Search Results for "batch normalize audio"
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Showing 21 open source projects for "batch normalize audio"
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Ultimate Vocal Remover (UVR5)
GUI for a Vocal Remover that uses Deep Neural Networks
This application uses state-of-the-art source separation models to remove vocals from audio files. UVR's core developers trained all of the models provided in this package (except for the Demucs v3 and v4 4-stem models). -
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abogen
Generate audiobooks from EPUBs, PDFs and text with captions
...In other words, it automates the pipeline of reading a digital book (or document), converting its text into speech via a TTS engine, and packaging the result into an audiobook format — likely along with timestamped captions or subtitles that align with the spoken audio. This can be very useful for accessibility, content consumption on the go, or for users who prefer audio over reading. The repository supports handling common ebook formats and generating outputs that combine audio plus caption metadata. By automating text-to-speech for arbitrary documents, abogen reduces the friction of producing audiobooks and could be integrated into larger workflows (e.g., batch converting a library of texts). -
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VoiceStudio
VoiceStudio is the open-source, fully-local ElevenLabs alternative
...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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WanGP
AI video generator optimized for low VRAM and older GPUs use
Wan2GP is an open source AI video generation toolkit designed to make modern generative models accessible on consumer-grade hardware with limited GPU memory. It acts as a unified interface for running multiple video, image, and audio generation models, including Wan-based models as well as other systems like Hunyuan Video, Flux, and Qwen. A key focus of the project is reducing VRAM requirements, enabling some workflows to run on as little as 6 GB while still supporting older Nvidia and... -
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5
OpenHiggsfield AI
A studio for image and video generation
...Its catalog includes dozens of image and video models from several providers. Each model exposes only the settings it supports, such as aspect ratio, resolution, duration, audio, batch size, and prompt enhancement. Users can attach uploads or reuse media from previous generations as references, start frames, end frames, video, or audio inputs. Completed generations are organized into a shared history and asset library. The application is built with Next.js and React and can be self-hosted or used with the hosted interface using the user’s own platform credentials. -
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edge-tts
Use Microsoft Edge's online text-to-speech service from Python
...The library is asynchronous under the hood, which makes it efficient for batch jobs or web services that need to synthesize many utterances concurrently. -
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Voice-Pro
Comprehensive Gradio WebUI for audio processing
Voice-Pro is the best gradio WebUI for transcription, translation and text-to-speech. It can be easily installed with one click. Create a virtual environment using Miniconda, running completely separate from the Windows system (fully portable). Supports real-time transcription and translation, as well as batch mode. -
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Hugging Face - Speech To Speech
Open speech-to-speech models and pipelines by Hugging Face toolkit AI
...It integrates with the broader Hugging Face ecosystem, making it easier to load pretrained models and run inference. It also serves as a foundation for building real-time or batch audio transformation systems. Overall, it highlights an emerging approach to voice technology that reduces latency and preserves more of the original speech characteristics. -
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Qwen3-ASR
Qwen3-ASR is an open-source series of ASR models
Qwen3-ASR is an automatic speech recognition system in the QwenLM family, developed to convert spoken language into text with strong accuracy and real-time performance. As a specialized ASR variant of the broader Qwen language model ecosystem, it focuses on capturing reliable transcriptions from audio sources such as recordings, live streams, or conversational inputs while supporting low latency use cases. The architecture combines advanced neural acoustic modeling with context-aware... -
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notebooklm-py
Unofficial Python API and agentic skill for Google NotebookLM
notebooklm-py is an unofficial Python API and agent-ready integration layer for Google NotebookLM that exposes NotebookLM functionality through code, the command line, and AI agent workflows. Its goal is to provide programmatic access not just to standard notebook operations, but also to many capabilities that are either limited or unavailable in the web interface, making it especially useful for automation and custom pipelines. The project covers notebook management, source ingestion,... -
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AI YouTube Shorts Generator
A python tool that uses GPT-4, FFmpeg, and OpenCV
AI-YouTube-Shorts-Generator is a Python-based tool that automates the creation of short-form vertical video clips (“shorts”) from longer source videos — ideal for adapting content for platforms like YouTube Shorts, Instagram Reels, or TikTok. It analyzes input video (whether a local file or a YouTube URL), transcribes audio (with optional GPU-accelerated speech-to-text), uses an AI model to identify the most compelling or engaging segments, and then crops/resizes the video and applies... -
12
LTX-Video
Official repository for LTX-Video
