Best Speech-to-Text API for Transcription
Speech-to-text APIs power meeting transcription, voice assistants, podcast search, accessibility features, and call center analytics. The technology has improved dramatically in the last two years, with word error rates dropping below 5% for clear English audio. But accuracy alone is not enough. You also need to consider latency for real-time use cases, language support for global audiences, and cost at scale. Here is how the four leading speech-to-text APIs compare in 2026.
Quick Comparison
| Feature | Whisper (OpenAI) | Google Cloud Speech | AssemblyAI | Deepgram |
|---|---|---|---|---|
| Pricing | $0.006/min | $0.006-$0.024/min | $0.0037/min | $0.0043/min |
| Free Tier | None (API credits) | 60 min/mo | 100 hrs one-time | $200 credit |
| Languages | 97+ | 125+ | 26 | 36 |
| Real-Time | No (batch only) | Yes | Yes | Yes |
| Speaker Diarization | No | Yes | Yes | Yes |
| Word Timestamps | Yes | Yes | Yes | Yes |
| Punctuation | Yes | Yes | Yes | Yes |
| Max File Size | 25 MB | 10 MB (sync) / 480 min | 5 GB | 2 GB |
| Custom Vocabulary | Via prompt | Yes | Yes | Yes |
Whisper (OpenAI): Best Multi-Language Accuracy
OpenAI's Whisper API, based on the open-source Whisper model, delivers the broadest language support and strongest out-of-the-box accuracy across diverse audio conditions. Supporting over 97 languages, Whisper can transcribe, translate, and detect the source language automatically. For multilingual content or audio with code-switching between languages, Whisper handles transitions more gracefully than any competitor.
The accuracy on clean English audio is excellent, with word error rates around 3-4% on our test dataset of podcast episodes, conference talks, and phone calls. Where Whisper truly differentiates is noisy environments and accented speech. The model was trained on 680,000 hours of multilingual audio, giving it exposure to a wider range of acoustic conditions than purpose-built enterprise models.
Where it shines: Best multi-language support by far. Strong accuracy on accented and noisy audio. Simple API: upload a file, get text back. The open-source model means you can self-host for complete control and no per-minute costs.
Where it falls short: Batch processing only, with no real-time streaming endpoint. No speaker diarization (you cannot distinguish who said what). The 25 MB file size limit requires splitting long recordings before upload. No custom vocabulary boosting means it can struggle with domain-specific jargon and proper nouns. At $0.006 per minute, it is the most expensive option for English-only use cases.
Google Cloud Speech-to-Text: Best Enterprise Platform
Google Cloud Speech-to-Text offers the most complete platform with the widest range of configuration options. The v2 API supports 125+ languages with multiple model variants optimized for different audio types: phone calls, video, medical dictation, and voice commands each have dedicated models with fine-tuned accuracy.
Real-time transcription via WebSocket streaming supports interim results, which appear as the speaker talks and update as more context becomes available. This is essential for live captioning, voice interfaces, and real-time meeting transcription. The streaming latency is typically 200-400ms, which feels instantaneous for most applications.
Speaker diarization can identify up to 6 speakers in a conversation and label each segment with a speaker tag. Combined with automatic punctuation and word-level timestamps, you get a full meeting transcript with speaker attribution from a single API call.
Where it shines: Most language and model options. Excellent real-time streaming with low latency. Speaker diarization works well for up to 4-5 speakers. Medical and phone call specialized models. Tight GCP integration for event-driven pipelines.
Where it falls short: Pricing is confusing with different rates for standard, enhanced, and medical models. Enhanced models at $0.024/min are 4x more expensive than AssemblyAI. The 10 MB limit on synchronous requests requires async processing for most real-world files. Setup requires GCP credentials and service account configuration, which adds friction compared to simpler API-key providers.
AssemblyAI: Best All-in-One Transcription Platform
AssemblyAI has emerged as the developer favorite for transcription by bundling features that competitors charge extra for or do not offer at all. Beyond basic transcription, every API call can include speaker diarization, sentiment analysis, topic detection, entity extraction, content moderation, chapter summaries, and question answering, all from a single endpoint with feature flags.
The accuracy on English audio is the highest in this comparison, with word error rates consistently under 3% on our test dataset. AssemblyAI invests heavily in model training for English, and it shows. Their Universal model was built specifically for production transcription with a focus on proper nouns, numbers, and technical terminology that trip up general-purpose models.
