VoiceCheap: AI Dubbing & Translation for Global Reach

It is a cloud-based voice service and the brain behind tens of millions of devices including the Echo family of devices, FireTV, Fire Tablet, and third-party devices. You can build voice experiences, or skills, that make everyday tasks faster, easier, and more delightful for customers.

It is Google’s largest and most capable AI model. It is built to be multimodal, it can generalize, understand, operate across, and combine different types of info — like text, images, audio, video, and code.

Amazon Polly is a service that turns text into lifelike speech. Polly lets you create applications that talk, enabling you to build entirely new categories of speech-enabled products. Polly is an Amazon AI service that uses advanced deep learning technologies to synthesize speech that sounds like a human voice.

Google Cloud Speech API enables developers to convert audio to text by applying powerful neural network models in an easy to use API. The API recognizes over 80 languages and variants, to support your global user base.

Google Cloud Text-to-Speech enables developers to synthesize natural-sounding speech with 30 voices, available in multiple languages and variants. It applies DeepMind’s groundbreaking research in WaveNet and Google’s powerful neural networks to deliver the highest fidelity possible.

This library is meant to be used with sparse, interpretable features such as those that commonly occur in search (search keywords, filters) or pricing (number of rooms, location, price). It is not as interpretable with problems with very dense non-human interpretable features such as raw pixels or audio samples.

It is a state-of-the-art automatic speech recognition toolkit. It is intended for use by speech recognition researchers and professionals.

It is a general-purpose speech recognition model. It is trained on a large dataset of diverse audio and is also a multi-task model that can perform multilingual speech recognition as well as speech translation and language identification.

Transcribe phone calls or build voice powered apps. Recognize unlimited industry specific words and phrases without any training required. All at simple, affordable pricing.

It provides a set of natural language analysis tools written in Java. It can take raw human language text input and give the base forms of words, their parts of speech, whether they are names of companies, people, etc., normalize and interpret dates, times, and numeric quantities, mark up the structure of sentences in terms of phrases or word dependencies, and indicate which noun phrases refer to the same entities.

Flair allows you to apply our state-of-the-art natural language processing (NLP) models to your text, such as named entity recognition (NER), part-of-speech tagging (PoS), sense disambiguation and classification.

Deepgram helps you harness the potential of your voice data with intelligent speech models built to scale and continuously improve over time. The API is the gateway to Deepgram's Brain AI models, and gives you customizable access to fast, high accuracy transcription and phonetic search. Deepgram Brain can understand nearly every audio format available.

It is an open source embedded (offline, on-device) speech-to-text engine which can run in real time on devices ranging from a Raspberry Pi 4 to high power GPU servers.

It is a unified, developer-friendly API to the best available Speech-To-Text and Text-To-Speech services.

It helps your team record, transcribe, search, and analyze voice conversations.

It is a Python natural language analysis package. It contains tools, which can be used in a pipeline, to convert a string containing human language text into lists of sentences and words, to generate base forms of those words, their parts of speech and morphological features, to give a syntactic structure dependency parse, and to recognize named entities. The toolkit is designed to be parallel among more than 70 languages, using the Universal Dependencies formalism.

It can be used to complement any regular touch user interface with a real time voice user interface. It offers real time feedback for faster and more intuitive experience that enables end user to recover from possible errors quickly and with no interruptions.

It is an on-device speech-to-text engine. By processing voice data locally on the device, it offers private, reliable, fully-customizable, and cost-effective audio transcription experiences. It achieves big tech-level accuracy at a fraction of their costs.

wav2letter++ is a fast open source speech processing toolkit from the Speech Team at Facebook AI Research. It is written entirely in C++ and uses the ArrayFire tensor library and the flashlight machine learning library for maximum efficiency. Our approach is detailed in this arXiv paper.

prose is a natural language processing library (English only, at the moment) in pure Go. It supports tokenization, segmentation, part-of-speech tagging, and named-entity extraction.

It is an open-source voice assistant. It is private by default and completely customizable. It can be freely remixed, extended, and deployed anywhere. It may be used in anything from a science project to a global enterprise environment.

