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    Best Speech-to-Text APIs in 2026

    We tested the top speech-to-text APIs on transcription accuracy, real-time latency, and language coverage. This guide covers cloud services, open-source models, and specialized solutions for different audio environments.

    Last tested: June 20, 2026
    11 tools evaluated

    Transcription is one stage. Point Mixpeek at your audio and it chains ASR, speaker diarization, and forced alignment, then embeds and indexes the result so an agent can search by what was said, who said it, and exactly when. First 1M vectors free.

    Search your audio, not just transcribe it

    Quick Answer

    The best overall option in this category is Deepgram, especially for applications needing the fastest, most accurate english transcription at competitive prices. The rankings below compare each tool by strengths, limitations, pricing, and fit for production use.

    Skip the comparison? Mixpeek runs speech-to-text on your own data: extraction, indexing, and search in one platform.

    How We Evaluated

    Accuracy

    30%

    Word error rate across clean speech, noisy environments, accented speakers, and domain-specific terminology.

    Real-Time Performance

    25%

    End-to-end latency for streaming transcription and responsiveness to speech-to-text conversion.

    Language Support

    25%

    Number of supported languages, dialect handling, and accuracy on non-English content.

    Advanced Features

    20%

    Speaker diarization, punctuation, PII redaction, and custom vocabulary support.

    Overview

    The speech-to-text market shifted hard toward accuracy in 2025 and 2026. ElevenLabs Scribe now tops independent English benchmarks (roughly 3-4% WER), with Deepgram Nova-3 (about 5.3% WER) close behind and far cheaper per minute, and OpenAI's gpt-4o-transcribe replacing the older Whisper API for higher-accuracy hosted use. Self-hosted Whisper large-v3 is still the go-to for free, private, multilingual transcription across 99 plus languages. Google and AWS continue to win on language breadth and enterprise compliance rather than raw accuracy. AssemblyAI has gotten aggressive on price (its Universal tier is now around $0.15 per hour) while keeping the richest audio intelligence suite. For teams already on a major cloud, the native offering (Google Speech-to-Text, AWS Transcribe, Azure Speech) often wins on latency and integration even if accuracy trails the specialists, and Rev AI, Speechmatics, and Gladia remain strong mid-market picks. If your real goal is searching across audio (and video) rather than just producing a transcript, a multimodal platform like Mixpeek runs transcription as one stage of a larger indexing and retrieval pipeline rather than leaving you to wire the search layer yourself.
    1

    Deepgram

    AI speech recognition platform whose Nova-3 model delivers around 5.3% WER on clean English at a fraction of the price of the accuracy leaders. ElevenLabs Scribe and gpt-4o-transcribe now edge it on raw accuracy, but Deepgram remains the best value for high-volume English work. Offers real-time streaming with sub-250ms latency and batch transcription at 100x real-time speed. Smart formatting adds punctuation, capitalization, and numerals automatically.

    What Sets It Apart

    Near-top English accuracy (about 5.3% WER) at commodity pricing plus sub-250ms streaming latency, which makes it the best value for high-volume real-time and batch English transcription.

    Strengths

    • +Nova-3 at about 5.3% WER on clean English at commodity per-minute pricing
    • +Real-time streaming under 250ms latency for live captioning
    • +Batch processing at 100x real-time for large audio archives
    • +Smart formatting, speaker diarization, and topic detection built in

    Limitations

    • -36 languages — fewer than Google (125+) or Whisper (99+)
    • -Custom vocabulary and model fine-tuning require Growth plan
    • -On-premises deployment requires enterprise agreement
    • -Non-English accuracy gap vs. Whisper for low-resource languages

    Real-World Use Cases

    • Live captioning for webinars and virtual events with sub-250ms latency
    • Transcribing podcast back-catalogs at 100x real-time for full-text search
    • Real-time voice agent pipelines where low latency is critical for conversational flow
    • Call center post-call analytics with speaker diarization and topic detection

    Choose This When

    When you want strong English accuracy and low latency at the lowest sustainable cost, especially for real-time voice applications or large-scale batch processing.

    Skip This If

    When you need broad multilingual coverage (36 languages vs. 99+ for Whisper) or require on-premise deployment without an enterprise contract.

