Emmanuel Schmidbauer (eschmidbauer)

eschmidbauer

User data from Github https://github.com/eschmidbauer

Location:Buffalo, NY

GitHub:@eschmidbauer

Emmanuel Schmidbauer's repositories

websocket-audio-stream

pyaudio & websocket to stream real-time audio to speakers

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voicefixer

General Speech Restoration

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acoustic-model

Acoustic models for: A Comparison of Discrete and Soft Speech Units for Improved Voice Conversion

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CoMoSpeech

one-step diffusion based speech synthesis

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faster-whisper

Faster Whisper ASR transcription with CTranslate2

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FlexFlow

A distributed deep learning framework.

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freeswitch

FreeSWITCH is a Software Defined Telecom Stack enabling the digital transformation from proprietary telecom switches to a versatile software implementation that runs on any commodity hardware. From a Raspberry PI to a multi-core server, FreeSWITCH can unlock the telecommunications potential of any device.

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greenswitch

Battle proven FreeSWITCH Event Socket Protocol client implementation with Gevent

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kamailio

Kamailio - The Open Source SIP Server

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metaseq

Repo for external large-scale work

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peerless

Peerless Animate API

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Speech-Backbones

This is the main repository of open-sourced speech technology by Huawei Noah's Ark Lab.

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Auralis

A Fast TTS Engine

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Kokoro-FastAPI

Dockerized FastAPI wrapper for Kokoro-82M text-to-speech model w/NVIDIA GPU support, queue handling, and auto-stitching

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MeloTTS

High-quality multi-lingual text-to-speech library by MyShell.ai. Support English, Spanish, French, Chinese, Japanese and Korean.

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mod_audio_stream

FreeSWITCH module to stream audio to websocket and receive response

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mod_vad

a voice activity detection module for freeswitch.

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NeMo-text-processing

NeMo text processing for ASR and TTS

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pkg-kamailio-docker

Docker files to easily build Kamailio on different Debian/Ubuntu releases

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RAD-MMM

A TTS model that makes a speaker speak new languages

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RVC_CLI

RVC CLI enables seamless interaction with Retrieval-based Voice Conversion through commands or HTTP requests.

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sherpa-onnx

Speech-to-text, text-to-speech, speaker recognition, and VAD using next-gen Kaldi with onnxruntime without 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, Object Pascal, Lazarus, Rust

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wenet

Production First and Production Ready End-to-End Speech Recognition Toolkit

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whisper-cpp-server

whisper-cpp-server

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whisperd

Unified API for various whisper implementations

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WhisperS2T

An Optimized Speech-to-Text Pipeline for the Whisper Model Supporting Multiple Inference Engine

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whisperX

WhisperX: Automatic Speech Recognition with Word-level Timestamps (& Diarization)

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X-E-Speech-code

X-E-Speech: Joint Training Framework of Non-Autoregressive Cross-lingual Emotional Text-to-Speech and Voice Conversion

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