Real-Time Voice: A Comprehensive Survey of Automatic Speech Recognition and Transcription
Human-computer interaction has been transformed by Automatic Speech Recognition (ASR), opening the door to uses including voice-activated devices, assistive technology, and real-time transcription. But it’s still difficult to get reliable results when dealing with different accents, loud surroundings, and domain-specific jargon. In this research, a high-performance ASR system utilizing a WaveNet-based architecture and sophisticated preprocessing methods such as SubSpectral Normalization and Mel-Frequency Cepstral Coefficients is presented. The suggested approach combines linguistic decoders with causal and dilated convolutions to improve contextual comprehension. Time-stretching and noise addition are two data augmentation techniques that guarantee resilience to real-world fluctuations. Results from experiments show notable gains in generalization, scalability, and transcription accuracy. By providing useful answers for a range of applications, including accessibility tools and virtual assistants, this work advances ASR technologies.
Authors
- Debangshu Dey (ORCID: https://orcid.org/0000-0002-7163-628X)
- Debashis Saha (ORCID: https://orcid.org/0000-0003-0523-9646)
- Ramendra N. Saha (ORCID: https://orcid.org/0000-0002-5494-2584)
- Debrupa Pal (ORCID: https://orcid.org/0000-0001-6351-9625)
- Mrinmoy Guria
- Apurba K. Das (ORCID: https://orcid.org/0000-0002-8141-6854)
Publication Details
- Published
- 2025-02-07
- DOI
- https://doi.org/10.1109/iccece61355.2025.10940301
- Primary Topic
- Speech Recognition and Synthesis
- Type
- article
- Field-Weighted Citation Impact
- 0.00