Search Results for "speech" - Page 10
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Showing 353 open source projects for "speech"
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Amiga Memories
A walk along memory lane
...An Amiga Memories video is mostly based on a narrative. The purpose of the script is to define the spoken and written content. The spoken text will be read by a voice synthesizer (Text To Speech or TTS), the written text is simply drawn on the image as subtitles. Here, in addition to the spoken & written narration, the script controls the camera movements as well as the LED activity of the computer. Amiga Memories' video images are computed by the GameStart 3D engine (pre-HARFANG 3D). Although the 3D assets are designed to be played back in real-time with a variable framerate, the engine is capable of breaking down the video sequence into the 30th or 60th of a second, as TGA files. -
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EnCodec
State-of-the-art deep learning based audio codec
...It employs a convolutional encoder–decoder architecture trained with perceptual loss functions that optimize for human auditory quality rather than raw waveform distance. The model can operate in real time and supports variable bandwidths, bitrates, and multi-band audio. Encodec has applications in speech and music compression, generative modeling, and efficient data transmission for communication systems. The repository includes pretrained checkpoints, PyTorch inference code, and examples for integrating Encodec as a module in downstream generative or streaming systems. -
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DiffSinger
Singing Voice Synthesis via Shallow Diffusion Mechanism
DiffSinger is an open-source PyTorch implementation of a diffusion-based acoustic model for singing-voice synthesis (SVS) and also text-to-speech (TTS) in a related variant. The core idea is to view generation of a sung voice (mel-spectrogram) as a diffusion process: starting from noise, the model iteratively “denoises” while being conditioned on a music score (lyrics, pitch, musical timing). This avoids some of the typical problems of prior SVS models — like over-smoothing or unstable GAN training — and produces more realistic, expressive, and natural-sounding singing. ... -
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Automatic YouTube subtitle generation
Using OpenAI's Whisper to automatically generate YouTube subtitles
...It allows users to download videos or audio from YouTube and automatically generate subtitles or transcripts. The tool processes media locally, extracting audio and applying speech recognition to produce accurate text outputs. It supports multiple languages and can handle different Whisper model sizes, balancing performance and accuracy. yt-whisperc is designed for automation, enabling batch processing of multiple videos for transcription workflows. It also provides options for exporting subtitles in common formats such as SRT. ... -
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Pattern
Web mining module for Python, with tools for scraping
...It includes modules for web scraping and crawling that can retrieve information from sources such as social media platforms, search engines, and online knowledge bases. In addition to data mining features, the library offers natural language processing functionality including part-of-speech tagging, sentiment analysis, and n-gram extraction. The framework also includes machine learning algorithms that support classification, clustering, and vector space modeling for text analysis tasks. Another component of the library provides tools for analyzing and visualizing networks, making it useful for studying relationships between entities in large datasets. -
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ASRT Speech Recognition
A Deep-Learning-Based Chinese Speech Recognition System
ASRT is an end-to-end deep-learning Chinese ASR system built with TensorFlow/Keras, using convolution + CTC and a Max-Entropy HMM language model. It provides a REST/gRPC server backend and client SDKs in multiple languages (Python, Java, Go, Windows). Notably lightweight, it performs well without needing GPU acceleration and runs across platforms, targeting developers and researchers building Chinese voice interfaces. -
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WaveRNN
WaveRNN Vocoder + TTS
WaveRNN is a PyTorch implementation of DeepMind’s WaveRNN vocoder, bundled with a Tacotron-style TTS front end to form a complete text-to-speech stack. As a vocoder, WaveRNN models raw audio with a compact recurrent neural network that can generate high-quality waveforms more efficiently than many traditional autoregressive models. The repository includes scripts and code for preprocessing datasets such as LJSpeech, training Tacotron to produce mel spectrograms, training WaveRNN on those spectrograms (with optional GTA data), and finally generating audio. ... -
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VoiceSmith
[WIP] VoiceSmith makes training text to speech models easy
VoiceSmith makes it possible to train and infer on both single and multispeaker models without any coding experience. It fine-tunes a pretty solid text to speech pipeline based on a modified version of DelightfulTTS and UnivNet on your dataset. Both models were pretrained on a proprietary 5000 speaker dataset. It also provides some tools for dataset preprocessing like automatic text normalization. Windows (only CPU supported currently) or any Linux based operating system. If you want to run this on macOS you have to follow the steps in build from source in order to create the installer. ... -
