| 0 |
| 0 |
| Match Graph | 0 | 0 |
| Pricing | Free | Free |
| Capabilities | 11 decomposed | 11 decomposed |
| Times Matched | 0 | 0 |
Captures audio and video from Zoom, Google Meet, and Teams calls directly through browser extension or native app integration without requiring a meeting bot to be invited. The capture mechanism operates client-side at the browser/app level, intercepting the media stream before it reaches the meeting platform's servers, then streams or buffers the raw audio/video for post-processing. This approach eliminates the need for explicit bot invitations and reduces meeting participant friction.
Unique: Eliminates bot-based recording by capturing at the browser/app level rather than injecting a participant into the meeting, reducing UX friction and meeting participant visibility compared to Otter.ai, Fireflies.io, or Fathom which use bot-based approaches
vs alternatives: Superior UX friction vs bot-based competitors because no bot appears in participant list and no explicit invite is required, though technical implementation details are opaque
Converts captured meeting audio into timestamped text transcripts with speaker identification, enabling users to search and reference specific moments in calls. The transcription pipeline processes audio post-meeting (latency unknown) and generates word-level timestamps, allowing clips and summaries to reference exact moments. Speaker attribution mechanism is undisclosed but implied by action item extraction and CRM logging features that track who said what.
Unique: Integrates speaker attribution with transcription to enable action-item tracking and CRM logging by speaker, whereas generic transcription tools (Otter.ai, Fireflies) treat transcripts as undifferentiated text without deep speaker-action mapping
vs alternatives: Tighter integration with downstream CRM and action-item systems because speaker attribution is built into the transcription pipeline rather than post-processed, reducing latency and improving accuracy of speaker-action mapping
Offers a free tier that includes unlimited meeting recording, transcription, and basic summarization without time limits or meeting count restrictions. The free tier is designed to reduce friction for individual users and small teams to adopt tl;dv before upgrading to paid features. Specific limitations of the free tier (e.g., storage limits, feature restrictions, user seat limits) are not disclosed in documentation.
Unique: Offers unlimited meeting recording on free tier without meeting count or storage limits (claimed), whereas competitors like Otter.ai and Fireflies impose strict free tier limits (e.g., 600 minutes/month for Otter.ai) to drive paid upgrades
vs alternatives: Lower barrier to entry for individual users and small teams because unlimited recording on free tier means no surprise paywalls when hitting quota limits, whereas competitors force upgrade after hitting free tier limits
Generates post-meeting summaries using AI models with user-selectable frameworks (MEDDIC, Smart AI Topics, custom) that structure the summary output to match sales, product, or marketing workflows. The summarization engine processes the full transcript and produces abstractive summaries (not just extractive highlights) in 1-5 minutes (claimed 'instantly' but latency unknown). Users can define custom summary templates via prompts, enabling role-specific summaries (e.g., 'extract only objections and how they were handled' for sales, 'extract feature requests and prioritize by frequency' for product).
Unique: Offers framework-based summarization (MEDDIC, Smart AI Topics) with custom prompt templates, whereas competitors like Otter.ai and Fireflies provide generic summaries without role-specific structuring or template customization
vs alternatives: Better for sales and product teams because summaries are pre-structured for domain-specific workflows (MEDDIC for sales, feature extraction for product) rather than generic bullet-point recaps, reducing post-processing work
Identifies commitments, tasks, and next steps mentioned during meetings and extracts them as structured action items with speaker attribution, due date inference, and optional CRM task creation. The extraction uses NLP/LLM-based pattern matching to identify phrases like 'I'll send you', 'we need to', 'by next week', etc., and maps them to speakers and inferred deadlines. Extracted action items can be automatically logged to CRM systems or exported as task lists.
Unique: Combines speaker attribution with action item extraction to automatically assign tasks to the right person, whereas generic action item tools (Otter.ai, Fireflies) extract items without reliable speaker mapping, requiring manual assignment
vs alternatives: More actionable than competitor action item extraction because items are pre-assigned to speakers, reducing manual work and improving accountability tracking in CRM workflows
Automatically logs meeting summaries, transcripts, action items, and call outcomes to CRM systems (specific platforms unknown) without manual data entry. The integration maps tl;dv outputs (summary, action items, speaker attribution) to CRM fields (call notes, next steps, deal stage, etc.) and creates or updates CRM records based on meeting participants and detected deal context. Supports auto-drafting of follow-up emails and task creation within the CRM.
Unique: Integrates meeting intelligence (summaries, action items, speaker attribution) directly into CRM workflows with auto-field population and follow-up drafting, whereas competitors like Otter.ai and Fireflies provide transcripts/summaries but require manual CRM entry or generic Zapier integration
vs alternatives: Reduces manual CRM data entry by 80%+ for sales teams because meeting outputs are automatically mapped to CRM fields and tasks, whereas competitors require copy-paste or generic workflow automation that doesn't understand meeting context
Enables full-text and semantic search across all recorded meetings to find specific topics, speakers, or moments, then generates shareable video/audio clips of matching segments. The search mechanism is undisclosed but likely combines transcript keyword matching with semantic embeddings to find conceptually similar moments across meetings. Clip generation extracts the relevant audio/video segment with context (speaker name, timestamp, summary) and produces a shareable link or downloadable file.
Unique: Combines semantic search with automatic clip generation to enable quick sharing of meeting moments, whereas competitors like Otter.ai and Fireflies provide search but require manual clip creation or don't support video clip generation
vs alternatives: Better for marketing and training use cases because clips are automatically generated from search results with context (speaker, timestamp, summary), enabling quick creation of highlight reels without manual video editing
Analyzes patterns across multiple meetings to identify trends, recurring themes, and aggregate insights, then generates custom reports via email or dashboard. The analysis engine processes summaries and transcripts from multiple meetings, applies user-defined custom prompts (e.g., 'extract all customer pain points and rank by frequency'), and produces structured reports with visualizations (charts, tables — format unknown). Reports can be scheduled (daily, weekly, monthly) or generated on-demand.
Unique: Enables custom prompt-based trend analysis across meeting archives, allowing teams to define their own analysis criteria rather than pre-built reports, whereas competitors like Otter.ai and Fireflies focus on individual meeting summaries without cross-meeting aggregation
vs alternatives: More flexible for diverse use cases (sales objection tracking, product feature extraction, marketing pain point identification) because custom prompts allow teams to define their own analysis logic rather than using pre-built report templates
+3 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 tl;dv at 56/100.
Need something different?
Search the match graph →© 2026 Unfragile. Stronger through disorder.
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