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| Ecosystem | 0 | 0 |
| Match Graph | 0 | 0 |
| Pricing | Free | Free |
| Starting Price | $0.00250/min | — |
| Capabilities | 16 decomposed | 11 decomposed |
| Times Matched | 0 | 0 |
Converts 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).
Unique: 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
vs alternatives: 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
Provides 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.
Unique: 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
vs alternatives: 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
Enables 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.
Unique: 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
vs alternatives: 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
Applies 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.
Unique: 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
vs alternatives: 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
Transcription 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).
Unique: 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
vs alternatives: 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
Handles 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.
Unique: 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
vs alternatives: 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
Provides 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.
Unique: 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
vs alternatives: 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
Returns 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.
Unique: Word-level timestamps with millisecond precision enable direct audio-text synchronization without external alignment tools, supporting interactive transcript players and caption generation
vs alternatives: More precise than Google Cloud Speech-to-Text word timing (which has documented latency issues); integrated into transcription output without separate alignment API
+8 more capabilities
Transcribes audio in 98 languages to text in the original language using a unified Transformer sequence-to-sequence architecture with a shared AudioEncoder that processes mel spectrograms into language-agnostic embeddings, then a TextDecoder that generates tokens autoregressively. The system handles variable-length audio by padding or trimming to 30-second segments and uses task-specific tokens to signal transcription mode, enabling a single model to handle multiple languages without language-specific branches.
Unique: Uses a single shared AudioEncoder across all 98 languages rather than language-specific encoders, trained on 680,000 hours of diverse internet audio enabling zero-shot cross-lingual transfer. The mel-spectrogram preprocessing pipeline (via log_mel_spectrogram) standardizes variable audio into fixed 30-second segments, allowing the same model weights to handle any language without retraining.
vs alternatives: Outperforms language-specific ASR models on low-resource languages and handles 98 languages in a single model, whereas Google Cloud Speech-to-Text and Azure Speech Services require separate API calls per language and have higher latency due to cloud round-trips.
Translates non-English speech directly to English text by using a task-specific token in the TextDecoder that signals translation mode, bypassing the need for intermediate transcription-then-translation pipelines. The AudioEncoder processes mel spectrograms identically to transcription, but the decoder generates English tokens directly from audio embeddings, reducing latency and error propagation compared to cascaded systems.
Unique: Implements end-to-end speech translation via task-specific decoder tokens rather than cascaded transcription-then-translation, eliminating intermediate text generation and reducing error propagation. The decoder uses a special token prefix to signal translation mode, allowing the same AudioEncoder and TextDecoder weights to handle both transcription and translation without separate model branches.
Whisper CLI scores higher at 58/100 vs AssemblyAI API at 55/100.
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vs alternatives: Faster and more accurate than cascaded pipelines (Google Translate + Speech-to-Text) because it avoids intermediate transcription errors and reduces round-trip latency; however, less flexible than specialized translation models for domain-specific or style-controlled output.
Exposes two levels of API abstraction: a high-level transcribe() function that handles end-to-end transcription with automatic audio loading, preprocessing, and result formatting, and a low-level decode() function that provides fine-grained control over decoding options (beam width, temperature, language constraints). The high-level API is suitable for simple use cases, while the low-level API enables advanced customization for researchers and developers building complex pipelines.
Unique: Provides dual-level API abstraction with transcribe() for simplicity and decode() for control, allowing users to start with simple code and gradually adopt lower-level APIs as needs become more complex. The high-level API automatically handles audio loading, preprocessing, and result formatting, while the low-level API exposes DecodingOptions for fine-grained control.
vs alternatives: More flexible than single-level APIs (like some cloud services that only expose high-level endpoints) because it supports both simple and advanced use cases; however, requires more learning and boilerplate than opinionated frameworks that make decisions for users.
Detects the spoken language in audio by generating a language token from the AudioEncoder embeddings before decoding text, using the model's multilingual training to recognize acoustic patterns distinctive to each language. The system identifies language during the initial decoding step and can be queried directly via the language identification task token, enabling language detection without full transcription.