    Integration Example

    const { createClient } = require("@deepgram/sdk");
    const deepgram = createClient(process.env.DEEPGRAM_API_KEY);
    
    const { result } = await deepgram.listen.prerecorded.transcribeUrl(
      { url: "https://example.com/audio.mp3" },
      { model: "nova-3", smart_format: true, diarize: true }
    );
    console.log(result.results.channels[0].alternatives[0].transcript);
    Pay-as-you-go from $0.0043/min ($0.26/hr); Growth plan from $0.0036/min
    Best for: Applications needing the fastest, most accurate English transcription at competitive prices
    Visit Website
    2

    ElevenLabs Scribe

    Speech-to-text model that topped independent English accuracy benchmarks on release (about 96.7% accuracy, roughly 3-4% WER) ahead of Deepgram Nova-3, Whisper v3, and Gemini. Scribe v2 covers 99 languages, adds speaker diarization for up to 32 speakers at about 98% label accuracy, word-level timestamps, and audio event tagging, and ships a real-time streaming API at around 150ms latency.

    What Sets It Apart

    Benchmark-leading English accuracy combined with 99-language coverage and high-accuracy diarization in one model, so you do not trade multilingual breadth for accuracy.

    Strengths

    • +Top-tier English accuracy (about 3-4% WER), among the best measured anywhere
    • +99 languages with strong non-English accuracy, not just an English specialist
    • +High-quality diarization up to 32 speakers with about 98% label accuracy
    • +Real-time streaming API at roughly 150ms latency plus word-level timestamps

    Limitations

    • -Newer to STT than Deepgram or Google, with a shorter production track record
    • -No self-hosted option, so audio must go to the ElevenLabs cloud
    • -Per-minute pricing higher than Deepgram for high-volume English-only work
    • -Best known for voice synthesis, so STT-specific enterprise tooling is still maturing

    Real-World Use Cases

    • High-stakes transcription (legal, medical, media) where accuracy outweighs per-minute cost
    • Multilingual transcription that needs both strong non-English accuracy and reliable diarization
    • Call recordings that need speaker labels (agent vs customer) at high accuracy
    • Subtitling and captioning workflows that depend on precise word-level timestamps

    Choose This When

    When transcription accuracy is the deciding factor, especially for multilingual or multi-speaker audio where you also need reliable diarization.

    Skip This If

    When you need self-hosted deployment, the lowest possible cost for English-only high-volume work (Deepgram wins), or a long enterprise compliance track record.

    Integration Example

    from elevenlabs.client import ElevenLabs
    
    client = ElevenLabs(api_key="YOUR_API_KEY")
    with open("interview.mp3", "rb") as audio:
        result = client.speech_to_text.convert(
            model_id="scribe_v2",
            file=audio,
            diarize=True,
            timestamps_granularity="word",
        )
    print(result.text)
    From $0.40/hr of input audio (about $0.0067/min); Scribe v2 lowered pricing roughly 40%
    Best for: Teams that want the highest transcription accuracy across many languages with built-in diarization
    Visit Website
    3

    OpenAI Whisper

    Open-source speech recognition model trained on 680,000 hours of multilingual audio. Whisper large-v3 achieves 10-15% WER across 99+ languages, making it the best free self-hostable multilingual model available. Fully self-hostable under MIT license. Note that for hosted use OpenAI now steers new projects to gpt-4o-transcribe (about 4.1% WER, $0.006/min) and gpt-4o-mini-transcribe ($0.003/min) rather than the legacy Whisper API endpoint.

    What Sets It Apart

    Unmatched multilingual breadth (99+ languages) with MIT-licensed self-hosting — the only model you can run entirely on your own infrastructure at no per-minute cost.

    Strengths

    • +99+ languages with strong non-English accuracy (10-15% WER)
    • +Free and open source (MIT license) for self-hosting
    • +Exceptionally robust against background noise and accents
    • +Large ecosystem — faster-whisper, whisper.cpp, WhisperX for diarization

    Limitations

    • -Self-hosted requires GPU (large-v3 needs ~10GB VRAM)
    • -No native real-time streaming — batch only unless using distilled variants
    • -No built-in speaker diarization (requires WhisperX or pyannote add-on)
    • -API version is batch-only with no streaming endpoint

    Real-World Use Cases

    • Transcribing multilingual customer support calls across 99+ languages
    • Processing field recordings with heavy background noise (construction, outdoor events)
    • Building a self-hosted transcription pipeline to avoid sending audio to third-party APIs
    • Academic research requiring reproducible, free-to-use speech recognition

    Choose This When

    When you need strong multilingual support, want to self-host for privacy or cost reasons, or are processing noisy audio where Whisper's robustness shines.

