Exploiting correlogram structure for robust speech recognition with multiple speech sources

This paper addresses the problem of separating and recognising speech in a monaural acoustic mixture with the presence of competing speech sources. The proposed system treats sound source separation and speech recognition as tightly coupled processes. In the first stage sound source separation is performed in the correlogram domain. For periodic sounds, the correlogram exhibits symmetric tree-like structures whose stems are located on the delay that corresponds to multiple pitch periods. These pitch-related structures are exploited in the study to group spectral components at each time frame. Local pitch estimates are then computed for each spectral group and are used to form simultaneous pitch tracks for temporal integration. These processes segregate a spectral representation of the acoustic mixture into several time-frequency regions such that the energy in each region is likely to have originated from a single periodic sound source. The identified time-frequency regions, together with the spectral representation, are employed by a `speech fragment decoder' which employs `missing data' techniques with clean speech models to simultaneously search for the acoustic evidence that best matches model sequences. The paper presents evaluations based on artificially mixed simultaneous speech utterances. A coherence-measuring experiment is first reported which quantifies the consistency of the identified fragments with a single source. The system is then evaluated in a speech recognition task and compared to a conventional fragment generation approach. Results show that the proposed system produces more coherent fragments over different conditions, which results in significantly better recognition accuracy

[[alternative]]Text-Independent Speaker Identification Systems Based on Multi-Layer Gaussian Mixture Models

計畫編號：NSC92-2213-E032-026研究期間：200308~200407研究經費：541,000[[sponsorship]]行政院國家科學委員

VOICE RECOGNITION SYSTEM: SPEECH-TO-TEXT

VOICE RECOGNITION SYSTEM:SPEECH-TO-TEXT is a software that lets the user control computer functions and dictates text by voice. The system consists of two  components , first component is  for processing acoustic signal which is captured by a microphone and second component is to interpret the processed signal, then  mapping of the signal to words. Model for each letter will be built using Hidden Markov Model(HMM). Feature extraction will be done using Mel Frequency Cepstral Coefficients(MFCC). Feature training of the dataset will be done using vector quantization and Feature testing of the dataset will be done using viterbi algorithm. Home automation will be completely based on voice recognition system

Noise robust speaker verification using mel-frequency discrete wavelet coefficients and parallel model compensation

Interfering noise severely degrades the performance of a speaker verification system. The Parallel Model Combination (PMC) technique is one of the most efficient techniques for dealing with such noise. Another method is to use features local in the frequency domain. Recently, Mel-Frequency Discrete Wavelet Coefficients (MFDWCs) [1, 2] were proposed as speech features local in frequency domain. In this paper, we discuss using PMC along with MFDWCs features to take advantage of both noise compensation and local features (MFDWCs) to decrease the effect of noise on speaker verification performance. We evaluate the performance of MFDWCs using the NIST 1998 speaker recognition and NOISEX-92 databases for various noise types and noise levels. We also compare the performance of these versus MFCCs and both using PMC for dealing with additive noise. The experimental results show significant performance improvements for MFDWCs versus MFCCs after compensating the Gaussian Mixture Models (GMMs) using the PMC technique. The MFDWCs gave 5.24 and 3.23 points performance improvement on average over MFCCs for -6 dB and 0 dB SNR values, respectively. These correspond to 26.44% and 23.73% relative reductions in equal error rate (EER), respectively

Detección automática de la enfermedad de Parkinson usando componentes moduladoras de señales de voz

Parkinson’s Disease (PD) is the second most common neurodegenerative disorder after Alzheimer’s disease. This disorder mainly affects older adults at a rate of about 2%, and about 89% of people diagnosed with PD also develop speech disorders. This has led scientific community to research information embedded in speech signal from Parkinson’s patients, which has allowed not only a diagnosis of the pathology but also a follow-up of its evolution. In recent years, a large number of studies have focused on the automatic detection of pathologies related to the voice, in order to make objective evaluations of the voice in a non-invasive manner. In cases where the pathology primarily affects the vibratory patterns of vocal folds such as Parkinson’s, the analyses typically performed are sustained over vowel pronunciations. In this article, it is proposed to use information from slow and rapid variations in speech signals, also known as modulating components, combined with an effective dimensionality reduction approach that will be used as input to the classification system. The proposed approach achieves classification rates higher than 88  %, surpassing the classical approach based on Mel Cepstrals Coefficients (MFCC). The results show that the information extracted from slow varying components is highly discriminative for the task at hand, and could support assisted diagnosis systems for PD.La Enfermedad de Parkinson (EP) es el segundo trastorno neurodegenerativo más común después de la enfermedad de Alzheimer. Este trastorno afecta principalmente a los adultos mayores con una tasa de aproximadamente el 2%, y aproximadamente el 89% de las personas diagnosticadas con EP también desarrollan trastornos del habla. Esto ha llevado a la comunidad científica a investigar información embebida en las señales de voz de pacientes diagnosticados con la EP, lo que ha permitido no solo un diagnóstico de la patología sino también un seguimiento de su evolución. En los últimos años, una gran cantidad de estudios se han centrado en la detección automática de patologías relacionadas con la voz, a fin de realizar evaluaciones objetivas de manera no invasiva. En los casos en que la patología afecta principalmente los patrones vibratorios de las cuerdas vocales como el Parkinson, los análisis que se realizan típicamente sobre grabaciones de vocales sostenidas. En este artículo, se propone utilizar información de componentes con variación lenta de las señales de voz, también conocidas como componentes de modulación, combinadas con un enfoque efectivo de reducción de dimensiónalidad que se utilizará como entrada al sistema de clasificación. El enfoque propuesto logra tasas de clasificación superiores al 88  %, superando el enfoque clásico basado en los Coeficientes Cepstrales de Mel (MFCC). Los resultados muestran que la información extraída de componentes que varían lentamente es altamente discriminatoria para el problema abordado y podría apoyar los sistemas de diagnóstico asistido para EP