LTX-Video is a sophisticated multimedia processing framework from Lightricks designed to handle high-quality video editing, compositing, and transformation tasks with performance and scalability. It provides runtime components that efficiently decode, encode, and manipulate video streams, frame buffers, and audio tracks while exposing a rich API for building customized editing features like transitions, effects, color grading, and keyframe automation. The toolkit is built with both real-time and offline workflows in mind, enabling applications from consumer editing to professional content creation and batch processing. Internally optimized for multi-core processors and hardware acceleration where available, LTX-Video makes it feasible to work with high-resolution content and complex timelines without sacrificing responsiveness. -
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gTTS
Python library and CLI tool to interface with Google Translate
...A small CLI utility, gtts-cli, makes it easy to test or batch-generate MP3 files right from the shell. -
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comfyui-mixlab-nodes
Workflow and speech recognition app
comfyui-mixlab-nodes is a large collection of custom nodes for ComfyUI that turns workflows into interactive apps and adds real-time multimedia, LLM, and TTS capabilities. It introduces a “Workflow-to-APP” concept, where a ComfyUI graph can be transformed into a Web App through an AppInfo node, complete with categories, batch prompts, and editable configurations. The project also brings Real-time Design features like screen capture and floating video nodes, enabling creative pipelines that mix live screen content, generative models, and visual effects. For audio and speech, it provides nodes for SpeechRecognition and SpeechSynthesis, plus workflows that combine voice generation with real-time face swapping and other audio-visual effects. ... -
15
ChatTTS_colab
One-click deployment (including offline integration package)
...It provides an integrated offline bundle and scripts for Windows and macOS so users can run ChatTTS locally without wrestling with complex environment setup. The repository includes Colab notebooks that launch a Gradio-based web UI and expose streaming TTS, making it possible to listen to generated audio as it is produced. A distinctive feature is the “voice gacha” system, which batch-generates many distinct voice timbres and allows users to save the ones they like into a curated voice library. It has first-class support for long-form audio generation, making it suitable for audiobooks, podcasts, or long narration tasks. The project also implements multi-speaker or role-based reading, letting users assign different voices to different characters in a script and even use a large language model to generate that script in one step. -
16
Demucs
Code for the paper Hybrid Spectrogram and Waveform Source Separation
...Demucs supports GPU-accelerated inference and can process multi-channel audio with chunked streaming for real-time or batch operation. It also provides training scripts and utilities to fine-tune on custom datasets, along with remixing and enhancement tools. -
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TTS-Vue
Microsoft speech synthesis tool, built with Electron
TTS-Vue is a desktop text-to-speech application built with Electron, Vue, ElementPlus, and Vite, focused on using Microsoft’s official Speech API for high-quality neural synthesis. It wraps the Microsoft TTS WebSocket interface in a clean UI so users can paste or load text, choose voices, tweak parameters, and export audio without touching raw API calls. The app supports SSML (Speech Synthesis Markup Language), letting power users specify fine-grained control over pronunciation, pauses, prosody, and emphasis using XML-like markup. It includes batch conversion: users can select multiple .txt files and convert them into audio in one go, making it handy for large text collections or repetitive tasks. ... -
18
NAM-Runner
Batch file to install and run NAM (neural-amp-modeler) easily.
A Windows 10 batch file, that installs and runs the NAM model trainer (neural-amp-modeler) by Steven Atkinson right into the GUI application. Fully automated. Custom one-time installation of everything you need to train neural network models of guitar amps and more for the NAM VST plugin, no Conda required. Runs as a launcher afterwards. Portable installation. New pyTorch inclues CUDA runtime for fast Nvidia GPU support. No command line, python or conda knowledge needed! Just double click. -
19
VideoSrt
Windows-GUI
...VideoSrtIt is written in Golanglanguage and developed based on lxn/walk Windows-GUI toolkit. Open source software tool that can recognize video speech and automatically generate subtitle SRT files. It is suitable for business scenarios that quickly and batch generate Chinese/English subtitles and text files for media (video/audio). Recognize video/audio speech to generate subtitle files (support Chinese-English translation, bilingual subtitles) Extract speech text from video/audio. Batch translation, filter processing/encoding SRT subtitle files. Using the Alibaba Cloud speech recognition interface, the accuracy is high, and the standard Mandarin/English recognition rate is over 95%. ... -
20
Denoiser
Real Time Speech Enhancement in the Waveform Domain (Interspeech 2020)
...The implementation includes data augmentation techniques applied to the raw waveforms (e.g. noise mixing, reverberation) to improve model robustness and generalization to diverse noise types. The project supports both offline denoising (batch inference) and live audio processing (e.g. via loopback audio interfaces), making it practical for real-time use in calls or recording. The codebase includes training and evaluation scripts, configuration management via Hydra, and pretrained models on standard noise datasets. -
21
cmdSayIt
Command Line Text to Speech tool for windows
This program uses SAPI voices for Text to speech from the commandline. its a simple tool written in C# Samples included for English and Japanese, but you will need a Japanese voice installed on your PC
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