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Tensorflow Transformers
State of the art faster Transformer with Tensorflow 2.0
...These models can be applied on text, for tasks like text classification, information extraction, question answering, summarization, translation, text generation, in over 100 languages. Images, for tasks like image classification, object detection, and segmentation. Audio, for tasks like speech recognition and audio classification. Faster AutoReggressive Decoding, TFlite support, creating TFRecords is simple. Auto-Batching tf.data.dataset or tf.ragged tensors. Everything is dictionary (inputs and outputs) Multiple mask modes like causal, user-defined, prefix. tensorflow-text tokenizer support. Supports GPU, TPU, multi-GPU trainer with wandb, multiple callbacks, auto tensorboard. -
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AugLy
A data augmentations library for audio, image, text, and video
...We designed AugLy to include many specific data augmentations that users perform in real life on internet platforms like Facebook's -- for example making an image into a meme, overlaying text/emojis on images/videos, reposting a screenshot from social media. While AugLy contains more generic data augmentations as well, it will be particularly useful to you if you're working on a problem like copy detection, hate speech detection, etc. -
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SVoice (Speech Voice Separation)
We provide a PyTorch implementation of the paper Voice Separation
SVoice is a PyTorch-based implementation of Facebook Research’s study on speaker voice separation as described in the paper “Voice Separation with an Unknown Number of Multiple Speakers.” This project presents a deep learning framework capable of separating mixed audio sequences where several people speak simultaneously, without prior knowledge of how many speakers are present. The model employs gated neural networks with recurrent processing blocks that disentangle voices over multiple... -
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Voice Cloning App
A Python/Pytorch app for easily synthesising human voices
A Python/Pytorch app for easily synthesizing human voices. If you are using a language other than English you can add it to the app. Firstly, you'll need to find a deep speech model for your language by going to coqui. You'll then need to download the model.pbmm and alphabet.txt files for your language. Requires Windows 10 or Ubuntu 20.04+ operating system, 5GB+ Disk space, and NVIDIA GPU with at least 4GB of memory & driver version 456.38+ (optional). Automatic dataset generation (with support for subtitles and audiobooks) Additional language support. ... -
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AutoSub
A CLI script to generate subtitle files (SRT/VTT/TXT) for any video
...AutoSub leverages FFmpeg for media handling and integrates with speech recognition engines for transcription. It is particularly useful for content creators who want to quickly produce subtitles without manual effort. Overall, it simplifies the process of making media content accessible and searchable. -
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KoNLPy
Python package for Korean natural language processing
KoNLPy is a natural language processing (NLP) library for the Korean language, offering tokenization, morphological analysis, and named entity recognition. -
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Music Source Separation
Separate audio recordings into individual sources
...It aims to give users the ability to take any existing song and decompose it into separate stems (vocals, accompaniment, etc.), or to train custom separation models on their own datasets (e.g. for speech enhancement, instrument isolation, or other audio-separation tasks). The repository provides training scripts (e.g. using datasets such as MUSDB18), preprocessing steps (audio-to-HDF5 packing, indexing), evaluation pipelines, and inference scripts to perform separation on arbitrary audio files. This makes the project useful both for researchers in music information retrieval / audio machine learning and for hobbyists or practitioners who want to experiment with remixing, karaoke, or audio editing. -
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Mycroft
Mycroft Core, the Mycroft Artificial Intelligence platform
Mycroft is the world’s leading open source voice assistant. It is private by default and completely customizable. Our software runs on many platforms, on desktop, our reference hardware, a Raspberry Pi, or your own custom hardware. Our open-source, modular system can be ported to your device or environment, at any price point. Whether you make voice-assistants, televisions, or microwaves. Whether you have a 5-room BnB or a 1000-room hotel. Your customers will get access to all the... -
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Project Alice
Main repository of Project Alice, contains main unit source code