    Skip This If

    When you need real-time streaming transcription or built-in speaker diarization without stitching together additional libraries.

    Integration Example

    import whisper
    
    model = whisper.load_model("large-v3")
    result = model.transcribe(
        "interview.mp3",
        language=None,          # auto-detect
        word_timestamps=True
    )
    for segment in result["segments"]:
        print(f"[{segment['start']:.1f}s] {segment['text']}")
    Free self-hosted; OpenAI API at $0.006/min ($0.36/hr)
    Best for: Multilingual transcription, noisy audio, and self-hosted deployments
    Visit Website
    4

    AssemblyAI

    Speech-to-text platform that goes beyond transcription into audio intelligence, offering speaker diarization, PII redaction, content safety, entity recognition, sentiment analysis, and auto-summarization in one API. Its Universal models cover 99 languages, and aggressive 2026 repricing dropped the base async rate to about $0.15/hr, making it one of the cheapest production options.

    What Sets It Apart

    The richest audio intelligence feature set in a single API — PII redaction, content safety, entity detection, sentiment, and LLM-powered summarization without integrating separate services.

    Strengths

    • +Audio intelligence suite: PII redaction, safety, entities, sentiment, summaries
    • +Excellent developer experience with strong docs and SDKs in the category
    • +Universal models cover 99 languages with competitive English accuracy
    • +LeMUR integration for LLM-powered audio Q&A and summarization

    Limitations

    • -Advanced features (diarization, PII, sentiment) are billed on top of the base rate
    • -Cloud-only with no self-hosted deployment option
    • -Highest-accuracy Universal-3 Pro tier covers far fewer languages
    • -Some features (safety, PII) add latency to the processing pipeline

    Real-World Use Cases

    • Healthcare call transcription with automatic PII redaction for HIPAA compliance
    • Podcast production workflows that auto-generate show notes, chapters, and summaries
    • Content moderation for user-generated audio on social platforms
    • Sales call analysis with entity extraction, sentiment tracking, and action item detection

    Choose This When

    When you need more than raw transcription, such as content moderation, entity extraction, summarization, or PII handling, and want it all from one vendor with strong docs.

    Skip This If

    When you need the absolute lowest English WER (ElevenLabs Scribe and gpt-4o-transcribe lead) or fully on-prem deployment, which AssemblyAI does not offer.

    Integration Example

    import assemblyai as aai
    
    aai.settings.api_key = "YOUR_API_KEY"
    transcriber = aai.Transcriber()
    
    config = aai.TranscriptionConfig(
        speaker_labels=True,
        auto_highlights=True,
        entity_detection=True,
        sentiment_analysis=True
    )
    transcript = transcriber.transcribe("meeting.mp3", config=config)
    for utterance in transcript.utterances:
        print(f"Speaker {utterance.speaker}: {utterance.text}")
    Async Universal from $0.0025/min ($0.15/hr); Universal-3 Pro from $0.0035/min; streaming and audio intelligence add-ons extra
    Best for: Developers wanting transcription plus content safety, entity detection, and summarization
    Visit Website
    5

    Google Cloud Speech-to-Text

    Google's speech recognition API with the widest language coverage at 125+ languages and dialects. Offers specialized models for medical dictation, phone calls, and short queries. V2 API with Chirp model available on-prem via Google Distributed Cloud.

    What Sets It Apart

    The widest language and dialect coverage (125+) combined with specialized domain models (medical, phone, short queries) that no competitor matches.

    Strengths

    • +125+ languages with dialect-level support — widest coverage available
    • +Specialized models: Medical Conversations, Phone Call, Short Queries
    • +Chirp model available on-device and on-prem
    • +Multi-channel recognition for call center stereo audio

    Limitations

    • -Standard model WER (~12-15%) behind Deepgram Nova and Whisper
    • -Complex pricing across 3+ model tiers (standard, enhanced, chirp)
    • -GCP lock-in for best integration and lowest latency
    • -Speaker diarization less accurate than AssemblyAI for multi-speaker

    Real-World Use Cases

    • Global SaaS products serving users in 100+ countries with dialect-specific recognition
    • Medical dictation systems requiring HIPAA-compliant specialized models
    • IVR and voice assistant systems using the Short Queries model for command recognition
    • Multi-channel call center recordings with per-channel speaker separation

    Choose This When

    When your application serves a global user base with rare languages, needs medical-grade transcription, or is already deeply integrated with GCP.

