AssemblyAI API
APIFreeSpeech-to-text with intelligence — Universal-2, summarization, PII redaction, LeMUR for audio LLM.
Capabilities16 decomposed
universal-3 pro multilingual speech-to-text transcription with context-aware prompting
Medium confidenceConverts pre-recorded audio to text using AssemblyAI's Universal-3 Pro model, trained on 12.5+ million hours of audio data. Supports context-aware prompting via plain-language instructions and keyterms (up to 1000 words/phrases, max 6 words per phrase) to control transcription behavior. Provides word-level timestamps, speaker role identification, code-switching support, and verbatim mode. Processes audio asynchronously via REST API with per-hour-of-audio billing ($0.21/hr for Universal-3 Pro, $0.15/hr for legacy Universal-2 supporting 99 languages).
Universal-3 Pro achieves market-leading multilingual accuracy through training on 12.5+ million hours of audio and supports context-aware prompting (plain-language instructions + keyterms) to customize transcription behavior without fine-tuning, differentiating from competitors like Google Cloud Speech-to-Text or AWS Transcribe that require separate model selection or lack flexible prompting
Faster time-to-accuracy than competitors for domain-specific vocabulary because keyterms prompting doesn't require model retraining, and word-level timestamps are native rather than post-processed
real-time streaming speech-to-text transcription with speaker role identification
Medium confidenceProvides real-time transcription of live audio streams using Universal-3 Pro model via WebSocket-based streaming API. Supports speaker role identification (by name or role, not generic diarization labels) and is built on AssemblyAI's proprietary Voice AI stack optimized for production voice agents. Processes audio with sub-second latency for interactive applications like live call transcription, voice agent interactions, and real-time meeting captions. Billed at $4.50/hr of audio processed.
Built on proprietary Voice AI stack end-to-end optimized for production voice agents with native speaker role identification (by name/role, not generic labels) and WebSocket streaming, whereas competitors like Google Cloud Speech-to-Text or Azure Speech Services use generic speaker diarization and require separate agent orchestration frameworks
Lower latency and more natural speaker identification for voice agents because it's purpose-built for conversational AI rather than adapted from batch transcription models
custom spelling and keyterms prompting with vocabulary control
Medium confidenceEnables customization of transcription output by providing domain-specific terminology, custom spellings, or keyterms that should be recognized and preserved in the transcript. Supports up to 1000 words/phrases with a maximum of 6 words per phrase. Implemented as a prompting feature that influences the transcription model's output without requiring model fine-tuning. Billed at $0.05/hr of audio processed for Universal-3 Pro (included in base price) and $0.05/hr for Universal-2. Enables accurate transcription of specialized vocabulary, proper nouns, product names, and domain-specific terminology.
Supports flexible prompting with up to 1000 keyterms (max 6 words per phrase) without requiring model fine-tuning, enabling rapid vocabulary customization for different domains. Implemented as a native feature of Universal-3 Pro (included in base price) and available for Universal-2 ($0.05/hr), whereas competitors like Google Cloud Speech-to-Text require separate phrase lists or custom model training
Faster vocabulary customization than fine-tuning custom models because keyterms prompting works with pre-trained models, and more flexible than static phrase lists because prompting can handle context-dependent variations
lemur llm integration for audio-native ai tasks
Medium confidenceApplies large language models (LLMs) directly to audio data via AssemblyAI's LeMUR (Language Model on Embedded Representations) framework, enabling AI-powered tasks like summarization, question-answering, entity extraction, and custom analysis without requiring separate transcript processing. Processes audio through the transcription pipeline and applies LLM reasoning directly on the transcript representation. Specific LLM models supported, pricing, and integration details not documented in available material. Enables end-to-end audio intelligence workflows without chaining multiple services.