Unique: Leverages the shared AudioEncoder's learned acoustic representations across 680,000 hours of multilingual training data to identify language without explicit language classification head — the language token emerges naturally from the decoder's first output token, making detection a byproduct of the transcription architecture rather than a separate classifier.
vs alternatives: Supports 98 languages in a single model with zero-shot capability on low-resource languages, whereas language identification libraries like langdetect or textcat require separate training or pre-built models for each language and cannot handle audio directly.
Converts raw audio files in multiple formats (MP3, WAV, M4A, FLAC, OGG) to mel-spectrogram features via FFmpeg decoding and log-scale mel-frequency filtering, then normalizes variable-length audio to fixed 30-second segments via padding or trimming. The pipeline uses whisper.load_audio() for format-agnostic decoding, whisper.pad_or_trim() for segment normalization, and whisper.log_mel_spectrogram() for feature extraction, enabling the model to process diverse audio sources with consistent preprocessing.
Unique: Integrates FFmpeg as a subprocess for format-agnostic audio decoding rather than using Python-only libraries, enabling support for any FFmpeg-compatible format without maintaining codec-specific parsers. The fixed 30-second segment design allows the model to use a single AudioEncoder without variable-length handling, simplifying the architecture at the cost of preprocessing inflexibility.
vs alternatives: Handles more audio formats than librosa-based pipelines (which require separate codec installations) and avoids the latency of cloud-based audio conversion services; however, less flexible than custom preprocessing pipelines that can adjust segment length or mel-spectrogram parameters.
Generates transcription or translation tokens autoregressively using a TextDecoder that processes AudioEncoder embeddings and previously generated tokens, with support for multiple decoding strategies including greedy decoding, beam search, and temperature-based sampling. The system uses a sliding-window context approach to handle audio longer than 30 seconds by processing overlapping segments and merging results, and supports DecodingOptions for fine-grained control over decoding behavior (beam width, temperature, language constraints).
Unique: Implements sliding-window decoding for long audio by processing overlapping 30-second segments and merging results via token-level overlap detection, avoiding the need to retrain the model for variable-length inputs. The DecodingOptions abstraction allows fine-grained control over beam width, temperature, language constraints, and other decoding parameters without modifying model weights.
vs alternatives: More flexible than fixed-greedy-decoding-only systems (like some edge-deployed models) because it supports beam search and temperature sampling; however, slower than specialized streaming decoders (like Kaldi or Vosk) that use HMM-based decoding optimized for low-latency online processing.
Generates precise word-level timestamps by aligning decoded tokens to audio segments using the model's internal attention weights and token probabilities, enabling subtitle generation and fine-grained audio-text synchronization. The system decodes text at the segment level (30 seconds), then uses token timing information to map each word back to its position in the original audio, producing timestamps accurate to ~100ms granularity.
Unique: Derives word-level timestamps from the model's token-to-audio alignment without a separate alignment model, using the decoder's implicit timing information from mel-spectrogram frame positions. The approach avoids the need for external forced-alignment tools (like Montreal Forced Aligner) by leveraging the model's learned audio-text correspondence.
vs alternatives: Simpler than forced-alignment pipelines (Montreal Forced Aligner + Whisper) because it uses a single model; however, less accurate than specialized alignment models trained specifically on timing prediction, and requires custom implementation to extract timing metadata from the model.
Provides six model sizes (tiny, base, small, medium, large, turbo) with parameter counts ranging from 39M to 1550M, enabling users to select optimal speed-accuracy tradeoffs based on hardware constraints and latency requirements. Each model has English-only variants (tiny.en, base.en, small.en) that sacrifice multilingual capability for 10-40% speed improvement, and the turbo model (809M) optimizes large-v3 for 8x faster inference with minimal accuracy degradation but no translation support.
Unique: Provides both multilingual and English-only variants for smaller models (tiny, base, small) to enable language-specific optimization, whereas most speech recognition systems offer only a single model per size. The turbo model represents a specialized optimization of large-v3 for inference speed using knowledge distillation or quantization techniques, not just parameter reduction.
vs alternatives: More granular model selection than Google Cloud Speech-to-Text (which offers only one model per language) and more transparent about speed-accuracy tradeoffs than commercial APIs that hide model details; however, requires manual model selection and management, whereas cloud services handle this automatically.
+3 more capabilities