Intelligent speech signal processing
Dey, Nilanjan, (1984-)

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  • Intelligent speech signal processing
  • 紀錄類型: 書目-電子資源 : Monograph/item
    正題名/作者: Intelligent speech signal processing/ edited by Nilanjan Dey.
    其他作者: Dey, Nilanjan,
    出版者: London :Academic Press, : 2019.,
    面頁冊數: 1 online resource :ill. (some col.)
    內容註: Front Cover; Intelligent Speech Signal Processing; Copyright; Contents; Contributors; About the Editor; Preface; Chapter 1: Speech Processing in Healthcare: Can We Integrate?; References; Chapter 2: End-to-End Acoustic Modeling Using Convolutional Neural Networks; 2.1. Introduction; 2.2. Related Work; 2.3. Various Architecture of ASR; 2.3.1. GMM/DNN; 2.3.2. Attention Mechanism; 2.3.3. Connectionist Temporal Classification; 2.4. Convolutional Neural Networks; 2.4.1. Type of Pooling; 2.4.1.1. Max Pooling; 2.4.1.2. Average Pooling; 2.4.1.3. Stochastic Pooling; 2.4.1.4. Lp Pooling
    內容註: 2.4.1.5. Mixed Pooling2.4.1.6. Multiscale Orderless Pooling; 2.4.1.7. Spectral Pooling; 2.4.2. Types of Nonlinear Functions; 2.4.2.1. Sigmoid Neurons; 2.4.2.2. Maxout Neurons; 2.4.2.3. Rectified Linear Units; 2.4.2.4. Parameterized Rectified Linear Units; 2.4.2.5. Dropout; 2.5. CNN-Based End-to-End Approach; 2.6. Experiments and Their Results; 2.7. Conclusion; References; Chapter 3: A Real-Time DSP-Based System for Voice Activity Detection and Background Noise Reduction; 3.1. Introduction; 3.2. Microchip dsPIC33 Digital Signal Controller; 3.2.1. VAD and Noise Suppression Algorithm
    內容註: 3.3. High Pass Filter3.4. Fast Fourier Transform; 3.5. Channel Energy Computation; 3.6. Channel SNR Computation; 3.7. VAD Decision; 3.8. VAD Hangover; 3.9. Computation of Scaling Factor; 3.10. Scaling of Frequency Channels; 3.11. Inverse Fourier Transform; 3.12. Application Programming Interface; 3.13. Resource Requirements; 3.14. Microchip PIC Programmer; 3.15. Audio Components; 3.16. VAD and Background Noise Reduction Techniques; 3.17. Results and Discussion; 3.18. Conclusion and Discussion; References; Further Reading; Chapter 4: Disambiguating Conflicting Classification Results in AVSR
    內容註: 4.1. Introduction4.2. Detection of Conflicting Classes; 4.3. Complementary Models for Classification; 4.4. Proposed Cascade of Classifiers; 4.5. Audio-Visual Databases; 4.5.1. AV-CMU Database; 4.5.2. AV-UNR Database; 4.5.3. AVLetters Database; 4.6. Experimental Results; 4.6.1. Hidden Markov Models; 4.6.2. Random Forest; 4.6.3. Support Vector Machine; 4.6.4. AdaBoost; 4.6.5. Analysis and Comparison; 4.7. Conclusions; References; Chapter 5: A Deep Dive Into Deep Learning Techniques for Solving Spoken Language Identification Problems; 5.1. Introduction; 5.2. Spoken Language Identification
    內容註: 5.3. Cues for Spoken Language Identification5.4. Stages in Spoken Language Identification; 5.5. Deep Learning; 5.6. Artificial and Deep Neural Network; 5.7. Comparison of Spoken LID System Implementations with Deep Learning Techniques; 5.8. Discussion; 5.9. Conclusion; References; Chapter 6: Voice Activity Detection-Based Home Automation System for People With Special Needs; 6.1. Introduction; 6.2. Conceptual Design of the System; 6.3. System Implementation; 6.3.1. Speech Recognition; 6.3.2. System Automation; 6.3.3. Other Applications; 6.3.4. Results and Discussion
    標題: Automatic speech recognition. -
    電子資源: https://www.sciencedirect.com/science/book/9780128181300
    ISBN: 9780128181317 (electronic bk.)
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