Reconstruction of Phonated Speech from Whispers Using Formant-Derived Plausible Pitch Modulation

Whispering is a natural, unphonated, secondary aspect of speech communications for most people. However, it is the primary mechanism of communications for some speakers who have impaired voice production mechanisms, such as partial laryngectomees, as well as for those prescribed voice rest, which often follows surgery or damage to the larynx. Unlike most people, who choose when to whisper and when not to, these speakers may have little choice but to rely on whispers for much of their daily vocal interaction. Even though most speakers will whisper at times, and some speakers can only whisper, the majority of today’s computational speech technology systems assume or require phonated speech. This article considers conversion of whispers into natural-sounding phonated speech as a noninvasive prosthetic aid for people with voice impairments who can only whisper. As a by-product, the technique is also useful for unimpaired speakers who choose to whisper. Speech reconstruction systems can be classified into those requiring training and those that do not. Among the latter, a recent parametric reconstruction framework is explored and then enhanced through a refined estimation of plausible pitch from weighted formant differences. The improved reconstruction framework, with proposed formant-derived artificial pitch modulation, is validated through subjective and objective comparison tests alongside state-of-the-art alternatives

Speaker Normalization Using Cortical Strip Maps: A Neural Model for Steady State vowel Categorization

Auditory signals of speech are speaker-dependent, but representations of language meaning are speaker-independent. The transformation from speaker-dependent to speaker-independent language representations enables speech to be learned and understood from different speakers. A neural model is presented that performs speaker normalization to generate a pitch-independent representation of speech sounds, while also preserving information about speaker identity. This speaker-invariant representation is categorized into unitized speech items, which input to sequential working memories whose distributed patterns can be categorized, or chunked, into syllable and word representations. The proposed model fits into an emerging model of auditory streaming and speech categorization. The auditory streaming and speaker normalization parts of the model both use multiple strip representations and asymmetric competitive circuits, thereby suggesting that these two circuits arose from similar neural designs. The normalized speech items are rapidly categorized and stably remembered by Adaptive Resonance Theory circuits. Simulations use synthesized steady-state vowels from the Peterson and Barney [J. Acoust. Soc. Am. 24, 175-184 (1952)] vowel database and achieve accuracy rates similar to those achieved by human listeners. These results are compared to behavioral data and other speaker normalization models.National Science Foundation (SBE-0354378); Office of Naval Research (N00014-01-1-0624

Kompetensi guru dalam pengajaran amali reka bentuk dan teknologi di Sekolah Rendah Daerah Batu Pahat

Kompetensi bermaksud kemampuan atau kecekapan seseorang individu dalam melakukan sesuatu tugasan. Kompetensi juga merujuk kepada kemampuan seseorang dalam melaksanakan sesuatu yang diperolehi melalui pendidikan dan juga merujuk kepada prestasi dan perbuatan yang rasional untuk memenuhi spesifikasi tertentu di dalam pelaksanaan tugas-tugas pendidikan. Objektif kajian ini dijalankan adalah untuk mengenalpasti tahap kompetensi guru terhadap pengajaran amali Reka Bentuk dan Teknologi di Sekolah Rendah Daerah Batu Pahat. Kajian ini berbentuk tinjauan deskriptif yang menggunakan borang soal selidik sebagai instrumen kajian. Borang soal selidik yang dibina adalah berdasarkan kepada tiga elemen iaitu elemen pengetahuan, kemahiran dan sikap. Seramai 118 orang guru yang mengajar mata pelajaran ini telah terlibat sebagai responden. Data yang dikumpulkan telah dianalisis dengan menggunakan perisian Statistical Package for Social Science (SPSS) versi 19 yang melibatkan statistik skor min dan ujian-T tidak bersandar. Hasil dapatan kajian yang diperolehi menunjukkan guru-guru Reka Bentuk dan Teknoogi mempunyai tahap kompetensi yang tinggi terhadap proses pengajaran amali iaitu skor min yang diperolehi pada elemen pengetahuan adalah 4.23, elemen kemahiran adalah 4.30, dan elemen sikap adalah 4.47. Dapatan kajian juga menunjukkan tidak terdapat perbezaan yang signifikan terhadap tahap kompetensi berdasarkan jantina guru lelaki dan guru perempuan dengan nilai sigifikan melebihi 0.05 iaitu sebanyak 0.059. Beberapa cadangan untuk penambahbaikan juga dikemukan dalam kajian ini. Hasil dari dapatan kajian ini dapat digunakan sebagai cadangan garis panduan kepada guru-guru Reka Bentuk dan Teknologi untuk mencapai Standard Kompetensi Guru

‘Did the speaker change?’: Temporal tracking for overlapping speaker segmentation in multi-speaker scenarios

