Note frequencies for equal-tempered scale
Schedule looped playback of Web Audio notes at 96 ticks per beat
React library for audio recording and visualization using Web Audio API
Web Assembly streaming Opus decoder with Machine Learning enhancements
<p align="center"> <img src="https://github.com/staskobzar/vue-audio-visual/blob/master/static/logo.png?raw=true"/> </p>
Decode audio data in node or browser
Javascript audio library for the modern web.
React webcam component
A simple React wrapper for the audio tag
A cross-browser wrapper for the Web Audio API which aims to closely follow the standard.
A custom element for the Spotify player with an API that aims to match the `<audio>` API
React library for audio recording and visualization using Web Audio API
The AudioWorkletProcessor which is used by the recorder-audio-worklet package.
JavaScript HLS client using MediaSourceExtension
This module provides a loader for the RecorderAudioWorkletProcessor and the corresponding RecorderAudioWorkletNode.
JSON Web Token implementation (symmetric and asymmetric)
Livekit plugin for noise cancellation of inbound AudioStream for NodeJS
Web Assembly streaming FLAC decoder
Web Assembly streaming Ogg Vorbis decoder
<div align="center"> <img width="200" height="200" src="https://s3.amazonaws.com/pix.iemoji.com/images/emoji/apple/ios-11/256/crayon.png"> <h1>@jimp/plugin-dither</h1> <p>Apply a dither effect to an image.</p> </div>
Detect the audio type of a Buffer/Uint8Array
semantic-release plugin to generate changelog content with conventional-changelog
Library that parses raw data from audio codecs into frames containing data, header values, duration, and other information.
Core library for media playback in the browser shared by mux elements
This gem can fingerprint from small to large pieces of wav audio and run a math to compare them (this is very handy to compare audio notes)
Google Speech-to-Text enables developers to convert audio to text by applying powerful neural network models in an easy-to-use API. The API recognizes more than 120 languages and variants to support your global user base. You can enable voice command-and-control, transcribe audio from call centers, and more. It can process real-time streaming or prerecorded audio, using Google's machine learning technology. Note that google-cloud-speech-v1 is a version-specific client library. For most uses, we recommend installing the main client library google-cloud-speech instead. See the readme for more details.
Google Speech-to-Text enables developers to convert audio to text by applying powerful neural network models in an easy-to-use API. The API recognizes more than 120 languages and variants to support your global user base. You can enable voice command-and-control, transcribe audio from call centers, and more. It can process real-time streaming or prerecorded audio, using Google's machine learning technology. Note that google-cloud-speech-v1p1beta1 is a version-specific client library. For most uses, we recommend installing the main client library google-cloud-speech instead. See the readme for more details.
Google Speech-to-Text enables developers to convert audio to text by applying powerful neural network models in an easy-to-use API. The API recognizes more than 120 languages and variants to support your global user base. You can enable voice command-and-control, transcribe audio from call centers, and more. It can process real-time streaming or prerecorded audio, using Google's machine learning technology. Note that google-cloud-speech-v2 is a version-specific client library. For most uses, we recommend installing the main client library google-cloud-speech instead. See the readme for more details.
Text-to-Speech converts text or Speech Synthesis Markup Language (SSML) input into audio data of natural human speech. Note that google-cloud-text_to_speech-v1 is a version-specific client library. For most uses, we recommend installing the main client library google-cloud-text_to_speech instead. See the readme for more details.
Text-to-Speech converts text or Speech Synthesis Markup Language (SSML) input into audio data of natural human speech. Note that google-cloud-text_to_speech-v1beta1 is a version-specific client library. For most uses, we recommend installing the main client library google-cloud-text_to_speech instead. See the readme for more details.
The library is based around an abstract representation for music notation, including pitch, note, dynamic, score, etc. A Ruby-based DSL is provided to aid in composition. Scores can be converted to common formats, like MIDI and LilyPond. Scores can also be rendered as audio via SuperCollider.
Media Translation API delivers real-time speech translation to your content and applications directly from your audio data. Leveraging Google???s machine learning technologies, the API offers enhanced accuracy and simplified integration while equipping you with a comprehensive set of features to further refine your translation results. Improve user experience with low-latency streaming translation and scale quickly with straightforward internationalization. Note that google-cloud-media_translation-v1beta1 is a version-specific client library. For most uses, we recommend installing the main client library google-cloud-media_translation instead. See the readme for more details.
Trim an audio or video file using ffmpeg - Works with all formats supported by ffmpeg, including mp3, mp4, mkv, and many more. - Seeks to the nearest frame positions by re-encoding the media. - Reduces file size procduced by OBS Studio by over 80 percent. - Can be used as a Ruby gem. - Installs the 'trim' command. When run as a command, output files are named by adding a 'trim.' prefix to the media file name, e.g. 'dir/trim.file.ext'. By default, the trim command does not overwrite pre-existing output files. When trimming is complete, the trim command displays the trimmed file, unless the -q option is specified Command-line Usage: trim [OPTIONS] dir/file.ext start [[to|for] end] - The start and end timecodes have the format [HH:[MM:]]SS[.XXX] Note that decimal seconds may be specified, bug frames may not; this is consistent with how ffmpeg parses timecodes. - end defaults to end of the audio/video file OPTIONS are: -d Enable debug output. -f Overwrite output file if present. -h Display help information. -v Verbose output. -V Do not @view the trimmed file when complete. Examples: # Crop dir/file.mp4 from 15.0 seconds to the end of the video, save to demo/trim.demo.mp4: trim demo/demo.mp4 15 # Crop dir/file.mkv from 3 minutes, 25 seconds to 9 minutes, 35 seconds, save to demo/trim.demo.mp4: trim demo/demo.mp4 3:25 9:35 # Same as the previous example, using optional 'to' syntax: trim demo/demo.mp4 3:25 to 9:35 # Save as the previous example, but specify the duration instead of the end time by using the for keyword: trim demo/demo.mp4 3:25 for 6:10
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