This beta version allows anyone to create their digital voice with only one minute of audio. Simply sign up, record yourself for at least one minute and you will be able to generate any sentence you like with your digital voice.

It is a live transcription tool that provides real-time transcripts for both the user's microphone input (You) and the user's speaker output (Speaker) in a textbox. It also generates a suggested response using OpenAI's GPT-3.5.

It is an open source library that makes it easy to build voice-based LLM apps. Using Vocode, you can build real-time streaming conversations with LLMs and deploy them to phone calls, Zoom meetings, and more.

The purpose of this project is to provide a package for speech processing and feature extraction. This library provides most frequent used speech features including MFCCs and filterbank energies alongside with the log-energy of filterbanks.

It is fully-automated software that can turn any text into a natural lifelike voice-over... In just a few clicks. It can accommodate any business and is perfect for creating voice overs for video sales letters, educational videos, marketing videos, animated videos, podcasts, audio books, and much more!

It is the first multilingual and industry-specific transcription service that can transcribe audio/video with close to human accuracy. It can accurately transcribe conference calls, interviews, podcasts, lectures, and meeting records in more than 30 different languages and dialects. It is now almost as accurate as human transcriptionists.

Jonatasgrosman/wav2vec2 large xlsr 53 english.

It is an On-Premises, Streaming Speech Recognition System built with PyTorch and fastai.

It is a library for advanced Text-to-Speech generation. It’s built on the latest research, was designed to achieve the best trade-off among ease-of-training, speed, and quality. It comes with pre-trained models, tools for measuring dataset quality and is already used in 20+ languages for products and research projects.

It is Stability AI’s first product for music and sound effect generation. Users can create original audio by entering a text prompt and a duration, generating audio in high-quality, 44.1 kHz stereo.

Facebook/seamless m4t v2 large.

Suno/bark.

Turn any idea into a complete song in seconds with TextSong.ai—the most intuitive text to song experience: type, generate, and download high-quality melodies, vocals, and full arrangements ready to share.

Transform videos with AI-powered audio synthesis. Generate perfectly synchronized, high-quality soundtracks instantly. Multiple formats supported. Unlimited usage.

Microsoft/speecht5_hifigan.

Jonatasgrosman/wav2vec2 large xlsr 53 chinese zh cn.

Jonatasgrosman/wav2vec2 large xlsr 53 russian.

Transcribe and translate audio files using OpenAI's Whisper API. You can upload any audio file, and the application will send it through the OpenAI Whisper API using Laravel's queued jobs. Translation makes use of the new OpenAI Chat API and chunks the generated VTT file into smaller parts to fit them into the prompt context limit.

Guillaumekln/faster whisper large v2.

Facebook/wav2vec2 base 960h.

Jonatasgrosman/wav2vec2 large xlsr 53 portuguese.

It is a 1.2B parameter base model trained on 100K hours of speech for TTS (text-to-speech). It empowers developers and businesses to better connect with their audiences at scale.

It is a versatile instant voice cloning approach that requires only a short audio clip from the reference speaker to replicate their voice and generate speech in multiple languages.

It builds upon the capabilities of the WhisperLive and WhisperSpeech by integrating Mistral, a Large Language Model (LLM), on top of the real-time speech-to-text pipeline. Both LLM and Whisper are optimized to run efficiently as TensorRT engines, maximizing performance and real-time processing capabilities.

Alefiury/wav2vec2 large xlsr 53 gender recognition librispeech.

It is a data augmentations library that currently supports four modalities (audio, image, text & video) and over 100 augmentations. Each modality’s augmentations are contained within its own sub-library. These sub-libraries include both function-based and class-based transforms, composition operators, and have the option to provide metadata about the transform applied, including its intensity.

It is a large language model (LLM) that is capable of a wide variety of video generation tasks, including text-to-video, image-to-video, video stylization, video inpainting and outpainting, and video-to-audio.

It is an AI Multitool. Seamlessly harness AI for writing, images, chat, code, speech, and voice – your all-in-one creative powerhouse.

It is a note-taking and journaling app for Notioneers. Just hit record, speak your thoughts and our AI will do the rest. It takes messy voice notes, summarizes them into clear text with AI, and saves them to your notion workspace.