Improving GANs for Speech Enhancement

Generative adversarial networks (GAN) have recently been shown to be efficient for speech enhancement. However, most, if not all, existing speech enhancement GANs (SEGAN) make use of a single generator to perform one-stage enhancement mapping. In this work, we propose to use multiple generators that are chained to perform multi-stage enhancement mapping, which gradually refines the noisy input signals in a stage-wise fashion. Furthermore, we study two scenarios: (1) the generators share their parameters and (2) the generators' parameters are independent. The former constrains the generators to learn a common mapping that is iteratively applied at all enhancement stages and results in a small model footprint. On the contrary, the latter allows the generators to flexibly learn different enhancement mappings at different stages of the network at the cost of an increased model size. We demonstrate that the proposed multi-stage enhancement approach outperforms the one-stage SEGAN baseline, where the independent generators lead to more favorable results than the tied generators. The source code is available at http://github.com/pquochuy/idsegan.Comment: This letter has been accepted for publication in IEEE Signal Processing Letter

Improvement of Speech Perception for Hearing-Impaired Listeners

Hearing impairment is becoming a prevalent health problem affecting 5% of world adult populations. Hearing aids and cochlear implant already play an essential role in helping patients over decades, but there are still several open problems that prevent them from providing the maximum benefits. Financial and discomfort reasons lead to only one of four patients choose to use hearing aids; Cochlear implant users always have trouble in understanding speech in a noisy environment. In this dissertation, we addressed the hearing aids limitations by proposing a new hearing aid signal processing system named Open-source Self-fitting Hearing Aids System (OS SF hearing aids). The proposed hearing aids system adopted the state-of-art digital signal processing technologies, combined with accurate hearing assessment and machine learning based self-fitting algorithm to further improve the speech perception and comfort for hearing aids users. Informal testing with hearing-impaired listeners showed that the testing results from the proposed system had less than 10 dB (by average) difference when compared with those results obtained from clinical audiometer. In addition, Sixteen-channel filter banks with adaptive differential microphone array provides up to six-dB SNR improvement in the noisy environment. Machine-learning based self-fitting algorithm provides more suitable hearing aids settings. To maximize cochlear implant users’ speech understanding in noise, the sequential (S) and parallel (P) coding strategies were proposed by integrating high-rate desynchronized pulse trains (DPT) in the continuous interleaved sampling (CIS) strategy. Ten participants with severe hearing loss participated in the two rounds cochlear implants testing. The testing results showed CIS-DPT-S strategy significantly improved (11%) the speech perception in background noise, while the CIS-DPT-P strategy had a significant improvement in both quiet (7%) and noisy (9%) environment