    Skip This If

    When English-only accuracy matters most (Deepgram and Whisper beat it) or when you want simple, flat-rate pricing.

    Integration Example

    from google.cloud import speech_v2
    
    client = speech_v2.SpeechClient()
    config = speech_v2.RecognitionConfig(
        auto_decoding_config=speech_v2.AutoDetectDecodingConfig(),
        language_codes=["en-US"],
        model="chirp_2",
        features=speech_v2.RecognitionFeatures(
            enable_automatic_punctuation=True
        ),
    )
    with open("audio.wav", "rb") as f:
        response = client.recognize(
            config=config, content=f.read(),
            recognizer="projects/my-project/locations/global/recognizers/_"
        )
    print(response.results[0].alternatives[0].transcript)
    Standard from $0.024/min; Enhanced $0.036/min; Chirp $0.048/min; Medical $0.078/min
    Best for: Global apps needing 125+ languages, medical transcription, or GCP integration
    Visit Website
    6

    AWS Transcribe

    Amazon's speech-to-text service supporting 100+ languages with automatic language identification. Includes Transcribe Medical for HIPAA-eligible clinical dictation and Contact Lens integration for call center analytics with sentiment and issue detection.

    What Sets It Apart

    Deepest AWS ecosystem integration (S3 triggers, Lambda, Contact Lens, Comprehend) making it the path of least resistance for AWS-native architectures.

    Strengths

    • +100+ languages with automatic language identification
    • +Transcribe Medical for HIPAA-eligible clinical dictation
    • +Contact Lens integration for call center analytics
    • +Custom vocabulary and custom language models for domain terms

    Limitations

    • -Base accuracy behind Deepgram and Whisper in benchmarks
    • -Per-second pricing expensive for long audio files
    • -Deep AWS dependency — hard to migrate away
    • -Custom language model training requires substantial data

    Real-World Use Cases

    • S3-triggered transcription pipelines for media assets uploaded to AWS
    • HIPAA-compliant clinical note dictation in healthcare EHR integrations
    • Amazon Connect call center analytics with real-time sentiment and issue detection
    • Subtitle generation for video-on-demand platforms hosted on AWS infrastructure

    Choose This When

    When your infrastructure is on AWS and you want seamless integration with S3, Lambda, and Contact Lens without managing cross-cloud networking.

    Skip This If

    When raw transcription accuracy is paramount (Deepgram and Whisper outperform) or when you need a cloud-agnostic solution.

    Integration Example

    import boto3
    
    transcribe = boto3.client("transcribe")
    transcribe.start_transcription_job(
        TranscriptionJobName="my-job",
        Media={"MediaFileUri": "s3://my-bucket/audio.mp3"},
        MediaFormat="mp3",
        LanguageCode="en-US",
        Settings={
            "ShowSpeakerLabels": True,
            "MaxSpeakerLabels": 5,
            "ShowAlternatives": True,
            "MaxAlternatives": 3,
        },
        OutputBucketName="my-output-bucket",
    )
    Standard from $0.024/min; Medical $0.075/min; volume discounts available
    Best for: AWS-native teams needing medical transcription or call center analytics
    Visit Website
    7

    Rev AI

    Speech-to-text API from Rev, the human transcription company that used millions of hours of human-corrected transcripts to train its ASR models. Offers both streaming and async transcription with strong accuracy on conversational speech, accented English, and multi-speaker audio.

    What Sets It Apart

    Models trained on human-corrected transcripts give Rev AI an edge on conversational speech, accented English, and multi-speaker scenarios that trip up competitors.

    Strengths

    • +Trained on millions of hours of human-corrected transcripts for high accuracy
    • +Strong performance on accented English and conversational speech
    • +Both streaming and async endpoints with simple REST API
    • +Custom vocabulary support for domain-specific terminology

    Limitations

    • -English-primary — multilingual support limited to ~15 languages
    • -Higher latency on streaming compared to Deepgram
    • -No built-in audio intelligence features (PII, sentiment)
    • -Smaller ecosystem and community compared to Whisper or Deepgram

    Real-World Use Cases

    • Legal deposition transcription where accuracy on conversational speech is critical
    • Transcribing interviews and focus groups with multiple accented speakers
    • Media production workflows requiring broadcast-quality transcripts
    • Accessibility compliance for video content with diverse speaker profiles

    Choose This When

    When you are transcribing conversational content (interviews, depositions, meetings) where speaker diversity and accent handling matter more than per-minute cost.

