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Duration 14 hours
Course Outline
Introduction to Speech Recognition Technologies
- The historical development and evolution of speech recognition
- Roles of acoustic models, language models, and decoding processes
- Contemporary architectures: RNNs, transformers, and Whisper
Fundamentals of Audio Preprocessing and Transcription
- Managing audio formats and sample rates
- Techniques for cleaning, trimming, and segmenting audio files
- Converting audio to text: comparing real-time and batch processing
Practical Work with Whisper and Alternative APIs
- Setup and utilization of OpenAI Whisper
- Accessing cloud-based APIs (such as Google and Azure) for transcription
- Analyzing differences in performance, latency, and cost
Managing Language Variations, Accents, and Domain-Specific Adaptation
- Processing content across multiple languages and accents
- Implementing custom vocabularies and improving noise resistance
- Addressing specialized terminology in legal, medical, or technical contexts
Structuring Output and System Integration
- Incorporating timestamps, punctuation, and speaker identification
- Exporting data to text, SRT, or JSON formats
- Embedding transcriptions into applications or database systems
Practical Labs for Real-World Use Cases
- Transcribing content from meetings, interviews, or podcasts
- Developing voice-to-text command interfaces
- Generating real-time captions for video and audio streams
Model Evaluation, Constraints, and Ethical Considerations
- Assessing accuracy metrics and conducting model benchmarking
- Examining bias and fairness within speech models
- Navigating privacy issues and compliance standards
Conclusion and Future Directions
Requirements
- Foundational knowledge of general AI and machine learning principles
- Familiarity with audio or media file formats and related tools
Target Audience
- Data scientists and AI engineers handling voice data
- Software developers creating transcription-based applications
- Organizations seeking to leverage speech recognition for automation