Integrates LLM reasoning directly into the audio processing pipeline via LeMUR framework, enabling audio-native AI tasks without separate transcript extraction or LLM service calls. Processes audio end-to-end with a single API call, whereas competitors require chaining transcription + separate LLM services
Simpler integration than separate services because LLM reasoning happens within AssemblyAI's pipeline, and potentially more accurate because LLM can leverage transcript confidence scores and audio metadata for better reasoning
verbatim transcription mode with filler word preservation
Medium confidenceTranscription mode that preserves filler words, false starts, and non-standard speech patterns exactly as spoken, without normalization or cleanup. Implemented as a transcription parameter that disables automatic filler word removal and speech normalization, returning a verbatim record of the audio content. Useful for linguistic analysis, legal documentation, or accessibility applications requiring exact speech representation. Included in base transcription cost (no additional billing).
Native verbatim mode that preserves exact speech without normalization, enabling accurate linguistic analysis and legal documentation. Implemented as a transcription parameter rather than a separate service, whereas competitors typically require post-processing or manual review to achieve verbatim accuracy
More accurate verbatim transcription than post-processing approaches because it preserves speech at the transcription level, and simpler integration because verbatim mode is a single API parameter
code-switching support for multilingual audio
Medium confidenceHandles audio containing multiple languages mixed within a single conversation (code-switching), accurately transcribing each language segment and optionally identifying language boundaries. Implemented as a native feature of Universal-3 Pro that detects language switches and transcribes each segment in the appropriate language. Enables accurate transcription of multilingual conversations without requiring separate language-specific models or manual language selection. Specific language pair support and language detection accuracy not documented in available material.
Native code-switching support in Universal-3 Pro that automatically detects and transcribes multiple languages without manual language selection, enabling accurate multilingual transcription. Implemented as a single model rather than requiring separate language-specific models or manual switching, whereas competitors typically require explicit language selection or separate models per language
More accurate code-switching transcription than language-specific models because it's trained to handle language mixing, and simpler integration because no manual language switching is required
word-level timestamps and confidence scores for transcript synchronization
Medium confidenceProvides precise timing information for each word in the transcript (start and end timestamps) along with per-word confidence scores indicating transcription accuracy. Implemented as a native feature of the transcription output that returns word-level metadata for synchronization with audio/video playback, interactive transcript building, or quality analysis. Enables downstream applications like interactive transcripts, video captions, and transcript-based search with playback seeking.
Native word-level timestamps and confidence scores integrated into the transcription output, enabling precise synchronization without separate alignment processing. Provides per-word confidence for quality analysis, whereas competitors typically provide only sentence-level or segment-level confidence
More precise transcript synchronization than post-processing alignment because timestamps are generated during transcription, and more granular quality analysis because per-word confidence enables identification of specific problem areas
word-level timestamps and temporal alignment
Medium confidenceReturns precise word-level timing information for each word in the transcript, enabling applications to synchronize text with audio playback, highlight words as they're spoken, or extract segments by time range. Timestamps are returned in milliseconds with start and end times per word.
Word-level timestamps with millisecond precision enable direct audio-text synchronization without external alignment tools, supporting interactive transcript players and caption generation
More precise than Google Cloud Speech-to-Text word timing (which has documented latency issues); integrated into transcription output without separate alignment API
speaker diarization with segment-level speaker labels
Medium confidenceSegments transcript by speaker and assigns speaker labels to each segment, enabling identification of who said what in multi-speaker audio. Implemented as an add-on feature to both Universal-3 Pro and Universal-2 models, processing audio asynchronously and returning speaker-labeled segments in the transcript JSON response. Billed at $0.02/hr of audio processed (in addition to base transcription cost). Does not require pre-configuration of speaker count or identities.