    Skip This If

    When you need multilingual support beyond English or want built-in audio intelligence features like PII redaction.

    Integration Example

    from rev_ai import apiclient
    
    client = apiclient.RevAiAPIClient("YOUR_ACCESS_TOKEN")
    job = client.submit_job_url(
        media_url="https://example.com/interview.mp3",
        metadata="legal-deposition-2026",
        skip_diarization=False,
        custom_vocabularies=[{
            "phrases": ["habeas corpus", "amicus curiae"]
        }]
    )
    # Poll for completion
    transcript = client.get_transcript_text(job.id)
    print(transcript)
    Async from $0.02/min; streaming from $0.035/min; volume discounts available
    Best for: Applications processing conversational speech with accented speakers where accuracy matters more than cost
    Visit Website
    8

    Speechmatics

    UK-based speech recognition provider with strong global language support (50+ languages) and an emphasis on accuracy across accents and dialects. Their Ursa model delivers competitive WER while offering on-prem deployment for data-sensitive industries.

    What Sets It Apart

    Best-in-class accuracy on non-US English accents and European languages, with on-prem deployment available to mid-market customers — not gated behind enterprise-only agreements.

    Strengths

    • +50+ languages with strong accent and dialect handling
    • +On-premises deployment available without enterprise-only gates
    • +Competitive accuracy on non-US English accents (UK, Australian, Indian)
    • +Real-time and batch processing with consistent API

    Limitations

    • -Less name recognition in the US market compared to Deepgram or AssemblyAI
    • -Smaller ecosystem and fewer community integrations
    • -Higher pricing than Deepgram for comparable English accuracy
    • -Limited audio intelligence features beyond transcription

    Real-World Use Cases

    • UK government and public sector transcription with on-premises data residency requirements
    • European multilingual customer support with strong accent handling across EU languages
    • Broadcasting and media companies needing accurate subtitles for diverse English accents
    • Financial services compliance recording where data cannot leave the corporate network

    Choose This When

    When your users speak with diverse accents (UK, Indian, Australian English) or European languages, and you may need on-prem deployment for data sovereignty.

    Skip This If

    When you need the absolute lowest English WER (Deepgram wins) or extensive audio intelligence features (AssemblyAI wins).

    Integration Example

    import speechmatics
    
    sm_client = speechmatics.client.WebsocketClient(
        speechmatics.models.ConnectionSettings(
            url="wss://eu2.rt.speechmatics.com/v2",
            auth_token="YOUR_API_KEY",
        )
    )
    conf = speechmatics.models.TranscriptionConfig(
        language="en",
        enable_partials=True,
        operating_point="enhanced",
    )
    sm_client.add_event_handler(
        event_name=speechmatics.models.ServerMessageType.AddTranscript,
        event_handler=lambda msg: print(msg["metadata"]["transcript"]),
    )
    await sm_client.run(audio_stream, conf)
    Pay-as-you-go from $0.017/min; on-prem licensing available
    Best for: Global enterprises needing accurate transcription across English accents and European languages with on-prem options
    Visit Website
    9

    Azure Speech Service

    Microsoft's speech recognition API within Azure Cognitive Services, supporting 100+ languages with custom speech models, real-time and batch transcription, and tight integration with Microsoft 365 and Teams. Offers on-prem deployment via Azure containers.

    What Sets It Apart

    The only speech API with native Microsoft 365 and Teams integration, plus container-based on-prem deployment — ideal for enterprises already invested in the Microsoft ecosystem.

