisition, we have additionally provided an estimation of the extent of information available in the ECG record. To provide a more objective assessment of the credibility of the identification method, some selected machine learning algorithms were used in two combinations: with and without compression. The results that we have obtained confirm that the ECG signal can be acclaimed as a valid biometric marker that is very robust to hardware variations, noise and artifacts presence, that is stable over time and that is scalable across quite a solid (~100) number of users. Our experiments indicate that the most promising algorithms for ECG identification are LDA, KNN and MLP algorithms. Moreover, our results show that PCA compression, used as part of data preprocessing, does not only bring any noticeable benefits but in some cases might even reduce accuracy.

Diarization is an important task when work with audiodata is executed, as it provides a solution to the problem related to the need of dividing one analyzed call recording into several speech recordings, each of which belongs to one speaker. Diarization systems segment audio recordings by defining the time boundaries of utterances, and typically use unsupervised methods to group utterances belonging to individual speakers, but do not answer the question “who is speaking?” On the other hand, there are biometric systems that identify individuals on the basis of their voices, but such systems are designed with the prerequisite that only one speaker is present in the analyzed audio recording. However, some applications involve the need to identify multiple speakers that interact freely in an audio recording. This paper proposes two architectures of speaker identification systems based on a combination of diarization and identification methods, which operate on the basis of segment-level or group-level classification. The open-source PyAnnote framework was used to develop the system. The performance of the speaker identification system was verified through the application of the AMI Corpus open-source audio database, which contains 100 h of annotated and transcribed audio and video data. The research method consisted of four experiments to select the best-performing supervised diarization algorithms on the basis of PyAnnote. The first experiment was designed to investigate how the selection of the distance function between vector embedding affects the reliability of identification of a speaker’s utterance in a segment-level classification architecture. The second experiment examines the architecture of cluster-centroid (group-level) classification, i.e., the selection of the best clustering and classification methods. The third experiment investigates the impact of different segmentation algorithms on the accuracy of identifying speaker utterances, and the fourth examines embedding window sizes. Experimental results demonstrated that the group-level approach offered better identification results were compared to the segment-level approach, and the latter had the advantage of real-time processing.

This research presents an extensive comparative analysis of a selection of popular deep speaker embedding models, namely WavLM, TitaNet, ECAPA, and PyAnnote, applied in speaker verification tasks. The study employs a specially curated dataset, specifically designed to mirror the real-world operating conditions of voice models as accurately as possible. This dataset includes short, non-English statements gathered from interviews on a popular online video platform. The dataset features a wide range of speakers, with 33 males and 17 females, making a total of 50 unique voices. These speakers vary in age from 20 to 70 years old. This variety helps in thoroughly testing speaker verification models. This dataset is especially useful for research on speaker verification with short recordings. It consists of 10 clips for each person, each clip being no longer than 10 s, adding up to 500 recordings in total. The total length of all recordings is about 1 h and 30 min, which averages to roughly 100 s for each speaker. This dataset is a valuable tool for research in speaker verification, particularly for studies involving short audio clips. The performance of these models is evaluated using common biometric metrics such as false acceptance rate (FAR), false rejection rate (FRR), equal error rate (EER) and detection cost function (DCF). The results reveal that the TitaNet and ECAPA models stand out by presenting the lowest EER (1.91% and 1.71%, respectively) and thus exhibiting higher discriminative features, ensuring, on the one hand, a reduction in intra-class distance (the same speaker), and, on the other hand, maximizing the distance between different speaker embeddings. This analysis also highlights the ECAPA model’s advantageous balance of performance and efficiency, achieving an inference time of 69.43 milliseconds, slightly longer than the PyAnnote models. This study not only compares the performance of models but also provides a comparative analysis of respective model embeddings, offering insights into their strengths and weaknesses. The presented findings serve as a foundation for guiding future research in speaker verification, especially in the context of short audio samples or limited data. This may be particularly relevant for applications requiring quick and accurate speaker identification from short voice clips.

Autobalancing bridge method is widely used to measure the impedance of various objects in many fields of science and technology. The article presents an innovative approach to address auto-balancing bridge dynamic errors to extend the range of operating frequencies. The mathematical model of auto-balancing bridge, implemented using operational amplifier, was analysed, obtaining its simplified version taking into account the most important sources of errors. The concept was developed and the algorithms for compensation of the impact of dynamic errors on the measurement of admittance components were synthesized. The presented algorithms allow the correction of admittance measurements only based on “raw” results obtained with the use of an auto-balancing bridge. Methodology was described and numerical simulation was carried out on a selected example. As shown in the paper, the developed correction algorithms allow extending significantly (400 times) the range of admittance measurement frequencies. These algorithms do not require interfering the auto-balancing bridge structure, but only engage the computing power of modern measurement channels.

Automatic speech recognition (ASR) systems have become increasingly popular in recent years due to their ability to convert spoken language into text. Nonetheless, despite their widespread use, existing speaker-independent ASR systems frequently encounter challenges related to variations in speaking styles, accents, and vocal characteristics, leading to potential recognition inaccuracies. This study delves into the feasibility of personalized ASR systems that adapt to the unique voice attributes of individual speakers, thereby enhancing recognition accuracy. It provides an overview of our methodology, focusing on the design, development, and evaluation of both speaker-independent and personalized ASR systems. The dataset used included diverse speakers selected from three extensive datasets: TedLIUM-3, CommonVoice, and GoogleVoice, demonstrating the capability of our methodology to accommodate various accents and challenges of both natural and synthetic voices. In terms of signal classification and interpretation, the personalized model eclipsed the speaker-independent variant, registering an enhancement of up to ~3% for natural voices and ~10% for synthetic voices in recognition accuracy for individual speakers. Our findings demonstrate that personalized ASR systems can significantly improve the accuracy of speech recognition for individual speakers and highlight the importance of adapting ASR models to individual voices.