<pre><p>Difficulty understanding speech in background noise is the most common hearing complaint. Elucidating the neurophysiological mechanisms underlying speech intelligibility in everyday environments with multiple sound sources and distortions is hence important for any technology that aims to improve real-world listening. Using a combination of behavioral, electroencephalography (EEG), and computational modeling experiments, this dissertation provides insight into how the brain analyzes such complex scenes, and what roles different acoustic cues play in facilitating this process and in conveying phonetic content. Experiment #1 showed that brain oscillations selectively track the temporal envelopes (i.e., modulations) of attended speech in a mixture of competing talkers, and that the strength and pattern of this attention effect differs between individuals. Experiment #2 showed that the fidelity of neural tracking of attended-speech envelopes is strongly shaped by the modulations in interfering sounds as well as the temporal fine structure (TFS) conveyed by the cochlea, and predicts speech intelligibility in diverse listening environments. Results from Experiments #1 and #2 support the theory that temporal coherence of sound elements across envelopes and/or TFS shapes scene analysis and speech intelligibility. Experiment #3 tested this theory further by measuring and computationally modeling consonant categorization behavior in a range of background noises and distortions. We found that a physiologically plausible model that incorporated temporal-coherence effects predicted consonant confusions better than conventional speech-intelligibility models, providing independent evidence that temporal coherence influences scene analysis. Finally, results from Experiment #3 also showed that TFS is used to extract speech content (voicing) for consonant categorization even when intact envelope cues are available. Together, the novel insights provided by our results can guide future models of speech intelligibility and scene analysis, clinical diagnostics, improved assistive listening devices, and other audio technologies.</p></pre>

Dialogue Enhancement (DE) is a process used in audio delivery systems to improve the clarity, intelligibility, and overall quality of the spoken dialogue in audio content. It is primarily used when dialogue is masked by music, surrounding noise, or other audio sources. This thesis project involves experiments to find the optimal time and frequency resolution needed for a DE system. The time resolution focuses on experimenting with various attack/release times for a DE system. The frequency domain analysis investigates whether people prefer a noise spectrum-dependent gain over a conventional full-band gain. The research methodology comprises three main parts. The first part focuses on system setup and choosing content/vectors to be used for the experiments. Next, the experiments are designed for time and frequency resolution. An exponential smoothing model is used to amplify/attenuate the dialogue stream at various times of attack/release. For the frequency counterpart, a banded gain model is designed which uses banded noise levels as input. Subsequently, a modified subjective listening test is designed to evaluate the experiments designed. The responses recorded for various types of content-noise combinations from the listeners are recorded and analyzed. Finally, the main outcome of this research emphasizes the advantages of a DE system. Further, it paves the way for further exploration of DE models and rigorous testing schemes with expert listeners. / Dialogue Enhancement (DE) är en process som används i ljudleveranssystem för att förbättra tydligheten, förståeligheten och den övergripande kvaliteten på den talade dialogen i ljudinnehåll. Det används främst när dialog maskeras av musik, omgivande brus eller andra ljudkällor. Detta examensarbete omfattar experiment för att hitta den optimala tids- och frekvensupplösningen för ett DEsystem. Tidsupplösningsexperimenten fokuserar på olika attack- och releasetider för ett DE-system. Frekvensdomänanalysen undersöker om människor föredrar en brusspektrumberoende förstärkning framför en konventionell fullbandsförstärkning. Forskningsmetodiken består av tre huvuddelar. Den första delen fokuserar på systeminställning och val av innehåll/vektorer som ska användas för experimenten. Därefter designas tids- och frekvensupplösningsexperimenten. En exponentiell tidsenvelopp används för att förstärka/dämpa dialogen vid olika tider för attack/release. För frekvensdomänexperimenten används en bandad förstärkningsmodell som använder bandade brusnivåer som insignal. I den tredje delen utformas ett subjektivt lyssningstest för att utvärdera experimenten. Lyssnarnas svar för olika typer av innehåll-bruskombinationer registreras och analyseras. Det huvudsakliga resultatet av denna forskning betonar fördelarna med ett DEsystem. Vidare banar det väg för utforskning av fler DE-modeller och rigorösa testscheman med expertlyssnare.

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