| 0 |
| 0 |
| Match Graph | 0 | 0 |
| Pricing | Free | Free |
| Capabilities | 15 decomposed | 12 decomposed |
| Times Matched | 0 | 0 |
Intercepts audio streams at the application or driver level during active communication sessions and applies real-time noise suppression to remove background noise, echo, and cross-talk before audio reaches the listener. Processing occurs locally on the client device to minimize latency, with claims of sub-500ms processing overhead. The system operates transparently across any communication application (Zoom, Teams, Google Meet, etc.) without requiring application-specific plugins.
Unique: Operates at audio driver level rather than application-level, enabling transparent integration with 'any communication application' without requiring per-app plugins or API integrations. Claims '#1 noise cancellation' positioning but provides no comparative benchmarks or technical specifications for validation.
vs alternatives: Broader application compatibility than Zoom's native noise suppression or Teams' background noise reduction, but lacks published latency metrics or accuracy benchmarks compared to specialized audio processing tools.
Converts spoken audio to text in real-time during active meetings, displaying captions as participants speak. The system captures audio from the communication application, processes it through a speech-to-text model (model type and training data unknown), and streams transcripts to the user interface with claimed support for multiple languages. Transcripts are stored in Krisp's cloud system for post-meeting access and integration with downstream tools via webhooks or API.
Unique: Integrates transcription directly into the meeting experience with live caption display, rather than post-meeting transcription. Claims 'bot-free' transcription (technical meaning unclear) and stores transcripts for persistent access and integration, but provides no model specifications or accuracy metrics.
vs alternatives: Captures transcripts automatically without requiring separate recording or transcription service, but lacks speaker identification and accuracy benchmarks compared to specialized services like Rev or Otter.ai.
Exposes voice translation as an API endpoint in the Krisp Voice AI SDK, allowing developers to programmatically translate audio from one language to another in voice applications and AI agents. The API accepts audio input in the source language and returns audio output in the target language. Supported language pairs, translation quality, and latency are not disclosed. Likely used for enabling multilingual voice agents or real-time translation in voice applications.
Unique: Exposes voice translation as a programmatic API for developers building voice applications, enabling real-time multilingual voice interactions. However, supported language pairs, translation quality, and pricing are completely undisclosed.
vs alternatives: Available as an SDK API for integration into voice applications, but lacks the language coverage transparency, quality metrics, and documented latency of specialized real-time translation APIs like Google Cloud Translation or Microsoft Translator.
Exposes noise cancellation as an API endpoint in the Krisp Voice AI SDK, allowing developers to programmatically remove background noise from audio streams in voice applications and AI agents. The API accepts noisy audio input and returns cleaned audio with noise suppressed. The noise cancellation algorithm, supported noise types, and effectiveness metrics are not disclosed. Likely used for improving speech recognition accuracy or voice quality in voice applications.
Unique: Exposes noise cancellation as a programmatic API for developers building voice applications, enabling audio preprocessing at scale. However, the algorithm, effectiveness metrics, supported formats, and pricing are completely undisclosed.
vs alternatives: Available as an SDK API for integration into voice applications, but lacks the algorithm transparency, effectiveness benchmarks, and documented latency of specialized audio processing APIs like Krisp's own real-time noise cancellation or Google Cloud Speech Enhancement.
Provides real-time AI assistance to call center agents during active customer calls, offering suggestions, guidance, or information to improve call quality and customer satisfaction. The system analyzes the call in real-time, detects customer intent or issues, and provides contextual suggestions to the agent via a sidebar or dashboard. The AI model, suggestion generation approach, and integration with call center systems (Genesys, Avaya, etc.) are not disclosed. Pricing and feature details are completely unknown.
Unique: Provides real-time AI assistance to call center agents during active calls, integrated into the call center workflow. However, the AI model, suggestion generation approach, call center system integrations, and pricing are completely undisclosed.
vs alternatives: Integrated into Krisp's call center product for real-time agent guidance, but lacks the documentation, integration transparency, and proven effectiveness of specialized agent assist platforms like Genesys Predictive Engagement or Avaya Oceana.
