Project notebook / Audio ML
ImplementedSpeech-to-Text Robustness & ASR Evaluation
I built a transcript-evaluation pipeline that aligns hypotheses to reference speech, separates substitution/deletion/insertion errors and reports corpus-weighted WER across noise, telephony and code-switching slices.
What I built
- Implemented Unicode-normalized word and character error rates with dynamic-programming edit alignment.
- Built slice-aware corpus aggregation with separate error counts for clean, noisy, telephony and code-switching examples.
- Built a JSONL evaluation CLI and deterministic test fixtures, so an actual recognizer adapter can be compared without changing the scoring contract.
Experiments & checks
- The included four-pair illustrative corpus scores 4 word errors across 27 reference words (WER 14.8%). Those hypotheses are authored fixtures, not output from a trained recognizer. The scoring algorithm is implemented and unit tested; plugging in an ASR model is a separate integration.
Notes & limitations
A single WER number can hide where recognition breaks. A caller reading an account number through a narrowband phone connection has a different failure profile from a quiet-room dictation sample.
This implementation evaluates transcripts; it does not train, fine-tune or serve an ASR model. Actual accent, noise, code-switching and language performance require representative labeled recordings and speaker-separated test sets.
A model can understand the sentence and still get the account number wrong. The database tends to care about the number.
Screenshots & media

Project updates
Dated notes tied to actual revisions of the site, not a simulated activity feed.
Audio experiments published
Added five implemented audio-ML projects with local experiments and measurement limits.
Site revision 8d75d02
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