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Speech-to-Text in Products: How Modern ASR Works and How to Use It Well
Speech recognition has improved dramatically with large end-to-end models. This guide explains how modern ASR works, how to measure accuracy, how to choose a service or model, and the design and privacy considerations for products that use voice.
TopicsAI and Machine LearningGenerative AI and LLMsMobile App Development
Key takeaways
- Modern automatic speech recognition (ASR) uses end-to-end neural models trained on very large amounts of audio, such as OpenAI’s Whisper, trained on 680,000 hours.
- Measure accuracy with word error rate on your own recordings; published benchmarks rarely match real conditions.
- Accents, background noise, domain vocabulary and overlapping speakers are the main sources of errors.
- Choose between cloud APIs and self-hosted models by accuracy, latency, cost at volume, languages and data privacy.
- Voice data is personal data; obtain consent, minimise retention and secure recordings and transcripts.
On this page
Speech-to-text, or automatic speech recognition (ASR), turns spoken language into text. It powers meeting transcription, call centre analytics, voice assistants, medical documentation, captions and accessibility features. Accuracy has improved markedly with large neural models, making voice practical in many more products.
How modern ASR works
Older systems combined separate acoustic, pronunciation and language models. Modern systems are typically end-to-end neural networks that map audio directly to text. A notable example is OpenAI’s Whisper, described in a 2022 paper, which was trained on 680,000 hours of multilingual audio and showed that large-scale weak supervision produces robust recognition across accents and conditions.
A typical pipeline includes:
- Audio capture and pre-processing: sample rate conversion, noise reduction, voice activity detection.
- Recognition: the model converts speech segments to text.
- Speaker diarisation: identifying who spoke when, if needed.
- Post-processing: punctuation, capitalisation, number formatting and custom vocabulary.
- Downstream use: search, summarisation, analytics or commands.
Measuring accuracy
Word error rate (WER) is the standard measure: the number of substitutions, deletions and insertions divided by the number of words in a reference transcript. Lower is better. Evaluate on a sample of your own recordings with human reference transcripts, broken down by speaker group, environment and topic, and pay particular attention to critical terms such as names, numbers and medication names.
What causes errors
| Factor | Mitigation |
|---|---|
| — | — |
| Background noise and poor microphones | Better capture hardware, noise suppression, guidance to users |
| Accents and dialects | Choose models trained on diverse data; evaluate per group |
| Domain vocabulary | Custom vocabularies, phrase hints or fine-tuning |
| Overlapping speakers | Separate channels where possible; diarisation |
| Phone-quality audio | Models suited to telephony audio |
| Long recordings | Chunking with overlap; consistent timestamps |
Most production issues come from audio quality rather than the model.
APIs or self-hosted models?
| Factor | Cloud ASR APIs | Self-hosted open models |
|---|---|---|
| — | — | — |
| Setup | Fast | Requires infrastructure and expertise |
| Accuracy | Strong, with domain features | Strong for many languages; tunable |
| Latency | Good streaming options | Depends on hardware |
| Cost | Per minute; grows with volume | Infrastructure; cheaper at high steady volume |
| Data control | Provider terms apply | Data stays in your environment |
Test both on your audio before deciding.
Product use cases
- Meeting and interview transcription with summaries and action items.
- Contact centre analytics: transcripts for quality review, compliance checks and topic trends.
- Clinical documentation: ambient notes reviewed by clinicians — see our guide to AI in EHR systems.
- Captions and accessibility for video and live events.
- Voice commands and assistants in apps, vehicles and field work.
Designing voice features
- Show live transcription where users speak to the product, so they can correct errors immediately.
- Confirm critical values such as amounts, dates and names before acting on them.
- Offer text alternatives for users who cannot or prefer not to speak.
- Handle silence and interruptions gracefully, with clear prompts.
- Combine with language models carefully: summaries and extracted actions should link back to the transcript so users can verify them.
Cost considerations
Cloud ASR is usually priced per minute of audio, with different rates for streaming, batch and enhanced models. At low volumes APIs are cheaper and simpler; at sustained high volumes, self-hosting on GPUs can cost less but adds operations work. Model your expected minutes per month, required latency and languages before deciding, and remember that diarisation, translation and summarisation may be priced separately.
Privacy and compliance
- Inform people that they are being recorded and transcribed, and obtain consent where required.
- Minimise retention of audio; keep transcripts only as long as needed.
- Redact sensitive information such as card numbers or health details.
- Check where providers process audio and whether it is used for training.
Frequently asked questions
How accurate is speech-to-text today?
For clear audio in widely spoken languages, very accurate; accuracy drops with noise, accents, jargon and overlapping speech. Measure on your own recordings.
Can ASR handle multiple languages in one conversation?
Many modern models handle language switching, but accuracy varies. Test with representative audio.
Is real-time transcription possible?
Yes, with streaming APIs or optimised models, typically with a short delay and lower accuracy than offline processing.
How do we improve accuracy for our terminology?
Use custom vocabulary features, provide context or fine-tune on your own labelled audio.
Sources and methodology
Facts in this article were checked against the sources below on 6 November 2026. Directory data (ratings, review counts, rate bands and team sizes) changes over time and is shown as published on the date accessed. Illustrative examples are labelled as such and are not quotes.
- Radford et al.: Robust Speech Recognition via Large-Scale Weak Supervision (arXiv, 2022) — accessed 6 November 2026. Whisper trained on 680,000 hours of multilingual audio; robust performance without fine-tuning
Related reading
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