Analyzes call center recordings to extract insights on call quality, compliance, and agent performance. The system processes recorded calls (audio and transcripts) to generate call scores, detect compliance violations, identify training opportunities, and track agent performance metrics. The analytics model, scoring methodology, and compliance rule definitions are not disclosed. Pricing and feature details are completely unknown.
Unique: Provides post-call analytics for compliance and quality monitoring in call centers, integrated into Krisp's call center product. However, the scoring methodology, compliance rule definitions, supported frameworks, and pricing are completely undisclosed.
vs alternatives: Integrated into Krisp's call center platform for compliance monitoring, but lacks the transparency, compliance certification, and proven effectiveness of specialized call analytics platforms like Verint or NICE.
Enhances the conversational flow of AI voice agents by improving turn-taking behavior (detecting when the user has finished speaking and the agent should respond). The system analyzes audio and speech patterns to determine optimal response timing, reducing awkward silences or interruptions. The algorithm and accuracy metrics are not disclosed. Likely used to improve the naturalness of voice agent interactions.
Unique: Provides turn-taking improvement as an SDK capability for voice agents, enabling more natural conversational flow. However, the algorithm, accuracy metrics, supported languages, and pricing are completely undisclosed.
vs alternatives: Integrated into Krisp's Voice AI SDK for voice agents, but lacks the documentation, accuracy benchmarks, and integration examples of specialized voice agent frameworks like Voiceflow or Rasa.
Processes the complete meeting transcript and audio after the meeting concludes, generating a natural language summary of key discussion points and extracting a structured list of action items with implied owners or deadlines. The summarization model type, training approach, and context window size are not disclosed. Summaries are generated server-side and stored in Krisp's cloud system, with export to integrations (Slack, HubSpot, Pipedrive, Zapier) via webhook API.
Unique: Combines summarization and action item extraction in a single post-meeting process, with direct integration to business tools (HubSpot, Pipedrive, Slack) via webhook API. However, no model specifications, accuracy metrics, or customization options are disclosed.
vs alternatives: Integrated into the meeting workflow with automatic export to CRM/task tools, but lacks the customization, accuracy transparency, and speaker attribution of specialized meeting intelligence platforms like Gong or Chorus.
+7 more capabilities
Transcribes audio in 98 languages to text in the original language using a Transformer sequence-to-sequence architecture trained on 680,000 hours of diverse internet audio. The system uses mel spectrogram feature extraction via FFmpeg integration, processes audio through an AudioEncoder that generates embeddings, then applies an autoregressive TextDecoder with task-specific tokens to produce language-native transcriptions. Language-specific models (e.g., tiny.en, base.en) optimize for English-only workloads with reduced parameter count.
Unique: Unified multitasking Transformer model replaces traditional multi-stage speech pipelines (VAD → language detection → ASR → post-processing) with single forward pass; trained on 680K hours of internet audio providing robustness to background noise, accents, and technical speech unlike studio-trained competitors
vs alternatives: Outperforms Google Cloud Speech-to-Text and Azure Speech Services on non-English languages and noisy audio due to diverse training data; open-source allows local deployment without API latency or privacy concerns
Translates non-English speech directly to English text in a single forward pass using the same Transformer architecture as transcription, but with a translation task token prepended to the decoder input. The model learns to skip intermediate transcription and generate English output directly from audio embeddings, avoiding cascading errors from intermediate transcription steps. Supports 98 source languages translating to English only.
Unique: Direct audio-to-English translation without intermediate transcription step — the decoder learns to skip source language text generation and output English directly, reducing error propagation and latency compared to cascade approaches (transcribe → translate)
vs alternatives: Faster and more accurate than Google Translate + Google Speech-to-Text pipeline because it avoids intermediate transcription errors; open-source allows offline deployment unlike cloud translation APIs
Whisper Large v3 scores higher at 59/100 vs Krisp at 55/100.
Need something different?
Search the match graph →© 2026 Unfragile. Stronger through disorder.
Normalizes variable-length audio to exactly 30 seconds via `whisper.pad_or_trim()`: audio shorter than 30 seconds is padded with silence (zeros) to reach 30 seconds, audio longer than 30 seconds is trimmed to first 30 seconds. This ensures consistent input shape (80×3000 mel spectrogram) for the model, avoiding shape mismatches and enabling batch processing. Padding strategy is simple zero-padding rather than sophisticated techniques like repetition or interpolation.
