- Speech-to-Text
Whisper 1 is OpenAI’s hosted automatic speech recognition model based on the open-source Whisper family, designed for high-quality transcription and translation of audio. It is notable…
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Whisper Large V3 Turbo is OpenAI’s optimized, high‑speed variant of the Whisper Large V3 automatic speech recognition model, designed to provide fast transcriptions while preserving strong accuracy across many languages.
Output tokens per second · Higher is better
Seconds · Lower is better
USD per 1M tokens (blended) · Lower is better
About the model
Whisper Large V3 Turbo is a large-scale multilingual automatic speech recognition model from OpenAI optimized for low-latency, high-throughput transcription workloads. It is mainly used to convert spoken audio into text for applications like live captioning, call and meeting transcription, and voice-driven interfaces. It is also deployed in batch transcription pipelines for large audio archives and media processing, where its speed and cost efficiency are important. It belongs to the Whisper family of speech recognition models and is a turbo-optimized successor to earlier Whisper Large variants such as Large V2 and Large V3.
Model capabilities
Converts spoken audio to text across many languages, handling varied accents, recording conditions, and conversational or long-form content.
Provides fast, streaming speech-to-text suitable for live applications, meetings, captions, and interactive voice-driven user experiences.
Transcribes and translates speech between multiple languages in a single step, enabling cross-lingual communication from audio sources.
Maintains strong transcription accuracy even with imperfect microphones, background noise, overlapping speech, or challenging acoustic environments.
Extracts textual content from audio sources like lectures, podcasts, or voice notes, enabling search, summarization, and downstream language processing.
Use cases
Transparent pricing
Save up to 60% on Whisper Large V3 Turbo-compatible transcription versus major cloud APIs.
| Provider | Region | Latency | Throughput | Uptime | Input ($/1M) | Output ($/1M) | Context |
|---|---|---|---|---|---|---|---|
| LLM API BEST | Global | ~220ms | ~220 min/s | 99.99% | $0.004/min | $0.004/min | ~6 hours audio |
| OpenAI | Global | ~350ms | ~160 min/s | 99.9% | ~$0.006/min | ~$0.006/min | ~3 hours audio |
| Azure OpenAI | US East / EU West | ~380ms | ~140 min/s | 99.9% | ~$0.007/min | ~$0.007/min | ~3 hours audio |
| Google Cloud Speech-to-Text (latest model) | Global | ~400ms | ~120 min/s | 99.9% | ~$0.008/min | ~$0.008/min | ~4 hours audio |
| Amazon Transcribe (highest accuracy tier) | US East / EU | ~420ms | ~110 min/s | 99.9% | ~$0.009/min | ~$0.009/min | ~4 hours audio |
Performance benchmarks
| Metric | Whisper Large V3 Turbo (OpenAI) | Whisper Large V3 (OpenAI) | Nova-2 (Deepgram) |
|---|---|---|---|
| Avg Latency | ~200ms | ~300ms | ~250ms |
| Languages Supported | ~100 | ~100 | ~30 |
| Price per Minute | $0.006 | $0.006 | $0.013 |
| Max Duration per Request | 2h | 2h | 4h |
| Accuracy (WER) | ~6% | ~7% | ~8% |
| Streaming Support | Yes | Yes | Yes |
| Uptime | 99.9% | 99.9% | 99.9% |
30-day usage via LLM API
Architecture & Integration
One unified API. Every major model. Built-in reliability, cost control, and observability.
Automatically route each request to the optimal model across providers based on latency, price, or quality—without changing your app code or wiring custom logic.
One endpoint, every modelControl spend with configurable pricing policies, dynamic model selection, and real-time cost insights so you can experiment freely without surprise bills or manual tracking.
Optimize quality per dollarDefine automatic failover chains across providers so timeouts, rate limits, or outages transparently fall back to alternatives—keeping your production workloads reliably online.
No single point of failureTrace every request across models with logs, metrics, and structured events so you can debug prompts, compare providers, and prove reliability to stakeholders.
See every token, everywhereCall high-level tasks like chat, tools, embeddings, and RAG through one consistent interface, while LLM.API handles provider quirks, parameters, and model-specific features.
Think tasks, not providersSend large batches of prompts, embeddings, or tool calls in a single request to maximize throughput, cut network overhead, and simplify large-scale processing pipelines.
Ship millions of calls fastDecision guide
FAQ
Whisper Large V3 Turbo is OpenAI’s high-throughput speech recognition model optimized for fast, accurate transcription and translation of audio.
Whisper Large V3 Turbo takes audio as input and outputs text transcriptions or translations.
LLM.API exposes Whisper Large V3 Turbo using its own metered pricing; check your LLM.API billing or pricing docs for current per-minute rates.
Whisper-style models process long-form audio by chunking and can handle multi-hour recordings, but exact limits depend on LLM.API’s request size constraints.
Latency depends on audio length and load, but V3 Turbo is designed for near real-time or faster-than-real-time transcription on typical server hardware.
Use the LLM.API endpoint with the model identifier for OpenAI Whisper Large V3 Turbo and send your audio file or stream plus configuration parameters.
Whisper Large V3 Turbo generally provides higher throughput and better cost-performance while maintaining or improving accuracy versus earlier Whisper Large models.
Yes, Whisper Large V3 Turbo can transcribe speech and optionally translate it into a target language, configured via API parameters.
Whisper Large V3 Turbo supports many widely used languages for transcription and translation, but quality varies by language and accent.
It can struggle with very noisy audio, highly domain-specific jargon, overlapping speakers, or low-resource languages and may produce occasional hallucinated words.
Streaming support depends on LLM.API’s interface; if exposed, you can send incremental audio chunks and receive partial transcripts.
Whisper Large V3 Turbo outputs text only and does not natively perform speaker diarization or identification; you must use external tools for that.
Compare
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