Offered as a low-cost add-on ($0.02/hr) to existing transcription models rather than a separate service, enabling flexible speaker diarization without model switching. Integrates seamlessly with both Universal-3 Pro and Universal-2, whereas competitors like Google Cloud Speech-to-Text or AWS Transcribe require separate API calls or model selection
Cheaper than competitors for speaker diarization when combined with AssemblyAI's base transcription cost, and simpler integration because it's a single API parameter rather than separate service calls
pii redaction with entity detection and masking
Medium confidenceAutomatically detects and redacts personally identifiable information (PII) from transcripts, including person names, company names, email addresses, dates, and locations. Implemented as a speech understanding add-on feature that processes the transcript output and masks or removes sensitive entities. Returns redacted transcript with optional entity metadata for compliance and privacy workflows. Specific masking strategy (replacement tokens, hashing, removal) not documented in available material.
Integrated as a native speech understanding feature within the transcription pipeline rather than a post-processing step, enabling PII detection at the acoustic level before transcript generation. Detects multiple entity types (names, companies, emails, dates, locations) in a single pass, whereas competitors like AWS Transcribe require separate entity recognition services or manual configuration
Faster PII redaction than post-processing approaches because detection happens during transcription, and simpler integration than chaining multiple NLP services for entity recognition
content moderation with policy violation detection
Medium confidenceAnalyzes transcript content for policy violations, inappropriate language, or flagged content categories. Implemented as a speech understanding add-on feature that processes the transcript and returns moderation scores or flags for content categories. Specific moderation categories, confidence thresholds, and flagging logic not documented in available material. Enables content filtering workflows for platforms with community guidelines or compliance requirements.
Integrated into the transcription pipeline as a native speech understanding feature rather than a separate moderation service, enabling policy violation detection at the acoustic level. Processes audio directly without requiring separate text moderation APIs, whereas competitors typically require chaining transcription + text moderation services
Simpler integration than separate moderation services because it's a single API feature, and potentially more accurate for audio-specific violations (tone, speech patterns) that text-only moderation might miss
automatic transcript summarization with key point extraction
Medium confidenceGenerates abstractive summaries of transcribed audio content, extracting key points, main topics, and action items from the full transcript. Implemented as a speech understanding add-on feature that processes the transcript and returns a structured summary. Specific summarization algorithm (extractive vs abstractive), summary length control, and key point extraction logic not documented in available material. Enables rapid content review and knowledge extraction from long-form audio.
Integrated as a native speech understanding feature within the transcription pipeline rather than a separate summarization service, enabling summary generation directly from audio without intermediate transcript processing. Combines transcription + summarization in a single API call, whereas competitors require chaining transcription + separate text summarization services
Faster time-to-summary than separate services because summarization happens during transcription processing, and potentially more accurate because it can leverage audio-level features (emphasis, tone, speech patterns) that text-only summarization misses
sentiment analysis with emotion detection per speaker segment
Medium confidenceAnalyzes the emotional tone and sentiment of transcript segments, detecting positive, negative, or neutral sentiment and optionally emotion categories (confidence, frustration, satisfaction, etc.). Implemented as a speech understanding add-on feature that processes the transcript and returns sentiment scores per segment or speaker. Specific emotion categories, scoring methodology, and segment granularity not documented in available material. Enables sentiment-driven insights from customer interactions, user research, and team communications.
Integrated as a native speech understanding feature within the transcription pipeline, enabling sentiment detection directly from audio without separate text analysis. Can leverage acoustic features (tone, pitch, speech rate) in addition to transcript content for more accurate emotion detection, whereas text-only sentiment analysis services lack audio context
More accurate emotion detection than text-only services because it analyzes both transcript content and acoustic features (tone, emphasis, speech patterns), and simpler integration because sentiment analysis happens in a single API call rather than chaining services
medical-optimized transcription with healthcare terminology
Medium confidenceSpecialized transcription mode optimized for medical and healthcare conversations, with enhanced recognition of medical terminology, drug names, anatomical terms, and healthcare-specific vocabulary. Implemented as an add-on feature to both Universal-3 Pro and Universal-2 models, processing audio asynchronously and returning transcripts with improved accuracy for medical content. Billed at $0.15/hr of audio processed (in addition to base transcription cost). Enables compliance with healthcare documentation standards (HIPAA, medical record requirements).
