Evaluation of Audio Compression Artifacts

This paper deals with subjective evaluation of audio-coding systems. From this evaluation, it is found that, depending on the type of signal and the algorithm of the audio-coding system, different types of audible errors arise. These errors are called coding artifacts. Although three kinds of artifacts are perceivable in the auditory domain, the author proposes that in the coding domain there is only one common cause for the appearance of the artifact, inefficient tracking of transient-stochastic signals. For this purpose, state-of-the art audio coding systems use a wide range of signal processing techniques, including application of the wavelet transform, which is described here.

MOSRA: Joint Mean Opinion Score and Room Acoustics Speech Quality Assessment

The acoustic environment can degrade speech quality during communication (e.g., video call, remote presentation, outside voice recording), and its impact is often unknown. Objective metrics for speech quality have proven challenging to develop given the multi-dimensionality of factors that affect speech quality and the difficulty of collecting labeled data. Hypothesizing the impact of acoustics on speech quality, this paper presents MOSRA: a non-intrusive multi-dimensional speech quality metric that can predict room acoustics parameters (SNR, STI, T60, DRR, and C50) alongside the overall mean opinion score (MOS) for speech quality. By explicitly optimizing the model to learn these room acoustics parameters, we can extract more informative features and improve the generalization for the MOS task when the training data is limited. Furthermore, we also show that this joint training method enhances the blind estimation of room acoustics, improving the performance of current state-of-the-art models. An additional side-effect of this joint prediction is the improvement in the explainability of the predictions, which is a valuable feature for many applications.Comment: Submitted to Interspeech 202

Parallel task in Subjective Audio Quality and Speech Intelligibility Assessments

Tato disertační práce se zabývá subjektivním testováním jak kvality řeči, tak i srozumitelnosti řeči, prozkoumává existující metody, určuje jejich základní principy a podstaty a porovnává jejich výhody a nevýhody. Práce také porovnává testy z hlediska různých parametrů a poskytuje moderní řešení pro již existující metody testování. První část práce se zabývá opakovatelností subjektivních testování provedených v ideálních laboratorních podmínkách. Takové úlohy opakovatelnosti se provádí použitím Pearsonové korelace, porovnání po párech a jinými matematickými analýzami. Tyto úlohy dokazují správnost postupů provedených subjektivních testů. Z tohoto důvodu byly provedeny čtyři subjektivní testy kvality řeči ve třech různých laboratořích. Získané výsledky potvrzují, že provedené testy byly vysoce opakovatelné a testovací požadavky byly striktně dodrženy. Dále byl proveden výzkum pro ověření významnosti subjektivních testování kvality řeči a srozumitelnosti řeči v komunikačních systémech. Za tímto účelem bylo analyzováno více než 16 miliónů záznamů živých hovorů přes VoIP telekomunikační sítě. Výsledky potvrdily základní předpoklad, že lepší uživatelská zkušenost působí delší trvání hovorů. Kromě dosažených hlavních výsledků však byly učiněny další důležité závěry. Dalším krokem disertační práce bylo prozkoumat techniku paralelních zátěží, existující přístupy a jejich výhody a nevýhody. Ukázalo se, že většina paralelních zátěží používaných v testech byla buď fyzicky, nebo mentálně orientovaná. Jelikož subjekty ve většině případů nejsou stejně fyzicky nebo mentálně zdatní, jejich výkony během úkolů nejsou stejné, takže výsledky nelze správně porovnat. V této disertační práci je navržen nový přístup, kdy jsou podmínky pro všechny subjekty stejné. Tento přístup představuje celou řadu úkolů, které zahrnují kombinaci mentálních a fyzických zátěží (simulátor laserové střelby, simulátor řízení auta, třídění předmětů apod.). Tyto metody byly použity v několika subjektivních testech kvality řeči a srozumitelnosti řeči. Závěry naznačují, že testy s paralelními zátěží mají realističtější výsledky než ty, které jsou prováděny v laboratorních podmínkách. Na základě výzkumu, zkušeností a dosažených výsledků byl Evropskému institutu pro normalizaci v telekomunikacích předložen nový standard s přehledem, příklady a doporučeními pro zajištění subjektivních testování kvality řeči a srozumitelnosti řeči. Standard byl přijat a publikován pod číslem ETSI TR 103 503.This thesis deals with the subjective testing of both speech quality and speech intelligibility, investigates the existing methods, record their main features, as well as advantages and disadvantages. The work also compares different tests in terms of various parameters and provides a modern solution for existing subjective testing methods. The first part of the research deals with the repeatability of subjective speech quality tests provided in perfect laboratory conditions. Such repeatability tasks are performed using Pearson correlations, pairwise comparison, and other mathematical analyses, and are meant to prove the correctness of procedures of provided subjective tests. For that reason, four subjective speech quality tests were provided in three different laboratories. The obtained results confirmed that the provided tests were highly repeatable, and the test requirements were strictly followed. Another research was done to verify the significance of speech quality and speech intelligibility tests in communication systems. To this end, more than 16 million live call records over VoIP telecommunications networks were analyzed. The results confirmed the primary assumption that better user experience brings longer call durations. However, alongside the main results, other valuable conclusions were made. The next step of the thesis was to investigate the parallel task technique, existing approaches, their advantages, and disadvantages. It turned out that the majority of parallel tasks used in tests were either physically or mentally oriented. As the subjects in most cases are not equally trained or intelligent, their performances during the tasks are not equal either, so the results could not be compared correctly. In this thesis, a novel approach is proposed where the conditions for all subjects are equal. The approach presents a variety of tasks, which include a mix of mental and physical tasks (laser-shooting simulator, car driving simulator, objects sorting, and others.). Afterward, the methods were used in several subjective speech quality and speech intelligibility tests. The results indicate that the tests with parallel tasks have more realistic values than the ones provided in laboratory conditions. Based on the research, experience, and achieved results, a new standard was submitted to the European Telecommunications Standards Institute with an overview, examples, and recommendations for providing subjective speech quality and speech intelligibility tests. The standard was accepted and published under the number ETSI TR 103 503