Unique: Simple zero-padding strategy is computationally efficient and deterministic, but acoustically naive — alternative approaches (silence detection, repetition) not implemented in base library
vs alternatives: Simpler than librosa-based preprocessing with sophisticated padding; deterministic behavior aids reproducibility; zero-padding is fast but may introduce artifacts vs more sophisticated techniques
Returns transcription results as structured JSON objects containing: transcribed text, language code, duration, segments (with timing and text), and optional confidence metrics. The `model.transcribe()` API returns a dictionary with keys like 'text' (full transcript), 'language' (detected language), 'segments' (list of segment objects with start/end times and text). This structured format enables downstream processing (subtitle generation, database storage, API responses) without string parsing.
Unique: Structured output format is built into high-level API rather than requiring manual parsing — segments include timing and text, enabling direct use for subtitle generation or timeline-based applications
vs alternatives: More structured than raw text output; less detailed than forced alignment tools that provide phoneme-level information; JSON format is language-agnostic and integrates easily with web APIs
Detects the spoken language in audio by processing mel spectrograms through the AudioEncoder and using a language classification head that outputs probability distributions over 98 supported languages. The model leverages 680K hours of multilingual training data to recognize language characteristics from acoustic features alone, without requiring transcription. Language detection occurs as a preliminary step in the transcription pipeline and can be called independently via the language detection task token.
Unique: Language detection is integrated into the same Transformer model as transcription/translation via task tokens, allowing shared AudioEncoder computation and single model load — not a separate classifier, reducing memory footprint and inference overhead
vs alternatives: More accurate than acoustic-only language identification (e.g., librosa-based approaches) because it leverages semantic understanding from 680K hours of training; faster than transcription-based detection (identify language from first few words) because it uses acoustic features directly
Provides six model variants (tiny 39M, base 74M, small 244M, medium 769M, large 1550M, turbo 809M) with different parameter counts, VRAM requirements (1-10GB), and inference speeds (10x-1x relative to large). Each size trades accuracy for speed — tiny runs ~10x faster but with ~5-10% lower WER (word error rate), while large provides best accuracy at 10GB VRAM cost. Turbo variant (809M params) optimizes large-v3 for 8x speedup with minimal accuracy loss but lacks translation support.
Unique: Discrete model size family with published speed/accuracy/VRAM tradeoff matrix allows developers to make informed selection based on deployment constraints; turbo variant represents architectural optimization (knowledge distillation or pruning) achieving 8x speedup with <5% accuracy loss, distinct from simply using smaller base model
vs alternatives: More transparent tradeoff options than Whisper API (single model) or competitors like Deepgram (proprietary size selection); open-source allows local benchmarking on own hardware rather than relying on vendor performance claims
Automatically segments audio longer than 30 seconds into overlapping windows, processes each window independently through the transcription pipeline, and merges results with overlap handling to produce seamless full-length transcripts. The system uses `whisper.pad_or_trim()` to normalize each segment to exactly 30 seconds (padding with silence if needed), then applies the decoder to each segment and concatenates outputs while managing word-level boundaries and timestamp continuity across segment edges.
Unique: Sliding window approach with automatic overlap and boundary handling is built into high-level `model.transcribe()` API — developers don't manually implement segmentation, unlike lower-level APIs that require explicit window management
vs alternatives: Simpler than building custom segmentation logic; more robust than naive concatenation because it handles word-level boundary issues; faster than streaming approaches because it processes segments in parallel on GPU
Generates precise word-level timestamps (start and end times in milliseconds) for each word in the transcript by leveraging the decoder's attention weights and token alignment information. The system maps output tokens back to audio frames using the attention mechanism, then converts frame indices to millisecond timestamps based on the mel spectrogram hop length (20ms per frame). Timestamps are returned as part of the structured output alongside transcribed text.
Unique: Word-level timestamps are derived from attention weight alignment rather than separate timestamp prediction head — leverages existing decoder computation without additional model parameters, but introduces ±100-200ms uncertainty from frame quantization
vs alternatives: More granular than segment-level timestamps (which only mark 30-second boundaries); less accurate than forced alignment tools (e.g., Montreal Forced Aligner) but requires no phonetic lexicon or manual annotation
+4 more capabilities