Specialized transcription mode trained on medical audio and healthcare vocabulary, enabling higher accuracy for medical terminology without requiring separate medical transcription services or manual correction workflows. Integrated as an add-on to standard models rather than a separate service, whereas competitors like Google Cloud Speech-to-Text or AWS Transcribe lack healthcare-specific optimization
Lower error rates for medical terminology than generic transcription services because the model is specifically trained on healthcare language, and simpler integration than separate medical transcription services that require manual review
entity extraction with named entity recognition (ner)
Medium confidenceAutomatically detects and extracts named entities from transcripts, including person names, company names, email addresses, dates, and locations. Implemented as a native feature of the transcription output that identifies entity boundaries and types without requiring separate NLP processing. Returns entity metadata with positions in the transcript for downstream processing, indexing, or knowledge base construction. Enables rapid information extraction from unstructured audio content.
Native entity extraction integrated into the transcription pipeline rather than a separate NLP service, enabling entity detection directly from audio without intermediate transcript processing. Detects multiple entity types (names, companies, emails, dates, locations) in a single pass with position metadata for precise extraction, whereas competitors require chaining transcription + separate NER services
Faster entity extraction than separate NER services because detection happens during transcription, and more accurate because it can leverage acoustic context (emphasis, speech patterns) that text-only NER misses
filler word detection and removal
Medium confidenceIdentifies filler words and non-speech sounds (um, uh, ah, like, you know, etc.) in transcripts and optionally removes or flags them. Implemented as a native feature of the transcription output that detects filler words at the word level and returns them with position metadata. Enables transcript cleanup for professional documentation, presentation materials, or speech analysis. Specific filler word list and detection methodology not documented in available material.
Native filler word detection integrated into the transcription pipeline rather than a post-processing step, enabling detection at the acoustic level where filler words are most accurately identified. Provides position metadata for precise removal or analysis, whereas competitors require separate text processing or manual editing
More accurate filler word detection than text-only approaches because it analyzes acoustic features (duration, pitch, speech patterns) in addition to transcript content, and simpler integration because detection happens during transcription
Capabilities are decomposed by AI analysis. Each maps to specific user intents and improves with match feedback.
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Best For
- ✓Teams building transcription features into SaaS products (meeting recorders, podcast platforms, accessibility tools)
- ✓Enterprises processing multilingual audio content (customer support recordings, international conferences)
- ✓Developers needing high-accuracy transcription with domain-specific vocabulary control
- ✓Teams building voice agent platforms or conversational AI applications
- ✓Contact centers and customer support platforms requiring live call transcription
- ✓Meeting software providers adding real-time captioning features
- ✓Live event platforms (webinars, conferences) needing instant transcription
- ✓Technical and specialized industries (software, medical, legal) with domain-specific vocabulary
Known Limitations
- ⚠Maximum audio duration and file size limits not documented in available material
- ⚠Supported audio formats not specified in provided documentation
- ⚠Asynchronous processing only for pre-recorded audio — real-time transcription requires Voice Agent API at higher cost ($4.50/hr vs $0.21/hr)
- ⚠Keyterms prompting limited to 1000 total words/phrases with 6-word maximum per phrase
- ⚠Universal-3 Pro language support limited to English, Spanish, German, French, Italian, Portuguese (expanding); legacy Universal-2 supports 99 languages but is less accurate
- ⚠Significantly higher cost than pre-recorded transcription ($4.50/hr vs $0.21/hr for Universal-3 Pro)
Requirements
Input / Output
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About
AI speech-to-text API with intelligence features. Universal-2 model for transcription. Features speaker labels, content moderation, PII redaction, summarization, sentiment analysis, and LeMUR for applying LLMs to audio data.
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