Diarization systems are an essential part of many speech processing applications, such as speaker indexing, improving automatic speech recognition (ASR) performance and making single speaker-based algorithms available for use in multi-speaker domains. This thesis will focus on the first task of the diarization process, that being the task of speaker segmentation which can be thought of as trying to answer the question ‘Did the speaker change?’ in an audio recording. This thesis starts by showing that time-varying pitch properties can be used advantageously within the segmentation step of a multi-talker diarization system. It is then highlighted that an individual’s pitch is smoothly varying and, therefore, can be predicted by means of a Kalman filter. Subsequently, it is shown that if the pitch is not predictable, then this is most likely due to a change in the speaker. Finally, a novel system is proposed that uses this approach of pitch prediction for speaker change detection. This thesis then goes on to demonstrate how voiced harmonics can be useful in detecting when more than one speaker is talking, such as during overlapping speaker activity. A novel system is proposed to track multiple harmonics simultaneously, allowing for the determination of onsets and end-points of a speaker’s utterance in the presence of an additional active speaker. This thesis then extends this work to explore the use of a new multimodal approach for overlapping speaker segmentation that tracks both the fundamental frequency (F0) and direction of arrival (DoA) of each speaker simultaneously. The proposed multiple hypothesis tracking system, which simultaneously tracks both features, shows an improvement in segmentation performance when compared to tracking these features separately. Lastly, this thesis focuses on the DoA estimation part of the newly proposed multimodal approach. It does this by exploring a polynomial extension to the multiple signal classification (MUSIC) algorithm, spatio-spectral polynomial (SSP)-MUSIC, and evaluating its performance when using speech sound sources.Open Acces

Improved Separation of Polyphonic Chamber Music Signals by Integrating Instrument Activity Labels

The separation of music signals is a very challenging task, especially in case of polyphonic chamber music signals because of the similar frequency ranges and sound characteristics of the different instruments to separate. In this work, a joint separation approach in the time domain with a U-Net architecture is extended to incorporate additional time-dependent instrument activity information for improved instrument track extractions. Different stages are investigated to integrate the additional information, but an input before the deepest encoder block achieves best separation results as well as highest robustness against randomly wrong labels. This approach outperforms a label integration by multiplication and the input of a static instrument label. Targeted data augmentation by incoherent mixtures is used for a trio example of violin, trumpet, and flute to improve separation results. Moreover, an alternative separation approach with one independent separation model for each instrument is investigated, which enables a more flexible architecture. In this case, an input after the deepest encoder block achieves best separation results, but the robustness is slightly reduced compared to the joint model. The improvements by additional information on active instruments are verified by using real instrument activity predictions for both the joint and the independent separation approaches

ARSTREAM: A Neural Network Model of Auditory Scene Analysis and Source Segregation

Multiple sound sources often contain harmonics that overlap and may be degraded by environmental noise. The auditory system is capable of teasing apart these sources into distinct mental objects, or streams. Such an "auditory scene analysis" enables the brain to solve the cocktail party problem. A neural network model of auditory scene analysis, called the AIRSTREAM model, is presented to propose how the brain accomplishes this feat. The model clarifies how the frequency components that correspond to a give acoustic source may be coherently grouped together into distinct streams based on pitch and spatial cues. The model also clarifies how multiple streams may be distinguishes and seperated by the brain. Streams are formed as spectral-pitch resonances that emerge through feedback interactions between frequency-specific spectral representaion of a sound source and its pitch. First, the model transforms a sound into a spatial pattern of frequency-specific activation across a spectral stream layer. The sound has multiple parallel representations at this layer. A sound's spectral representation activates a bottom-up filter that is sensitive to harmonics of the sound's pitch. The filter activates a pitch category which, in turn, activate a top-down expectation that allows one voice or instrument to be tracked through a noisy multiple source environment. Spectral components are suppressed if they do not match harmonics of the top-down expectation that is read-out by the selected pitch, thereby allowing another stream to capture these components, as in the "old-plus-new-heuristic" of Bregman. Multiple simultaneously occuring spectral-pitch resonances can hereby emerge. These resonance and matching mechanisms are specialized versions of Adaptive Resonance Theory, or ART, which clarifies how pitch representations can self-organize durin learning of harmonic bottom-up filters and top-down expectations. The model also clarifies how spatial location cues can help to disambiguate two sources with similar spectral cures. Data are simulated from psychophysical grouping experiments, such as how a tone sweeping upwards in frequency creates a bounce percept by grouping with a downward sweeping tone due to proximity in frequency, even if noise replaces the tones at their interection point. Illusory auditory percepts are also simulated, such as the auditory continuity illusion of a tone continuing through a noise burst even if the tone is not present during the noise, and the scale illusion of Deutsch whereby downward and upward scales presented alternately to the two ears are regrouped based on frequency proximity, leading to a bounce percept. Since related sorts of resonances have been used to quantitatively simulate psychophysical data about speech perception, the model strengthens the hypothesis the ART-like mechanisms are used at multiple levels of the auditory system. Proposals for developing the model to explain more complex streaming data are also provided.Air Force Office of Scientific Research (F49620-01-1-0397, F49620-92-J-0225); Office of Naval Research (N00014-01-1-0624); Advanced Research Projects Agency (N00014-92-J-4015); British Petroleum (89A-1204); National Science Foundation (IRI-90-00530); American Society of Engineering Educatio