...However, as an option, since we've built Project Alice on top of Snips, Project Alice can be configured to use some online alternatives and fall backs (for example, using Amazon or Google’s Text to Speech engines), just like Snips. Since Snips (and the Project Alice team) strongly believe that decisions about your privacy should be made by you and you alone, these options are all disabled by default. The original code base was started at the end 2015 and several rewrites made it what it is today. It was entirely written by me Psycho until recently, where I decided to make the code openly available to the world. -
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Mocking Bird
Clone a voice in 5 seconds to generate arbitrary speech in real-time
MockingBird is an open-source voice cloning and real-time speech generation toolkit that lets you clone a speaker’s voice from a short audio sample (reportedly as little as 5 seconds) and then synthesize arbitrary speech in that voice. It builds on deep-learning based TTS / voice-cloning technology (in the lineage of projects such as Real-Time-Voice-Cloning), but extends it with support for Mandarin Chinese and multiple Chinese speech datasets — broadening its applicability beyond English. ... -
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VoiceFixer
General Speech Restoration
...Unlike many single-purpose noise reduction tools, VoiceFixer targets a “general speech restoration” problem (GSR), capable of handling multiple types of distortions at once, which makes it suitable for old recordings, phone-call audio, amateur voice recordings, or archival media. Evaluations show that VoiceFixer significantly improves both objective and subjective audio quality compared to baseline speech-enhancement methods. -
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Parakeet
PAddle PARAllel text-to-speech toolKIT
PAddle PARAllel text-to-speech toolKIT (supporting Tacotron2, Transformer TTS, FastSpeech2/FastPitch, SpeedySpeech, WaveFlow and Parallel WaveGAN) Parakeet aims to provide a flexible, efficient and state-of-the-art text-to-speech toolkit for the open-source community. It is built on PaddlePaddle dynamic graph and includes many influential TTS models. In order to facilitate exploiting the existing TTS models directly and developing the new ones, Parakeet selects typical models and provides their reference implementations in PaddlePaddle. ... -
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Jarvis Python AI Assistant
Python AI assistant
Jarvis is a voice commanding assistant service in Python 3.8 It can recognize human speech, talk to user and execute basic commands. Opens a web page (e.g 'Jarvis open youtube') Play music in Youtube (e.g 'Jarvis play mozart') Increase/decrease the speakers master volume (also can set max/mute speakers volume) (e.g 'Jarvis volume up!') Opens libreoffice suite applications (calc, writer, impress) (e.g 'Jarvis open calc') Tells about something, by searching on the internet (e.g 'Jarvis tells me about oranges') Tells the weather for a place (e.g 'Jarvis tell_the_skills me the weather in London') Tells the current time and/or date (e.g 'Jarvis tell me time or date') Set an alarm (e.g 'Jarvis create a new alarm') Tells the internet speed (ping, uplink and downling) (e.g 'Jarvis tell_the_skills me the internet speed') Tells the internet availability (e.g 'Jarvis is the internet connection ok?') ... -
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TensorFlowTTS
Real-Time State-of-the-art Speech Synthesis for Tensorflow 2
TensorFlowTTS is a state-of-the-art, open-source speech synthesis library built on TensorFlow 2. It offers a variety of architectures for text-to-speech, including classic and modern models such as Tacotron‑2, FastSpeech / FastSpeech2, and neural vocoders like MelGAN and Multiband‑MelGAN. Because it’s based on TensorFlow 2, it can leverage optimizations such as fake-quantization aware training and pruning — which allow models to run faster than real time and to be deployable on mobile or embedded platforms. ... -
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VITS
Conditional Variational Autoencoder with Adversarial Learning
...The repository provides training and inference pipelines for common datasets such as LJ Speech (single-speaker) and VCTK (multi-speaker), including filelists, configs, and preprocessing scripts. It also includes monotonic alignment search code and g2p preprocessing, which are crucial components for aligning text and speech in an end-to-end setup. -
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Kashgari
Kashgari is a production-level NLP Transfer learning framework
Kashgari is a simple and powerful NLP Transfer learning framework, build a state-of-art model in 5 minutes for named entity recognition (NER), part-of-speech tagging (PoS), and text classification tasks. -
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Transformer TTS
Implementation of a Transformer based neural network
TransformerTTS is an implementation of a non-autoregressive Transformer-based neural network for text-to-speech, built with TensorFlow 2. It takes inspiration from architectures like FastSpeech, FastSpeech 2, FastPitch, and Transformer TTS, and extends them with its own aligner and forward models. The system separates alignment learning and acoustic modeling: an autoregressive Transformer is used as an aligner to extract phoneme-to-frame durations, while a non-autoregressive “ForwardTransformer” generates mel-spectrograms conditioned on text and durations. ...