Block-Online Multi-Channel Speech Enhancement Using DNN-Supported Relative Transfer Function Estimates

This work addresses the problem of block-online processing for multi-channel speech enhancement. Such processing is vital in scenarios with moving speakers and/or when very short utterances are processed, e.g., in voice assistant scenarios. We consider several variants of a system that performs beamforming supported by DNN-based voice activity detection (VAD) followed by post-filtering. The speaker is targeted through estimating relative transfer functions between microphones. Each block of the input signals is processed independently in order to make the method applicable in highly dynamic environments. Owing to the short length of the processed block, the statistics required by the beamformer are estimated less precisely. The influence of this inaccuracy is studied and compared to the processing regime when recordings are treated as one block (batch processing). The experimental evaluation of the proposed method is performed on large datasets of CHiME-4 and on another dataset featuring moving target speaker. The experiments are evaluated in terms of objective and perceptual criteria (such as signal-to-interference ratio (SIR) or perceptual evaluation of speech quality (PESQ), respectively). Moreover, word error rate (WER) achieved by a baseline automatic speech recognition system is evaluated, for which the enhancement method serves as a front-end solution. The results indicate that the proposed method is robust with respect to short length of the processed block. Significant improvements in terms of the criteria and WER are observed even for the block length of 250 ms.Comment: 10 pages, 8 figures, 4 tables. Modified version of the article accepted for publication in IET Signal Processing journal. Original results unchanged, additional experiments presented, refined discussion and conclusion

Evaluations on underdetermined blind source separation in adverse environments using time-frequency masking

The successful implementation of speech processing systems in the real world depends on its ability to handle adverse acoustic conditions with undesirable factors such as room reverberation and background noise. In this study, an extension to the established multiple sensors degenerate unmixing estimation technique (MENUET) algorithm for blind source separation is proposed based on the fuzzy c-means clustering to yield improvements in separation ability for underdetermined situations using a nonlinear microphone array. However, rather than test the blind source separation ability solely on reverberant conditions, this paper extends this to include a variety of simulated and real-world noisy environments. Results reported encouraging separation ability and improved perceptual quality of the separated sources for such adverse conditions. Not only does this establish this proposed methodology as a credible improvement to the system, but also implies further applicability in areas such as noise suppression in adverse acoustic environments