    Strengths

    • +100+ languages with custom speech model training
    • +On-prem deployment via Docker containers
    • +Deep integration with Microsoft 365, Teams, and Dynamics
    • +Pronunciation assessment and custom voice features

    Limitations

    • -Base accuracy behind Deepgram and Whisper on English benchmarks
    • -Complex pricing tiers (standard, custom, real-time, batch)
    • -Azure ecosystem lock-in for optimal performance
    • -Custom model training UI less intuitive than competitors

    Real-World Use Cases

    • Transcribing Microsoft Teams meetings with speaker identification for enterprise compliance
    • Building voice-enabled Dynamics 365 workflows for CRM and customer service
    • On-premises speech processing in air-gapped environments using Azure containers
    • Custom speech models for industry-specific vocabulary in manufacturing or finance

    Choose This When

    When your organization runs on Microsoft infrastructure and needs speech recognition that plugs directly into Teams, Dynamics, and Azure services.

    Skip This If

    When you need best-in-class accuracy on English (Deepgram leads) or want a vendor-neutral solution not tied to a cloud provider.

    Integration Example

    import azure.cognitiveservices.speech as speechsdk
    
    speech_config = speechsdk.SpeechConfig(
        subscription="YOUR_KEY",
        region="eastus"
    )
    speech_config.speech_recognition_language = "en-US"
    
    audio_config = speechsdk.audio.AudioConfig(filename="meeting.wav")
    recognizer = speechsdk.SpeechRecognizer(
        speech_config=speech_config,
        audio_config=audio_config
    )
    result = recognizer.recognize_once()
    print(result.text)
    Standard from $0.016/min (batch); real-time from $0.016/min; custom models from $1.40/model/hr
    Best for: Microsoft-stack enterprises needing speech recognition integrated with Teams, Dynamics, and Azure infrastructure
    Visit Website
    10

    Gladia

    European speech-to-text API that wraps Whisper and proprietary models with production-grade features. Offers real-time streaming, speaker diarization, and audio intelligence in a single API, with GDPR-compliant EU data residency.

    What Sets It Apart

    GDPR-compliant EU data residency with code-switching detection — the best option for European companies that need multilingual transcription without sending data outside the EU.

    Strengths

    • +GDPR-compliant with EU data residency options
    • +Combines Whisper accuracy with production features (streaming, diarization)
    • +Code-switching detection for multilingual conversations
    • +Simple per-minute pricing with no feature add-on costs

    Limitations

    • -Newer entrant with less production track record than Deepgram or Google
    • -Accuracy depends on underlying Whisper models — not custom-trained
    • -Smaller language support than Google or AWS
    • -Limited enterprise features (SSO, audit logs) compared to incumbents

    Real-World Use Cases

    • EU-based SaaS products requiring GDPR-compliant speech processing with data residency
    • Multilingual meetings where speakers switch between languages mid-sentence
    • Startups wanting Whisper-level accuracy with managed streaming infrastructure
    • European media companies needing compliant transcription for broadcast content

    Choose This When

    When GDPR compliance and EU data residency are requirements, or when your audio contains multilingual conversations with code-switching.

    Skip This If

    When you need a battle-tested enterprise platform with extensive compliance certifications beyond GDPR.

    Integration Example

    const response = await fetch("https://api.gladia.io/v2/transcription", {
      method: "POST",
      headers: {
        "x-gladia-key": process.env.GLADIA_API_KEY,
        "Content-Type": "application/json",
      },
      body: JSON.stringify({
        audio_url: "https://example.com/audio.mp3",
        diarization: true,
        language_behaviour: "automatic multiple languages",
      }),
    });
    const { id } = await response.json();
    // Poll for result at /v2/transcription/{id}
    Free tier with 10 hrs/month; paid from $0.0061/min
    Best for: European companies needing GDPR-compliant transcription with production-grade streaming and diarization
    Visit Website
    11

    Picovoice Leopard

    On-device speech-to-text engine that runs entirely locally without any cloud dependency. Optimized for edge deployment on mobile, embedded, and IoT devices with small model sizes and low resource requirements.

    What Sets It Apart

    The only production-grade speech-to-text engine designed for fully offline, on-device operation — no internet, no cloud, no per-minute costs.

    Strengths

    • +Fully on-device — no internet connection required
    • +Low resource footprint suitable for mobile and embedded devices
    • +No per-minute API costs — one-time or annual licensing
    • +Complete data privacy — audio never leaves the device

    Limitations

    • -Lower accuracy than cloud models (higher WER than Deepgram or Whisper large)
    • -Limited language support compared to cloud alternatives
    • -No speaker diarization or advanced audio intelligence
    • -Smaller model means less robustness on noisy audio

    Real-World Use Cases

    • Offline voice control for IoT devices in warehouses or field environments without internet
    • Mobile apps that transcribe voice notes without sending audio to the cloud
    • Automotive in-car voice systems requiring real-time recognition without cellular connectivity
    • Healthcare point-of-care devices where patient audio cannot leave the device for privacy reasons

    Choose This When

    When your deployment environment has no reliable internet connectivity, when audio data cannot leave the device for privacy reasons, or when you want to eliminate per-minute API costs.

