
You send audio in, you get text out. That is speech-to-text in a single sentence. But choosing the right STT API for your application involves much deeper decisions.
In 2026, the speech-to-text landscape has matured significantly. Transformer-based models have replaced legacy approaches, real-time streaming supports dozens of languages, and accuracy rivals human transcriptionists in controlled conditions. The challenge is no longer "can AI transcribe speech?" but "which API handles my specific audio, languages, and scale requirements best?"
Speech-to-text (also called automatic speech recognition, or ASR) APIs convert spoken audio into written text. Your application sends an audio file or a live audio stream to the API, and the service returns a transcript.
Modern STT APIs process audio through several stages. First, the audio is preprocessed to reduce noise and normalize levels. Then, the acoustic model identifies phonemes (individual speech sounds) in the audio. A language model predicts the most likely sequence of words from those phonemes. Finally, the output is formatted with punctuation, capitalization, and timestamps.
Today's STT APIs go well beyond raw transcription. Features like speaker diarization (identifying who said what), word-level timestamps, automatic language detection, sentiment analysis, and PII redaction turn a simple transcript into structured, actionable data. For applications like contact center analytics or meeting intelligence, these features matter as much as transcription accuracy itself.
Speech-to-text (STT) converts audio into text. Text-to-speech (TTS) converts text into audio. The two are completely different technologies with different models, different APIs, and different use cases. CAMB.AI offers both: Speech-to-Text for transcription and captioning, and the MARS8 model family for high-quality voice generation. Confusing them leads to selecting the wrong tool entirely.
Lab benchmarks and real-world performance are very different things. An API that transcribes clean studio audio at near-perfect accuracy might struggle with real-world recordings.
Call center recordings have hold music and keyboard clatter. Field interviews have wind and traffic. Meeting recordings have HVAC hum and multiple people talking at once. Real-world audio is messy, and your STT API needs to handle that mess without degrading into gibberish. Look for APIs that include noise suppression and voice activity detection as standard features.
A model trained primarily on American English broadcast speech will underperform on Scottish English, Indian English, or African American Vernacular English. Accent coverage is a major differentiator between STT providers. Ask vendors about their training data diversity, and test with audio that matches your actual user base.
Medical dictation, legal proceedings, and financial earnings calls all contain specialized vocabulary that general-purpose models may misrecognize. Custom vocabulary features (also called phrase boosting or keyword hints) let you tell the API to expect specific terms, improving accuracy for domain-specific content. Some providers also offer pre-built domain models for healthcare, legal, and finance.
The timing of transcription matters as much as the accuracy. Different use cases demand different processing modes.
Real-time STT uses WebSocket connections to transcribe audio as it is spoken. The API returns partial results immediately and refines them into final transcripts as more context becomes available. Live captioning, voice agents, and real-time meeting notes all require streaming transcription with sub-second latency.
Podcast episodes, recorded interviews, and archived call recordings do not need real-time processing. Batch transcription processes complete audio files, often with higher accuracy than streaming because the model can consider the full context before finalizing the transcript. Batch is also typically cheaper than real-time.
If users are waiting for the output in real time, you need streaming. If the audio is already recorded and the transcript can arrive minutes later, batch mode gives you better accuracy at lower cost. Many applications use both: streaming for live interactions and batch for processing recorded content overnight.
Global applications need STT that works across languages, and multilingual transcription brings its own set of challenges.
A provider claiming "100+ languages" may have excellent accuracy for the top 10 and mediocre performance for the rest. Always test accuracy in your specific target languages rather than relying on marketing claims. CAMB.AI's Speech-to-Text supports multi-language transcription for organizations processing audio across multiple markets.
Real conversations often mix languages. A bilingual customer support call might shift between English and Spanish mid-sentence. STT APIs handle code-switching with varying degrees of success. If your audio contains mixed-language speech, test this scenario specifically during evaluation.
Mandarin spoken in Beijing sounds different from Mandarin spoken in Singapore. Spanish from Mexico City differs from Buenos Aires. Language coverage is only meaningful if the models also handle regional accent variation. Leading providers now train on geographically diverse speech data. For voice generation in those same languages, CAMB.AI's MARS8 covers languages representing 99% of the world's speaking population.
With dozens of options available, narrowing the field requires a structured evaluation approach.
Every provider claims high accuracy. The only number that matters is accuracy on your specific audio. Request trial access and test with recordings that represent your actual use case: the same audio quality, the same languages, the same domain vocabulary. Word Error Rate (WER) measured on your data is the single most important evaluation metric.
Voice agents need sub-300ms latency. Live captioning needs sub-500ms. Batch transcription can tolerate minutes. Define your latency requirement before evaluating providers, and test under realistic load conditions rather than on idle demo servers.
Speaker diarization, timestamps, language detection, PII redaction, topic detection, and sentiment analysis may all be relevant depending on your application. Some providers include these features in the base price; others charge separately. Map your feature requirements before comparing pricing.
Regulated industries (healthcare, finance, government) may require on-premise or VPC deployment. Data residency requirements may restrict which cloud regions your audio can be processed in. Security certifications (SOC 2, HIPAA, GDPR compliance) matter for enterprise adoption. CAMB.AI holds SOC 2 Type II certification, providing the security assurance that enterprise customers require.
STT pricing varies: per-minute, per-second, per-hour, or tiered plans. Compare pricing at your projected volume, not at the free tier. Factor in premium features, overage charges, and infrastructure overhead. A provider cheapest at 100 hours may not be cheapest at 10,000 hours.
For teams evaluating CAMB.AI's STT capabilities, the process is straightforward:
The right STT API is the one that delivers consistent accuracy on your actual audio, at your required latency, within your budget. Test before you commit, and re-evaluate as your requirements evolve.
Ya seas un profesional de los medios de comunicación o un desarrollador de productos de IA de voz, este boletín es tu guía de referencia sobre todo lo relacionado con la tecnología de voz y localización.


