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ufal avatar

ufal/whisper_streaming

0
View on GitHub↗
3,642 stars·408 forks·Python·MIT·33 views

Whisper Streaming

Whisper streaming is an automated speech recognition engine designed to convert live audio into text. It functions as a network-based transcription server that accepts raw audio data from remote clients and returns incremental text results in real-time.

The system distinguishes itself through its ability to process audio streams incrementally, allowing for immediate transcription and translation as speech is captured. It incorporates voice activity detection to isolate human speech from background noise and utilizes sliding-window buffering to manage incoming audio segments, ensuring that processing remains efficient and coherent over long durations.

Beyond live transcription, the project supports remote audio processing and provides tools to simulate real-time conditions using pre-recorded files. These capabilities allow for the benchmarking of speech recognition models and the testing of latency under controlled environments. The software is distributed as a Python-based package.

Features

  • Real-Time Audio Transcribers - Processes incoming audio streams incrementally to generate text transcriptions immediately as speech is captured.
  • Transcription APIs - Provides a network interface for remote clients to send raw audio and receive live text results.
  • Incremental Inference Streaming - Feeds audio chunks into a pre-trained model to generate partial text results before the entire stream completes.
  • Speech Recognition Engines - Identifies human voice segments and converts spoken language into written text with low-latency buffering.
  • Real-Time Speech Transcription - Converts live audio streams into text instantly as speech occurs to support automated note-taking workflows.
  • Real-Time Speech-to-Text Servers - Provides a communication interface for transmitting raw audio data to remote servers to receive incremental text results.
  • Speech-to-Text Transcribers - Processes audio files or streams to generate accurate text transcriptions or language translations.
  • Voice Activity Detection - Analyzes incoming audio streams to isolate human speech and prevent background noise from triggering unnecessary inference.
  • Live Voice-to-Text Translation - Processes spoken language in real-time to generate immediate text translations for cross-lingual communication.
  • Audio Over IP - Transmits raw microphone data over a network to a central server for speech recognition tasks.
  • Backend Audio Streaming - Transmits raw audio data over network connections to enable remote microphone input for real-time processing.
  • Audio Streaming Servers - Transmits raw audio bytes over persistent network connections for real-time communication between clients and the server.
  • Real-Time Audio Streaming Buffers - Maintains a rolling memory buffer of incoming audio segments to allow incremental processing while discarding already transcribed data.

Star history

Star history chart for ufal/whisper_streamingStar history chart for ufal/whisper_streaming

How this analysis was created: This summary and feature list are AI-generated from collected project material and can contain mistakes. Stars, license and language are imported from GitHub. Inclusion does not mean that we have tested or audited this project. Check the source documentation for any feature you depend on. Learn more on our About page.

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Projects sharing features with Whisper Streaming

These projects share indexed features with Whisper Streaming. Shared tags can include platform or build tooling; verify the primary use case before treating a result as a replacement.
  • cmusphinx/pocketsphinxcmusphinx avatar

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    PocketSphinx is an offline speech recognition engine that converts raw audio from files or live microphone streams into written text without requiring a network connection. It functions as a speech-to-text library, a real-time transcription engine, and a voice command processor, capable of detecting and transcribing spoken commands from continuous audio streams with configurable acoustic and language models. The engine uses weighted finite-state transducers to represent acoustic, phonetic, and language models as a single search graph for efficient decoding. It employs fixed-point acoustic mod

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  • julius-speech/juliusjulius-speech avatar

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    Julius is a high-performance, open-source speech recognition engine designed for large vocabulary continuous speech recognition. It functions as a comprehensive framework utilizing Hidden Markov Model-based acoustic modeling and N-gram language models to convert live or recorded audio into text. The engine is built to support real-time streaming and provides a network-accessible service that allows external applications to manage recognition sessions and receive transcription results through programmatic commands. The engine distinguishes itself through its modular architecture and support fo

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  • jianchang512/sttjianchang512 avatar

    jianchang512/stt

    4,629View on GitHub↗

    This project is a hardware-accelerated transcription server and offline subtitle generator. It functions as a speech-to-text tool that converts audio and video files into plain text, JSON, and SRT subtitle formats using the Whisper model. The system operates as an OpenAI Audio API emulator, providing a local server that mimics a specific audio interface. This allows it to serve transcriptions to existing client configurations without requiring changes to the client software. The service utilizes GPU acceleration to increase voice recognition speed and includes utilities for hardware detectio

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  • soniqo/speech-swiftsoniqo avatar

    soniqo/speech-swift

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    This project is a comprehensive toolkit for on-device speech recognition, synthesis, and audio processing, specifically engineered for Apple Silicon. It provides a framework for building real-time, full-duplex voice agents that operate entirely offline, leveraging native hardware acceleration to maintain performance and privacy. By utilizing optimized machine learning models, the library enables local execution of complex audio tasks without reliance on external cloud services. The library distinguishes itself through its specialized focus on local, high-performance voice interaction. It incl

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Frequently asked questions

What does ufal/whisper_streaming do?

Whisper streaming is an automated speech recognition engine designed to convert live audio into text. It functions as a network-based transcription server that accepts raw audio data from remote clients and returns incremental text results in real-time.

What are the main features of ufal/whisper_streaming?

The main features of ufal/whisper_streaming are: Real-Time Audio Transcribers, Transcription APIs, Incremental Inference Streaming, Speech Recognition Engines, Real-Time Speech Transcription, Real-Time Speech-to-Text Servers, Speech-to-Text Transcribers, Voice Activity Detection.

Which projects share features with ufal/whisper_streaming?

Projects with overlapping indexed features include: cmusphinx/pocketsphinx — PocketSphinx is an offline speech recognition engine that converts raw audio from files or live microphone streams… julius-speech/julius — Julius is a high-performance, open-source speech recognition engine designed for large vocabulary continuous speech… jianchang512/stt — This project is a hardware-accelerated transcription server and offline subtitle generator. It functions as a… soniqo/speech-swift — This project is a comprehensive toolkit for on-device speech recognition, synthesis, and audio processing,… nl8590687/asrt_speechrecognition — This project is a Chinese automatic speech recognition framework and deep learning system designed to convert spoken… davabase/whisper_real_time — Whisper Real-Time is a speech-to-text engine designed to convert continuous microphone input into written transcripts.…

Curated searches featuring Whisper Streaming

Hand-picked collections where Whisper Streaming appears.
  • Real-Time Speech Transcription Tools
  • Open Source Speech Recognition Engines