    Skip This If

    When you need cloud-grade accuracy, broad language support, or advanced features like diarization and audio intelligence.

    Integration Example

    import pvleopard
    
    leopard = pvleopard.create(
        access_key="YOUR_ACCESS_KEY",
        model_path="leopard_params.pv"
    )
    transcript, words = leopard.process_file("recording.wav")
    print(transcript)
    for word in words:
        print(f"{word.word} [{word.start_sec:.1f}s - {word.end_sec:.1f}s] "
              f"confidence: {word.confidence:.2f}")
    leopard.delete()
    Free for personal use; commercial from $5/device/year or enterprise licensing
    Best for: Edge and IoT applications that require offline speech recognition with no cloud dependency
    Visit Website
    Managed Mixpeek

    Put speech-to-text to work

    Connect a bucket and Mixpeek runs the whole speech-to-text pipeline for you: extraction, indexing, and search over your own objects. No models to wire up, nothing to host.

    Start with Managed
    MVS · bring your own

    Already have vectors?

    Keep your embeddings on your own cloud and run dense, sparse, and BM25 search directly on object storage. First 1M vectors free.

    Start with MVS

    Frequently Asked Questions

    How is a full audio search pipeline different from a speech-to-text API?

    A speech-to-text API returns a transcript. Making audio actually *searchable* takes three more stages these APIs leave to you: **speaker diarization** (who spoke each segment), **forced alignment** (word-level timestamps so a match jumps to the exact moment), and **embedding plus indexing** so an agent can retrieve by meaning, speaker, and time. Mixpeek runs that whole chain over your storage: transcription with models like Qwen3-ASR, speaker turns with pyannote 3.1, and word timing with Qwen3-ForcedAligner, so a query like 'when did the CEO mention pricing' returns the exact clip, the speaker, and the timestamp. The underlying concepts are in the speaker diarization and forced alignment guides.

    What is the best speech-to-text API for accuracy?

    As of mid-2026, ElevenLabs Scribe leads independent English benchmarks at roughly 3-4% WER, with OpenAI's gpt-4o-transcribe (about 4.1% WER) and Deepgram Nova-3 (about 5.3% WER) close behind. Deepgram remains the best value because it is far cheaper per minute. For free, private, multilingual transcription, self-hosted Whisper large-v3 is still the practical pick across 99 plus languages. The best choice depends on your language, accent, latency, and budget requirements.

    How much does speech-to-text cost per hour of audio?

    Pricing ranges from about $0.15/hour (AssemblyAI Universal) and $0.26/hour (Deepgram pay-as-you-go) up to $3.60/hour or more for premium cloud tiers like Google's enhanced models. ElevenLabs Scribe runs about $0.40/hour and gpt-4o-transcribe about $0.36/hour. Self-hosted Whisper is free aside from your own GPU costs. For high-volume workloads, committed-use contracts can cut costs further, and remember that diarization and PII redaction are often billed on top of the base rate.

    Can speech-to-text APIs handle noisy audio?

    Modern APIs handle moderate background noise well, but accuracy degrades significantly in very noisy environments. Whisper is particularly robust to noise. For best results, preprocess audio with noise reduction and choose models trained on noisy data. Expect 5-15% higher word error rates in noisy conditions versus clean speech.

    Do I need a separate platform if I want to search across my transcripts?

    A speech-to-text API gives you the transcript. If your goal is searching across many audio or video files (find the moment someone says X, or retrieve clips that are semantically about a topic), you still have to chunk, embed, index, and rank that text yourself. Multimodal platforms like Mixpeek run transcription as one stage of an indexing and retrieval pipeline over raw objects, so the search layer is handled rather than bolted on. If you only need transcripts for display or one-off processing, a standalone STT API is simpler and cheaper.

    See how Mixpeek handles this

    Purpose-built for speech-to-text apis — not bolted on.

    Audio Search

    Mixpeek's dedicated page for this capability — architecture, benchmarks, and how it works.

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