Search Results for "speaker recognition"
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Showing 24 open source projects for "speaker recognition"
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sherpa-onnx
Speech-to-text, text-to-speech, and speaker recognition
Speech-to-text, text-to-speech, and speaker recognition using next-gen Kaldi with onnxruntime without an Internet connection. Support embedded systems, Android, iOS, Raspberry Pi, RISC-V, x86_64 servers, websocket server/client, C/C++, Python, Kotlin, C#, Go, NodeJS, Java, Swift, Dart, JavaScript, Flutter. -
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WhisperX
Automatic Speech Recognition with Word-level Timestamps
WhisperX is an advanced speech recognition system built on top of OpenAI’s Whisper model, designed to improve transcription accuracy and timing precision for long-form audio. It addresses key limitations of standard Whisper implementations by introducing voice activity detection and forced alignment techniques to produce word-level timestamps. The system enables batched inference, significantly increasing transcription speed while maintaining high accuracy. -
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The SpeechBrain Toolkit
A PyTorch-based Speech Toolkit
...It is designed to be simple, extremely flexible, and user-friendly. Competitive or state-of-the-art performance is obtained in various domains. SpeechBrain supports state-of-the-art methods for end-to-end speech recognition, including models based on CTC, CTC+attention, transducers, transformers, and neural language models relying on recurrent neural networks and transformers. Speaker recognition is already deployed in a wide variety of realistic applications. SpeechBrain provides different models for speaker recognition, including X-vector, ECAPA-TDNN, PLDA, and contrastive learning. ... -
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Buzz
Buzz transcribes and translates audio offline
Buzz is a desktop application for transcribing and translating audio locally with speech recognition models based on Whisper. It can process audio files, video files, and YouTube links without requiring cloud transcription. Live microphone transcription supports real-time captions and a presentation view for accessible events. Speech separation can improve results on noisy recordings, while speaker identification distinguishes voices within transcribed media. -
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VoiceStudio
VoiceStudio is the open-source, fully-local ElevenLabs alternative
...It can clone voices from short reference recordings or design new voices from characteristics such as accent, pitch, age, and delivery style. Video dubbing combines transcription, translation, speaker preservation, speech synthesis, and final video export. It also supports multi-voice stories, audiobooks, dictation, vocal isolation, and speaker diarization. Multiple TTS and speech-recognition engines can run through CUDA, Apple Silicon, ROCm, or CPU hardware. Desktop, API, MCP, remote-worker, and batch-processing interfaces make it suitable for both interactive and automated workflows while keeping core processing local by default. -
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MiMo-V2.5-ASR
Robust Speech Recognition Across Languages, Dialects
MiMo-V2.5-ASR is an advanced automatic speech recognition system developed as part of Xiaomi’s MiMo AI ecosystem. It is designed to handle complex acoustic environments, including noisy conditions and diverse speaker variations. The model supports multiple languages and dialects, enabling robust transcription across global use cases. It leverages modern deep learning architectures to improve accuracy and adaptability in real-world scenarios. -
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Whisper-WebUI
A Web UI for easy subtitle using whisper model
...It supports multiple input sources including local files, YouTube content, and microphone input, making it versatile for different workflows. Whisper WebUI also includes advanced preprocessing and postprocessing features such as voice activity detection, background music separation, and speaker diarization, enabling more accurate and structured outputs. -
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Scriberr
Self-hosted AI audio transcription
...Unlike cloud-based transcription services, Scriberr runs entirely on the user’s machine, ensuring that sensitive recordings are never sent to third-party servers and remain fully under user control. It leverages modern speech recognition models such as Whisper and other advanced architectures to deliver precise transcripts with word-level timing and speaker identification. The application includes a polished user interface that simplifies the management of recordings, transcripts, and annotations, making it suitable for both casual users and professionals handling large volumes of audio. ... -
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pyVideoTrans
Translate the video from one language to another and embed dubbing
pyVideoTrans is an ambitious open-source multimedia processing project that assembles speech recognition, subtitle generation, AI translation, voice synthesis, and video assembly into a unified pipeline for converting videos from one language to another with embedded dubbing and captions. At its core it runs speech-to-text models to transcribe audio tracks, translates the resulting text into a target language using local or cloud-based translation engines, synthesizes new speech to match the... -
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Step-Audio 2
Multi-modal large language model designed for audio understanding
Step-Audio2 is an advanced, end-to-end multimodal large language model designed for high-fidelity audio understanding and natural speech conversation: unlike many pipelines that separate speech recognition, processing, and synthesis, Step-Audio2 processes raw audio, reasons about semantic and paralinguistic content (like emotion, speaker characteristics, non-verbal cues), and can generate contextually appropriate responses — including potentially generating or transforming audio output. It integrates a latent-space audio encoder, discrete acoustic tokens, and reinforcement-learning–based training (CoT + RL) to enhance its ability to capture and reproduce voice styles, intonations, and subtle vocal cues. ... -
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ESPnet
End-to-end speech processing toolkit
ESPnet is a comprehensive end-to-end speech processing toolkit covering a wide spectrum of tasks, including automatic speech recognition (ASR), text-to-speech (TTS), speech translation (ST), speech enhancement, speaker diarization, and spoken language understanding. It uses PyTorch as its deep learning engine and adopts a Kaldi-style data processing pipeline for features, data formats, and experimental recipes. This combination allows researchers to leverage modern neural architectures while still benefiting from the robust data preparation practices developed in the speech community. ... -