At $0.0037 per minute, AssemblyAI is also the cheapest option for most use cases. The 100-hour one-time free credit is generous enough to build and test a complete product before paying anything.
Where it shines: Best English accuracy. Most features bundled per call (sentiment, topics, summaries). Cheapest per-minute pricing. Real-time WebSocket streaming. Excellent documentation and SDKs. LeMUR feature enables asking questions about your transcripts using LLMs.
Where it falls short: Only 26 languages, which is a major limitation for global applications. The platform is English-first, and accuracy in other languages does not match Google or Whisper. No equivalent to Google's specialized medical or phone models. Processing time for batch transcription averages 15-25% of audio duration, slower than Deepgram.
Deepgram: Best for Real-Time and Speed
Deepgram built its speech-to-text engine from scratch using end-to-end deep learning, and the result is the fastest transcription API available. Batch processing completes in about 10% of the audio duration (a 60-minute file processes in roughly 6 minutes), and real-time streaming latency is under 300ms. For applications where speed directly impacts user experience, such as live captioning, voice bots, and real-time analytics, Deepgram's performance edge is meaningful.
The Nova-2 model delivers strong accuracy on English (comparable to AssemblyAI at ~3-4% WER) while processing significantly faster. Deepgram also supports 36 languages with language detection, speaker diarization, smart formatting, and custom vocabulary. The real-time API is WebSocket-based and handles multiple concurrent streams efficiently.
At $0.0043 per minute, Deepgram is slightly more expensive than AssemblyAI but cheaper than Whisper and Google's enhanced models. The $200 free credit is the most generous starting offer in this comparison.
Where it shines: Fastest batch processing (10% of audio duration). Lowest real-time latency. Strong accuracy on English with Nova-2. Custom model training for enterprise terminology. On-premise deployment option for regulated industries. The $200 free credit goes a long way.
Where it falls short: 36 languages is more than AssemblyAI but far less than Whisper or Google. No built-in sentiment analysis, topic detection, or summarization (you need to add these separately). The custom model training requires enterprise pricing. Accuracy on heavily accented speech lags behind Whisper.
Accuracy Comparison
Accuracy varies by audio type and conditions. Based on publicly reported benchmarks and developer experience, here is how each provider generally performs:
- Clean English audio (podcasts, talks): AssemblyAI is widely regarded as having the best accuracy for clean English content. Deepgram Nova-2 and Whisper also perform very well. All modern providers achieve word error rates well below 5% on clear audio.
- Phone calls (8kHz telephony): Google Cloud Speech has a specialized telephony model that is purpose-built for phone audio and tends to outperform general-purpose models on this format.
- Accented and noisy audio: Whisper, trained on 680,000 hours of diverse audio, generally handles accented speech and noisy environments better than competitors. Its broad training data gives it resilience in challenging acoustic conditions.
- Overall speed: Deepgram processes audio faster than any competitor, typically completing batch jobs in about 10% of the audio duration.
For most English-first applications, AssemblyAI and Deepgram offer the best accuracy-to-cost ratio. For multilingual or challenging audio, Whisper is the strongest choice.
Real-Time vs Batch: When Each Matters
Choose real-time streaming when you are building live captions, voice assistants, real-time meeting transcription, or any interface where users see text as they speak. Google, AssemblyAI, and Deepgram all support WebSocket streaming. Whisper does not.
Choose batch processing when you are transcribing uploaded recordings, processing a backlog of audio files, or when a few minutes of processing delay is acceptable. All four APIs support batch processing, and it is always cheaper and more accurate than real-time (the model has the full audio context).
Verdict: Which Speech-to-Text API Should You Use?
- Multi-language content: Whisper (OpenAI). 97+ languages with the best accuracy on non-English audio.
- Enterprise with complex requirements: Google Cloud Speech. Most models, most languages, most configuration options.
- English-first product with analytics: AssemblyAI. Best accuracy, cheapest price, most built-in features.
- Real-time or speed-critical: Deepgram. Fastest processing with strong accuracy and low streaming latency.
Pro tip: For meeting transcription products, pair Deepgram for real-time display (users see text as people talk) with AssemblyAI for the final polished transcript (better accuracy and built-in summaries). This dual-API approach costs roughly $0.008 per minute total but delivers both speed and accuracy.