Noise processing in the auditory system with applications in speech enhancement

Abstract: The auditory system is extremely efficient in extracting auditory information in the presence of background noise. However, speech enhancement algorithms, aimed at removing the background noise from a degraded speech signal, are not achieving results that are near the efficacy of the auditory system. The purpose of this study is thus to first investigate how noise affects the spiking activity of neurons in the auditory system and then use the brain activity in the presence of noise to design better speech enhancement algorithms. In order to investigate how noise affects the spiking activity of neurons, we first design a generalized linear model that relates the spiking activity of neurons to intrinsic and extrinsic covariates that can affect their activity, such as noise. From this model, we extract two metrics, one that shows the effects of noise on the spiking activity and another the relative effects of vocalization compared to noise. We use these metrics to analyze neural data, recorded from a structure of the auditory system named the inferior colliculus (IC), while presenting noisy vocalizations. We studied the effect of different kinds of noises (non-stationary, white and natural stationary), different vocalizations, different input sound levels and signal-to-noise ratios (SNR). We found that the presence of non-stationary noise increases the spiking activity of neurons, regardless of the SNR, input level or vocalization type. The presence of white or natural stationary noises however causes a great diversity of responses where the activity of sites could increase, decrease or remain unchanged. This shows that the noise invariance previously reported in the IC depends on the noisy conditions, which had not been observed before. We then address the problem of speech enhancement using information from the brain's processing in the presence of noise. It has been shown before that the brain waves of a listener strongly correlates with the speaker to which the listener attends. Given this, we design two speech enhancement algorithms with a denoising autoencoder structure, namely the Brain Enhanced Speech Denoiser (BESD) and U-shaped Brain Enhanced Speech Denoiser (U-BESD). These algorithms take advantage of the attended auditory information present in the brain activity of the listener to denoise a multi-talker speech. The U-BESD is built upon the BESD with the addition of skip connections and dilated convolutions. Compared to previously proposed approaches, BESD and U-BESD are trained in a single neural architecture, lowering the complexity of the algorithm. We investigate two experimental settings. In the first one, the attended speaker is known, referred to as the speaker-specific setting, and in the second one no prior information is available about the attended speaker, referred to as the speaker-independent setting. In the speaker-specific setting, we show that both the BESD and U-BESD algorithms surpass a similar denoising autoencoder. Moreover, we also show that in the speaker-independent setting, U-BESD surpasses the performance of the only known approach that also uses the brain's activity.Le système auditif est extrêmement efficace pour extraire de l’information pertinente en présence d’un bruit de fond. Par contre, les algorithmes de rehaussement de la parole, visant à supprimer le bruit d’un signal de parole bruité, n’atteignent pas des résultats proches de l’efficacité du système auditif. Le but de cette étude est donc d’abord d’étudier comment le bruit affecte l’activité neuronale dans le système auditif, puis d’utiliser l’activité cérébrale en présence de bruit pour concevoir de meilleurs algorithmes de rehaussement. Afin d’étudier comment le bruit peut affecter l’activité des neurones, nous concevons d’abord un modèle linéaire généralisé qui relie l’activité des neurones aux covariables intrinsèques et extrinsèques qui peuvent affecter leur activité, comme le bruit. De ce modèle, nous extrayons deux métriques, l’une qui permet d’étudier les effets du bruit sur l’activité neuronale et l’autre les effets relatifs sur cette activité de la vocalisation par rapport au bruit. Nous utilisons ces métriques pour analyser l’activité neuronale d’une structure du système auditif, nomée le colliculus inférieur (IC), enregistrée lors de la présentation de vocalisations bruitées. Nous avons étudié l’effet de différents types de bruits, différentes vocalisations, différents niveaux sonores d’entrée et différents rapports signal sur bruit (SNR). Nous avons constaté que la présence de bruit non stationnaire augmente l’activité des neurones, quel que soit le SNR, le niveau d’entrée ou le type de vocalisation. La présence de bruits stationnaires blancs ou naturels provoque cependant une grande diversité de réponses où l’activité des sites d’enregistrement pouvait augmenter, diminuer ou rester inchangée. Cela montre que l’invariance du bruit précédemment signalée dans l’IC dépend des conditions de bruit, ce qui n’avait pas été observé auparavant. Nous abordons ensuite le problème du rehaussement de la parole en utilisant de l’information provenant du cerveau. Il a été démontré auparavant que les ondes cérébrales d’un auditeur sont fortement corrélées avec le locuteur auquel l’auditeur porte attention. Compte tenu de cette corrélation, nous concevons deux algorithmes de rehaussement de la parole, le Brain Enhanced Speech Denoiser (BESD) et le U-shaped Brain Enhanced Speech Denoiser (U-BESD), qui tirent parti de l’information présente dans l’activité cérébrale de l’auditeur pour débruiter un signal de parole multi-locuteurs. L’U-BESD est construit à partir du BESD avec l’ajout de sauts de connexions (skip connections) et de convolutions dilatées. De plus, BESD et U-BESD sont constitués respectivement d’un seul réseau qui nécessite un seul entraînement, ce qui réduit la complexité de l’algorithme en comparaison avec les approches existantes. Nous étudions deux conditions expérimentales. Dans la première, le locuteur auquel l’auditeur porte attention est connu, et dans la seconde, ce locuteur n’est pas connu. Dans le cadre du locuteur connu, nous montrons que les algorithmes BESD et U-BESD surpassent un autoencodeur similaire. De plus, nous montrons également que dans le cadre du locuteur inconnu, le U-BESD surpasse les performances de la seule approche existante connue qui utilise également l’activité cérébrale