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JSpeech
Java library designed to integrate Speech-to-Text
jSpeech is a Java library designed to integrate Speech-to-Text (STT) capabilities, command control, and diarization (speaker identification) into applications in a simple, modular, and decoupled way. -
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Vosk Speech Recognition Toolkit
Offline speech recognition API for Android, iOS, Raspberry Pi
Vosk is an offline open source speech recognition toolkit. It enables speech recognition for 20+ languages and dialects - English, Indian English, German, French, Spanish, Portuguese, Chinese, Russian, Turkish, Vietnamese, Italian, Dutch, Catalan, Arabic, Greek, Farsi, Filipino, Ukrainian, Kazakh, Swedish, Japanese, Esperanto, Hindi, Czech, Polish. More to come. Vosk models are small (50 Mb) but provide continuous large vocabulary transcription, zero-latency response with streaming API, reconfigurable vocabulary and speaker identification. ... -
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wukong-robot
Chinese voice dialogue robot/smart speaker project
wukong-robot is a Chinese voice assistant / smart speaker project built to let makers and hackers design highly customizable voice-controlled devices. It combines wake-word detection, automatic speech recognition, natural language understanding, and text-to-speech into a single framework aimed at the Chinese-speaking ecosystem. The project is positioned as a simple, flexible, and elegant platform that can run on devices like Raspberry Pi and other Linux-based boards, making it suitable for DIY smart speakers and home-automation hubs. ... -
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CMU Sphinx
Speech Recognition Toolkit
...----> Maintenance and improvement work has MOVED to https://cmusphinx.github.io/ Please go there for the most recent software and documentation. <---- CMUSphinx is a speaker-independent large vocabulary continuous speech recognizer released under BSD style license. It is also a collection of open source tools and resources that allows researchers and developers to build speech recognition systems. -
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Lip Reading
Cross Audio-Visual Recognition using 3D Architectures
...This code is aimed to provide the implementation for Coupled 3D Convolutional Neural Networks for audio-visual matching. Lip-reading can be a specific application for this work. Audio-visual recognition (AVR) has been considered as a solution for speech recognition tasks when the audio is corrupted, as well as a visual recognition method used for speaker verification in multi-speaker scenarios. The approach of AVR systems is to leverage the extracted information from one modality to improve the recognition ability of the other modality by complementing the missing information. ... -
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Distant Speech Recognition
Beamforming and Speech Recognition Toolkit
BTK contains C++ and Python libraries that implement speech processing and microphone array techniques such as speech feature extraction, speech enhancement, speaker tracking, beamforming, dereverberation and echo cancellation algorithms. The Millennium ASR provides C++ and python libraries for automatic speech recognition. The Millennium ASR implements a weighted finite state transducer (WFST) decoder, training and adaptation methods. These toolkits are meant for facilitating research and development of automatic distant speech recognition. -
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These Matlab codes are the implementation of the TASLP paper, "Speech enhancement based on student t modeling of Teager energy operated perceptual wavelet packet coefficients and a custom thresholding function". In this paper, we showed how the student t distribution can be used to model the perceptual wavelet packet coefficients and how a proper thresholding based on that perform speech enhancement in an efficient way. Teager energy has been used to amplify the difference between the noisy...
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Speaker Recognition System
Speaker Recognition System - Matlab source code
Speaker identity is correlated with the physiological and behavioral characteristics of the speaker. These characteristics exist both in the spectral envelope (vocal tract characteristics) and in the supra-segmental features (voice source characteristics and dynamic features spanning several segments). Index Terms: speaker, recognition, verification, sound, words. -
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Speech Sentiment Analysis
Voice to Text Sentiment Analysis
Voice to text Sentiment analysis converts the audio signal to text to calculate appropriate sentiment polarity of the sentence. The code currently works on one sentence at a time. Sentiment scoring is done on the spot using a speaker. The Speech to text processing system currently being used is the MS Windows speech to text converter. However significant modifications can be made for audio recognition by a refined signal processing system. The sentiment operator in textblob is used for sentiment orientation scoring. The code has been developed in Python 2.7 The following packages are required to be installed before running the program. ... -
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avimmir
(audio, video, image) Multimedia Multimodal Information Retrieval
audio classification; speaker segmentation; speaker clustering; speaker recognition; spoken document retrieval; image retrieval; video retrieval; etc. -
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SRM2 Sound Recognizer Mobile 2
A Sound Recognition Framework developed to J2ME plataform
The SRM Framework 2 was developed for the J2ME platform supporting sound recognition dependent and speaker-independent, through the recognition techniques DTW (Dynamic Time Warping) and HMM (Hidden Markov Models), in addition to providing support to the import and export of data any stage of recognition enabling the use of external resources as well as the tool running steps on remote terminals. It has the following characteristics: - DTW - HMM (Discrete Models) - Suports Sound in PCM - Vector quantization using the k-means algorithm - Uses the algorithm Forward - Uses the algorithm Backward - Uses the Viterbi algorithm - Uses the Baum-Welch algorithm for training - Has features import and export data Further information is on Gaita 2012. -
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Speaker recognition with Matlab
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Arabisc is speaker independent large vocabulary continuous speech recognizer for Arabic language released under GNU license.It is also a collection of open source tools that allows researchers and developers to build speech recognition systems for Arab
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